Hello all, I forget what the concensus was when we discussed this before, but I seem to be finding difficulties finding a softphone that has both accessibility as well as a good feature set; particularly, I need the ability to pull contacts and what not from Active Directory since we use that; I've heard of Microsip, though I know that it does not support this feature. If X-Lite/Bria used to be accessible, they aren't now; unlabeled fields everywhere which cannot be seemingly deciphered via object navigation with NVDA (Haven't tried JAWS, but I will.) We're using Asterisk for the server if that matters. Thanks.
I didn't even consider it, honestly since I wasn't sure whether or not it would work with Asterisk, considering they love to tout their own PBX server. Thanks for the reminder, though. ----- Original Message ----- From: "Billy Irwin" <billy.irwin@outlook.com> To: "Blind sysadmins list" <blind-sysadmins@lists.hodgsonfamily.org> Sent: Tuesday, September 19, 2017 4:42:13 PM Subject: Re: [Blind-sysadmins] softphone recommendations Hi Cathryn, Have you looked at 3CX? Thanks, Billy _______________________________________________ Blind-sysadmins mailing list Blind-sysadmins@lists.hodgsonfamily.org https://lists.hodgsonfamily.org/listinfo/blind-sysadmins
I didn't even consider it, honestly since I wasn't sure whether or not it would work with Asterisk, considering they love to tout their own PBX server. Thanks for the reminder, though. ----- Original Message ----- From: "Billy Irwin" <billy.irwin@outlook.com> To: "Blind sysadmins list" <blind-sysadmins@lists.hodgsonfamily.org> Sent: Tuesday, September 19, 2017 4:42:13 PM Subject: Re: [Blind-sysadmins] softphone recommendations Hi Cathryn, Have you looked at 3CX? Thanks, Billy _______________________________________________ Blind-sysadmins mailing list Blind-sysadmins@lists.hodgsonfamily.org https://lists.hodgsonfamily.org/listinfo/blind-sysadmins
I've used it several times with Asterisk. It is very easy to configure. I run Asterisk as my server for my customers who want a hosted PBX. Billy L. Irwin - K9OH Section Emergency Coordinator ARRL South Carolina Section www.ares-sc.org Phone: 803-497-5560 Office: 803-567-2400 E-Mail: billy.irwin@outlook.com -----Original Message----- From: Blind-sysadmins [mailto:blind-sysadmins-bounces@lists.hodgsonfamily.org] On Behalf Of Katherine M. Moss Sent: Tuesday, September 19, 2017 8:15 PM To: Blind sysadmins list <blind-sysadmins@lists.hodgsonfamily.org> Subject: Re: [Blind-sysadmins] softphone recommendations I didn't even consider it, honestly since I wasn't sure whether or not it would work with Asterisk, considering they love to tout their own PBX server. Thanks for the reminder, though. ----- Original Message ----- From: "Billy Irwin" <billy.irwin@outlook.com> To: "Blind sysadmins list" <blind-sysadmins@lists.hodgsonfamily.org> Sent: Tuesday, September 19, 2017 4:42:13 PM Subject: Re: [Blind-sysadmins] softphone recommendations Hi Cathryn, Have you looked at 3CX? Thanks, Billy _______________________________________________ Blind-sysadmins mailing list Blind-sysadmins@lists.hodgsonfamily.org https://lists.hodgsonfamily.org/listinfo/blind-sysadmins _______________________________________________ Blind-sysadmins mailing list Blind-sysadmins@lists.hodgsonfamily.org https://lists.hodgsonfamily.org/listinfo/blind-sysadmins
I've used it several times with Asterisk. It is very easy to configure. I run Asterisk as my server for my customers who want a hosted PBX. Billy L. Irwin - K9OH Section Emergency Coordinator ARRL South Carolina Section www.ares-sc.org Phone: 803-497-5560 Office: 803-567-2400 E-Mail: billy.irwin@outlook.com -----Original Message----- From: Blind-sysadmins [mailto:blind-sysadmins-bounces@lists.hodgsonfamily.org] On Behalf Of Katherine M. Moss Sent: Tuesday, September 19, 2017 8:15 PM To: Blind sysadmins list <blind-sysadmins@lists.hodgsonfamily.org> Subject: Re: [Blind-sysadmins] softphone recommendations I didn't even consider it, honestly since I wasn't sure whether or not it would work with Asterisk, considering they love to tout their own PBX server. Thanks for the reminder, though. ----- Original Message ----- From: "Billy Irwin" <billy.irwin@outlook.com> To: "Blind sysadmins list" <blind-sysadmins@lists.hodgsonfamily.org> Sent: Tuesday, September 19, 2017 4:42:13 PM Subject: Re: [Blind-sysadmins] softphone recommendations Hi Cathryn, Have you looked at 3CX? Thanks, Billy _______________________________________________ Blind-sysadmins mailing list Blind-sysadmins@lists.hodgsonfamily.org https://lists.hodgsonfamily.org/listinfo/blind-sysadmins _______________________________________________ Blind-sysadmins mailing list Blind-sysadmins@lists.hodgsonfamily.org https://lists.hodgsonfamily.org/listinfo/blind-sysadmins
What made you choose Asterisk over FreeSwitch? On Wed, 20 Sep 2017 00:17:02 +0000, you wrote:
I've used it several times with Asterisk. It is very easy to configure. I run Asterisk as my server for my customers who want a hosted PBX.
Billy L. Irwin - K9OH Section Emergency Coordinator ARRL South Carolina Section www.ares-sc.org Phone: 803-497-5560 Office: 803-567-2400 E-Mail: billy.irwin@outlook.com
-----Original Message----- From: Blind-sysadmins [mailto:blind-sysadmins-bounces@lists.hodgsonfamily.org] On Behalf Of Katherine M. Moss Sent: Tuesday, September 19, 2017 8:15 PM To: Blind sysadmins list <blind-sysadmins@lists.hodgsonfamily.org> Subject: Re: [Blind-sysadmins] softphone recommendations
I didn't even consider it, honestly since I wasn't sure whether or not it would work with Asterisk, considering they love to tout their own PBX server. Thanks for the reminder, though.
----- Original Message ----- From: "Billy Irwin" <billy.irwin@outlook.com> To: "Blind sysadmins list" <blind-sysadmins@lists.hodgsonfamily.org> Sent: Tuesday, September 19, 2017 4:42:13 PM Subject: Re: [Blind-sysadmins] softphone recommendations
Hi Cathryn,
Have you looked at 3CX?
Thanks,
Billy
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Express Talk from NCH looks accessible. http://www.nch.com.au/talk/index.html The command-line version of Linphone may also be worth a look; the Windows GUI version is not accessible. http://www.linphone.org/ Please let me know if you find anything better. Chris On Tue, Sep 19, 2017 at 04:30:22PM -0400, Katherine M. Moss wrote:
Hello all,
I forget what the concensus was when we discussed this before, but I seem to be finding difficulties finding a softphone that has both accessibility as well as a good feature set; particularly, I need the ability to pull contacts and what not from Active Directory since we use that; I've heard of Microsip, though I know that it does not support this feature. If X-Lite/Bria used to be accessible, they aren't now; unlabeled fields everywhere which cannot be seemingly deciphered via object navigation with NVDA (Haven't tried JAWS, but I will.) We're using Asterisk for the server if that matters. Thanks.
_______________________________________________ Blind-sysadmins mailing list Blind-sysadmins@lists.hodgsonfamily.org https://lists.hodgsonfamily.org/listinfo/blind-sysadmins
Express Talk has a strange issue for me where several seconds of delay will build up after about an hour of a SIP call being connected. I haven't had this delay issue when using Bria on iOS. Express Talk is very accessible. Let me know if you try it and have the delay issue. Chris On Tue, Sep 19, 2017 at 08:21:34PM -0500, Chris Nestrud wrote:
Express Talk from NCH looks accessible.
http://www.nch.com.au/talk/index.html
The command-line version of Linphone may also be worth a look; the Windows GUI version is not accessible.
Please let me know if you find anything better.
Chris
On Tue, Sep 19, 2017 at 04:30:22PM -0400, Katherine M. Moss wrote:
Hello all,
I forget what the concensus was when we discussed this before, but I seem to be finding difficulties finding a softphone that has both accessibility as well as a good feature set; particularly, I need the ability to pull contacts and what not from Active Directory since we use that; I've heard of Microsip, though I know that it does not support this feature. If X-Lite/Bria used to be accessible, they aren't now; unlabeled fields everywhere which cannot be seemingly deciphered via object navigation with NVDA (Haven't tried JAWS, but I will.) We're using Asterisk for the server if that matters. Thanks.
_______________________________________________ Blind-sysadmins mailing list Blind-sysadmins@lists.hodgsonfamily.org https://lists.hodgsonfamily.org/listinfo/blind-sysadmins
Sounds like it could be a problem in Jitter buffer management. What do you think? Have you tried adjusting the type and size of buffer negotiated by the PBX?
On Sep 21, 2017, at 8:56 AM, Chris Nestrud <ccn@chrisnestrud.com> wrote:
Express Talk has a strange issue for me where several seconds of delay will build up after about an hour of a SIP call being connected. I haven't had this delay issue when using Bria on iOS. Express Talk is very accessible. Let me know if you try it and have the delay issue.
Chris
On Tue, Sep 19, 2017 at 08:21:34PM -0500, Chris Nestrud wrote:
Express Talk from NCH looks accessible.
http://www.nch.com.au/talk/index.html
The command-line version of Linphone may also be worth a look; the Windows GUI version is not accessible.
Please let me know if you find anything better.
Chris
On Tue, Sep 19, 2017 at 04:30:22PM -0400, Katherine M. Moss wrote:
Hello all,
I forget what the concensus was when we discussed this before, but I seem to be finding difficulties finding a softphone that has both accessibility as well as a good feature set; particularly, I need the ability to pull contacts and what not from Active Directory since we use that; I've heard of Microsip, though I know that it does not support this feature. If X-Lite/Bria used to be accessible, they aren't now; unlabeled fields everywhere which cannot be seemingly deciphered via object navigation with NVDA (Haven't tried JAWS, but I will.) We're using Asterisk for the server if that matters. Thanks.
_______________________________________________ Blind-sysadmins mailing list Blind-sysadmins@lists.hodgsonfamily.org https://lists.hodgsonfamily.org/listinfo/blind-sysadmins
_______________________________________________ Blind-sysadmins mailing list Blind-sysadmins@lists.hodgsonfamily.org https://lists.hodgsonfamily.org/listinfo/blind-sysadmins
I haven't had a chance to dig into this further. That sounds like a logical place to begin investigating. Thanks. Chris On Thu, Sep 21, 2017 at 02:29:25PM +0000, Scott Granados wrote:
Sounds like it could be a problem in Jitter buffer management. What do you think? Have you tried adjusting the type and size of buffer negotiated by the PBX?
On Sep 21, 2017, at 8:56 AM, Chris Nestrud <ccn@chrisnestrud.com> wrote:
Express Talk has a strange issue for me where several seconds of delay will build up after about an hour of a SIP call being connected. I haven't had this delay issue when using Bria on iOS. Express Talk is very accessible. Let me know if you try it and have the delay issue.
Chris
On Tue, Sep 19, 2017 at 08:21:34PM -0500, Chris Nestrud wrote:
Express Talk from NCH looks accessible.
http://www.nch.com.au/talk/index.html
The command-line version of Linphone may also be worth a look; the Windows GUI version is not accessible.
Please let me know if you find anything better.
Chris
On Tue, Sep 19, 2017 at 04:30:22PM -0400, Katherine M. Moss wrote:
Hello all,
I forget what the concensus was when we discussed this before, but I seem to be finding difficulties finding a softphone that has both accessibility as well as a good feature set; particularly, I need the ability to pull contacts and what not from Active Directory since we use that; I've heard of Microsip, though I know that it does not support this feature. If X-Lite/Bria used to be accessible, they aren't now; unlabeled fields everywhere which cannot be seemingly deciphered via object navigation with NVDA (Haven't tried JAWS, but I will.) We're using Asterisk for the server if that matters. Thanks.
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My recommendation changes from Express Talk to Phoner Lite. Phoner Lite has some unlabeled buttons and is not as accessible as Express Talk., but Phoner Lite does not have this strange delay issue. Using same hardware and two softphones with same codec, looks like it might be an Express Talk issue. Express Talk: Dial 9196, immediately start saying numbers quickly, stopping when numbers are heard, 1-12 is heard from echo. The echo application is working and sends back pretty much all data but with this 1.5 second delay. Free Switch log of Express Talk connection: 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:4352 Set telephone-event payload to 101@8000 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:3056 Set Codec sofia/internal/1000@freeswitch.server PCMU/8000 20 ms 160 samples 64000 bits 1 channels 2017-09-23 14:44:06.997035 [DEBUG] switch_core_codec.c:111 sofia/internal/1000@freeswitch.server Original read codec set to PCMU:0 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:4695 Set telephone-event payload to 101@8000 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:4754 sofia/internal/1000@freeswitch.server Set 2833 dtmf send payload to 101 recv payload to 101 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:6865 AUDIO RTP [sofia/internal/1000@freeswitch.server] 72.14.x.x port 29276 -> 71.238.x.x port 8000 codec: 0 ms: 20 2017-09-23 14:44:06.997035 [DEBUG] switch_rtp.c:4096 Starting timer [soft] 160 bytes per 20ms 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:7166 sofia/internal/1000@freeswitch.server Set 2833 dtmf send payload to 101 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:7173 sofia/internal/1000@freeswitch.server Set 2833 dtmf receive payload to 101 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:7196 sofia/internal/1000@freeswitch.server Set rtp dtmf delay to 40 2017-09-23 14:44:06.997035 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/1000@freeswitch.server! 2017-09-23 14:44:06.997035 [DEBUG] switch_channel.c:3473 (sofia/internal/1000@freeswitch.server) Callstate Change RINGING -> EARLY 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:6848 Audio params are unchanged for sofia/internal/1000@freeswitch.server. Phoner Lite: Same computer and audio device as Express Talk. I set the codec to G.711 u-Law, 64 kbps which matches Express Talk. It was originally set to Opus, 50kbps. Dial 9196, immediately start saying numbers quickly, stopping when numbers are heard, 2 is heard from echo. The echo application is working and sends back data with minimal delay. Free Switch log of Phoner Lite connection: 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:4352 Set telephone-event payload to 101@8000 2017-09-23 15:08:52.917044 [WARNING] switch_core_media.c:4606 Invalid SDP for opus. Don't ask.. but it needs a /2 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:3056 Set Codec sofia/internal/1000@freeswitch.server PCMU/8000 20 ms 160 samples 64000 bits 1 channels 2017-09-23 15:08:52.917044 [DEBUG] switch_core_codec.c:111 sofia/internal/1000@freeswitch.server Original read codec set to PCMU:0 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:4695 Set telephone-event payload to 101@8000 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:4754 sofia/internal/1000@freeswitch.server Set 2833 dtmf send payload to 101 recv payload to 101 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:6865 AUDIO RTP [sofia/internal/1000@freeswitch.server] 72.14.x.x port 23668 -> 71.238.x.x port 5062 codec: 0 ms: 20 2017-09-23 15:08:52.917044 [DEBUG] switch_rtp.c:4096 Starting timer [soft] 160 bytes per 20ms 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:7166 sofia/internal/1000@freeswitch.server Set 2833 dtmf send payload to 101 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:7173 sofia/internal/1000@freeswitch.server Set 2833 dtmf receive payload to 101 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:7196 sofia/internal/1000@freeswitch.server Set rtp dtmf delay to 40 2017-09-23 15:08:52.917044 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/1000@freeswitch.server! 2017-09-23 15:08:52.917044 [DEBUG] switch_channel.c:3473 (sofia/internal/1000@freeswitch.server) Callstate Change RINGING -> EARLY 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:6848 Audio params are unchanged for sofia/internal/1000@freeswitch.server. Chris On Thu, Sep 21, 2017 at 09:50:13AM -0500, Chris Nestrud wrote:
I haven't had a chance to dig into this further. That sounds like a logical place to begin investigating. Thanks.
Chris
On Thu, Sep 21, 2017 at 02:29:25PM +0000, Scott Granados wrote:
Sounds like it could be a problem in Jitter buffer management. What do you think? Have you tried adjusting the type and size of buffer negotiated by the PBX?
On Sep 21, 2017, at 8:56 AM, Chris Nestrud <ccn@chrisnestrud.com> wrote:
Express Talk has a strange issue for me where several seconds of delay will build up after about an hour of a SIP call being connected. I haven't had this delay issue when using Bria on iOS. Express Talk is very accessible. Let me know if you try it and have the delay issue.
Chris
On Tue, Sep 19, 2017 at 08:21:34PM -0500, Chris Nestrud wrote:
Express Talk from NCH looks accessible.
http://www.nch.com.au/talk/index.html
The command-line version of Linphone may also be worth a look; the Windows GUI version is not accessible.
Please let me know if you find anything better.
Chris
On Tue, Sep 19, 2017 at 04:30:22PM -0400, Katherine M. Moss wrote:
Hello all,
I forget what the concensus was when we discussed this before, but I seem to be finding difficulties finding a softphone that has both accessibility as well as a good feature set; particularly, I need the ability to pull contacts and what not from Active Directory since we use that; I've heard of Microsip, though I know that it does not support this feature. If X-Lite/Bria used to be accessible, they aren't now; unlabeled fields everywhere which cannot be seemingly deciphered via object navigation with NVDA (Haven't tried JAWS, but I will.) We're using Asterisk for the server if that matters. Thanks.
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I haven't had a chance to dig into this further. That sounds like a logical place to begin investigating. Thanks.
Chris
On Thu, Sep 21, 2017 at 02:29:25PM +0000, Scott Granados wrote:
Sounds like it could be a problem in Jitter buffer management. What do you think? Have you tried adjusting the type and size of buffer negotiated by the PBX?
On Sep 21, 2017, at 8:56 AM, Chris Nestrud <ccn@chrisnestrud.com> wrote:
Express Talk has a strange issue for me where several seconds of delay will build up after about an hour of a SIP call being connected. I haven't had this delay issue when using Bria on iOS. Express Talk is very accessible. Let me know if you try it and have
I'm reviving an old thread here. Are there any SIP clients mentioned in this discussion as offering good screen reader accessibility that support IPv6? -----Original Message----- From: Blind-sysadmins [mailto:blind-sysadmins-bounces@lists.hodgsonfamily.org] On Behalf Of Chris Nestrud Sent: Saturday, September 23, 2017 4:30 PM To: Blind sysadmins list <blind-sysadmins@lists.hodgsonfamily.org> Subject: Re: [Blind-sysadmins] softphone recommendations My recommendation changes from Express Talk to Phoner Lite. Phoner Lite has some unlabeled buttons and is not as accessible as Express Talk., but Phoner Lite does not have this strange delay issue. Using same hardware and two softphones with same codec, looks like it might be an Express Talk issue. Express Talk: Dial 9196, immediately start saying numbers quickly, stopping when numbers are heard, 1-12 is heard from echo. The echo application is working and sends back pretty much all data but with this 1.5 second delay. Free Switch log of Express Talk connection: 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:4352 Set telephone-event payload to 101@8000 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:3056 Set Codec sofia/internal/1000@freeswitch.server PCMU/8000 20 ms 160 samples 64000 bits 1 channels 2017-09-23 14:44:06.997035 [DEBUG] switch_core_codec.c:111 sofia/internal/1000@freeswitch.server Original read codec set to PCMU:0 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:4695 Set telephone-event payload to 101@8000 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:4754 sofia/internal/1000@freeswitch.server Set 2833 dtmf send payload to 101 recv payload to 101 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:6865 AUDIO RTP [sofia/internal/1000@freeswitch.server] 72.14.x.x port 29276 -> 71.238.x.x port 8000 codec: 0 ms: 20 2017-09-23 14:44:06.997035 [DEBUG] switch_rtp.c:4096 Starting timer [soft] 160 bytes per 20ms 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:7166 sofia/internal/1000@freeswitch.server Set 2833 dtmf send payload to 101 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:7173 sofia/internal/1000@freeswitch.server Set 2833 dtmf receive payload to 101 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:7196 sofia/internal/1000@freeswitch.server Set rtp dtmf delay to 40 2017-09-23 14:44:06.997035 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/1000@freeswitch.server! 2017-09-23 14:44:06.997035 [DEBUG] switch_channel.c:3473 (sofia/internal/1000@freeswitch.server) Callstate Change RINGING -> EARLY 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:6848 Audio params are unchanged for sofia/internal/1000@freeswitch.server. Phoner Lite: Same computer and audio device as Express Talk. I set the codec to G.711 u-Law, 64 kbps which matches Express Talk. It was originally set to Opus, 50kbps. Dial 9196, immediately start saying numbers quickly, stopping when numbers are heard, 2 is heard from echo. The echo application is working and sends back data with minimal delay. Free Switch log of Phoner Lite connection: 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:4352 Set telephone-event payload to 101@8000 2017-09-23 15:08:52.917044 [WARNING] switch_core_media.c:4606 Invalid SDP for opus. Don't ask.. but it needs a /2 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:3056 Set Codec sofia/internal/1000@freeswitch.server PCMU/8000 20 ms 160 samples 64000 bits 1 channels 2017-09-23 15:08:52.917044 [DEBUG] switch_core_codec.c:111 sofia/internal/1000@freeswitch.server Original read codec set to PCMU:0 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:4695 Set telephone-event payload to 101@8000 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:4754 sofia/internal/1000@freeswitch.server Set 2833 dtmf send payload to 101 recv payload to 101 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:6865 AUDIO RTP [sofia/internal/1000@freeswitch.server] 72.14.x.x port 23668 -> 71.238.x.x port 5062 codec: 0 ms: 20 2017-09-23 15:08:52.917044 [DEBUG] switch_rtp.c:4096 Starting timer [soft] 160 bytes per 20ms 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:7166 sofia/internal/1000@freeswitch.server Set 2833 dtmf send payload to 101 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:7173 sofia/internal/1000@freeswitch.server Set 2833 dtmf receive payload to 101 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:7196 sofia/internal/1000@freeswitch.server Set rtp dtmf delay to 40 2017-09-23 15:08:52.917044 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/1000@freeswitch.server! 2017-09-23 15:08:52.917044 [DEBUG] switch_channel.c:3473 (sofia/internal/1000@freeswitch.server) Callstate Change RINGING -> EARLY 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:6848 Audio params are unchanged for sofia/internal/1000@freeswitch.server. Chris On Thu, Sep 21, 2017 at 09:50:13AM -0500, Chris Nestrud wrote: the delay issue.
Chris
On Tue, Sep 19, 2017 at 08:21:34PM -0500, Chris Nestrud wrote:
Express Talk from NCH looks accessible.
http://www.nch.com.au/talk/index.html
The command-line version of Linphone may also be worth a look; the Windows GUI version is not accessible.
Please let me know if you find anything better.
Chris
On Tue, Sep 19, 2017 at 04:30:22PM -0400, Katherine M. Moss wrote:
Hello all,
I forget what the concensus was when we discussed this before, but I seem to be finding difficulties finding a softphone that has both accessibility as well as a good feature set; particularly, I need the ability to pull contacts and what not from Active Directory since we use that; I've heard of Microsip, though I know that it does not support this feature. If X-Lite/Bria used to be accessible, they aren't now; unlabeled fields everywhere which cannot be seemingly deciphered via object navigation with NVDA (Haven't tried JAWS, but I will.) We're
using Asterisk for the server if that matters. Thanks.
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Yeah, and also how about Opus at 48 kHz? Bill
On Nov 1, 2017, at 12:52 AM, Jason White via Blind-sysadmins <blind-sysadmins@lists.hodgsonfamily.org> wrote:
I'm reviving an old thread here. Are there any SIP clients mentioned in this discussion as offering good screen reader accessibility that support IPv6?
-----Original Message----- From: Blind-sysadmins [mailto:blind-sysadmins-bounces@lists.hodgsonfamily.org] On Behalf Of Chris Nestrud Sent: Saturday, September 23, 2017 4:30 PM To: Blind sysadmins list <blind-sysadmins@lists.hodgsonfamily.org> Subject: Re: [Blind-sysadmins] softphone recommendations
My recommendation changes from Express Talk to Phoner Lite. Phoner Lite has some unlabeled buttons and is not as accessible as Express Talk., but Phoner Lite does not have this strange delay issue.
Using same hardware and two softphones with same codec, looks like it might be an Express Talk issue.
Express Talk:
Dial 9196, immediately start saying numbers quickly, stopping when numbers are heard, 1-12 is heard from echo. The echo application is working and sends back pretty much all data but with this 1.5 second delay.
Free Switch log of Express Talk connection:
2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:4352 Set telephone-event payload to 101@8000 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:3056 Set Codec sofia/internal/1000@freeswitch.server PCMU/8000 20 ms 160 samples 64000 bits 1 channels 2017-09-23 14:44:06.997035 [DEBUG] switch_core_codec.c:111 sofia/internal/1000@freeswitch.server Original read codec set to PCMU:0 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:4695 Set telephone-event payload to 101@8000 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:4754 sofia/internal/1000@freeswitch.server Set 2833 dtmf send payload to 101 recv payload to 101 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:6865 AUDIO RTP [sofia/internal/1000@freeswitch.server] 72.14.x.x port 29276 -> 71.238.x.x port 8000 codec: 0 ms: 20 2017-09-23 14:44:06.997035 [DEBUG] switch_rtp.c:4096 Starting timer [soft] 160 bytes per 20ms 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:7166 sofia/internal/1000@freeswitch.server Set 2833 dtmf send payload to 101 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:7173 sofia/internal/1000@freeswitch.server Set 2833 dtmf receive payload to 101 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:7196 sofia/internal/1000@freeswitch.server Set rtp dtmf delay to 40 2017-09-23 14:44:06.997035 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/1000@freeswitch.server! 2017-09-23 14:44:06.997035 [DEBUG] switch_channel.c:3473 (sofia/internal/1000@freeswitch.server) Callstate Change RINGING -> EARLY 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:6848 Audio params are unchanged for sofia/internal/1000@freeswitch.server.
Phoner Lite:
Same computer and audio device as Express Talk. I set the codec to G.711 u-Law, 64 kbps which matches Express Talk. It was originally set to Opus, 50kbps.
Dial 9196, immediately start saying numbers quickly, stopping when numbers are heard, 2 is heard from echo. The echo application is working and sends back data with minimal delay.
Free Switch log of Phoner Lite connection:
2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:4352 Set telephone-event payload to 101@8000 2017-09-23 15:08:52.917044 [WARNING] switch_core_media.c:4606 Invalid SDP for opus. Don't ask.. but it needs a /2 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:3056 Set Codec sofia/internal/1000@freeswitch.server PCMU/8000 20 ms 160 samples 64000 bits 1 channels 2017-09-23 15:08:52.917044 [DEBUG] switch_core_codec.c:111 sofia/internal/1000@freeswitch.server Original read codec set to PCMU:0 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:4695 Set telephone-event payload to 101@8000 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:4754 sofia/internal/1000@freeswitch.server Set 2833 dtmf send payload to 101 recv payload to 101 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:6865 AUDIO RTP [sofia/internal/1000@freeswitch.server] 72.14.x.x port 23668 -> 71.238.x.x port 5062 codec: 0 ms: 20 2017-09-23 15:08:52.917044 [DEBUG] switch_rtp.c:4096 Starting timer [soft] 160 bytes per 20ms 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:7166 sofia/internal/1000@freeswitch.server Set 2833 dtmf send payload to 101 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:7173 sofia/internal/1000@freeswitch.server Set 2833 dtmf receive payload to 101 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:7196 sofia/internal/1000@freeswitch.server Set rtp dtmf delay to 40 2017-09-23 15:08:52.917044 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/1000@freeswitch.server! 2017-09-23 15:08:52.917044 [DEBUG] switch_channel.c:3473 (sofia/internal/1000@freeswitch.server) Callstate Change RINGING -> EARLY 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:6848 Audio params are unchanged for sofia/internal/1000@freeswitch.server.
Chris
I haven't had a chance to dig into this further. That sounds like a logical place to begin investigating. Thanks.
Chris
On Thu, Sep 21, 2017 at 02:29:25PM +0000, Scott Granados wrote:
Sounds like it could be a problem in Jitter buffer management. What do you think? Have you tried adjusting the type and size of buffer negotiated by the PBX?
On Sep 21, 2017, at 8:56 AM, Chris Nestrud <ccn@chrisnestrud.com> wrote:
Express Talk has a strange issue for me where several seconds of delay will build up after about an hour of a SIP call being connected. I haven't had this delay issue when using Bria on iOS. Express Talk is very accessible. Let me know if you try it and have
On Thu, Sep 21, 2017 at 09:50:13AM -0500, Chris Nestrud wrote: the delay issue.
Chris
On Tue, Sep 19, 2017 at 08:21:34PM -0500, Chris Nestrud wrote:
Express Talk from NCH looks accessible.
http://www.nch.com.au/talk/index.html
The command-line version of Linphone may also be worth a look; the Windows GUI version is not accessible.
Please let me know if you find anything better.
Chris
On Tue, Sep 19, 2017 at 04:30:22PM -0400, Katherine M. Moss wrote:
Hello all,
I forget what the concensus was when we discussed this before, but I seem to be finding difficulties finding a softphone that has both accessibility as well as a good feature set; particularly, I need the ability to pull contacts and what not from Active Directory since we use that; I've heard of Microsip, though I know that it does not support this feature. If X-Lite/Bria used to be accessible, they aren't now; unlabeled fields everywhere which cannot be seemingly deciphered via object navigation with NVDA (Haven't tried JAWS, but I will.) We're
using Asterisk for the server if that matters. Thanks.
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Hello, Unfortunately Microsip does not provide above features, but you can the following application, it has ability to use IPV6 support: http://www.linphone.org/ 2017-11-01 16:23 GMT+03:00, Bill Dengler <codeofdusk@gmail.com>:
Yeah, and also how about Opus at 48 kHz?
Bill
On Nov 1, 2017, at 12:52 AM, Jason White via Blind-sysadmins <blind-sysadmins@lists.hodgsonfamily.org> wrote:
I'm reviving an old thread here. Are there any SIP clients mentioned in this discussion as offering good screen reader accessibility that support IPv6?
-----Original Message----- From: Blind-sysadmins [mailto:blind-sysadmins-bounces@lists.hodgsonfamily.org] On Behalf Of Chris Nestrud Sent: Saturday, September 23, 2017 4:30 PM To: Blind sysadmins list <blind-sysadmins@lists.hodgsonfamily.org> Subject: Re: [Blind-sysadmins] softphone recommendations
My recommendation changes from Express Talk to Phoner Lite. Phoner Lite has some unlabeled buttons and is not as accessible as Express Talk., but Phoner Lite does not have this strange delay issue.
Using same hardware and two softphones with same codec, looks like it might be an Express Talk issue.
Express Talk:
Dial 9196, immediately start saying numbers quickly, stopping when numbers are heard, 1-12 is heard from echo. The echo application is working and sends back pretty much all data but with this 1.5 second delay.
Free Switch log of Express Talk connection:
2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:4352 Set telephone-event payload to 101@8000 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:3056 Set Codec sofia/internal/1000@freeswitch.server PCMU/8000 20 ms 160 samples 64000 bits 1 channels 2017-09-23 14:44:06.997035 [DEBUG] switch_core_codec.c:111 sofia/internal/1000@freeswitch.server Original read codec set to PCMU:0 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:4695 Set telephone-event payload to 101@8000 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:4754 sofia/internal/1000@freeswitch.server Set 2833 dtmf send payload to 101 recv payload to 101 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:6865 AUDIO RTP [sofia/internal/1000@freeswitch.server] 72.14.x.x port 29276 -> 71.238.x.x port 8000 codec: 0 ms: 20 2017-09-23 14:44:06.997035 [DEBUG] switch_rtp.c:4096 Starting timer [soft] 160 bytes per 20ms 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:7166 sofia/internal/1000@freeswitch.server Set 2833 dtmf send payload to 101 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:7173 sofia/internal/1000@freeswitch.server Set 2833 dtmf receive payload to 101 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:7196 sofia/internal/1000@freeswitch.server Set rtp dtmf delay to 40 2017-09-23 14:44:06.997035 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/1000@freeswitch.server! 2017-09-23 14:44:06.997035 [DEBUG] switch_channel.c:3473 (sofia/internal/1000@freeswitch.server) Callstate Change RINGING -> EARLY 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:6848 Audio params are unchanged for sofia/internal/1000@freeswitch.server.
Phoner Lite:
Same computer and audio device as Express Talk. I set the codec to G.711 u-Law, 64 kbps which matches Express Talk. It was originally set to Opus, 50kbps.
Dial 9196, immediately start saying numbers quickly, stopping when numbers are heard, 2 is heard from echo. The echo application is working and sends back data with minimal delay.
Free Switch log of Phoner Lite connection:
2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:4352 Set telephone-event payload to 101@8000 2017-09-23 15:08:52.917044 [WARNING] switch_core_media.c:4606 Invalid SDP for opus. Don't ask.. but it needs a /2 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:3056 Set Codec sofia/internal/1000@freeswitch.server PCMU/8000 20 ms 160 samples 64000 bits 1 channels 2017-09-23 15:08:52.917044 [DEBUG] switch_core_codec.c:111 sofia/internal/1000@freeswitch.server Original read codec set to PCMU:0 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:4695 Set telephone-event payload to 101@8000 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:4754 sofia/internal/1000@freeswitch.server Set 2833 dtmf send payload to 101 recv payload to 101 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:6865 AUDIO RTP [sofia/internal/1000@freeswitch.server] 72.14.x.x port 23668 -> 71.238.x.x port 5062 codec: 0 ms: 20 2017-09-23 15:08:52.917044 [DEBUG] switch_rtp.c:4096 Starting timer [soft] 160 bytes per 20ms 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:7166 sofia/internal/1000@freeswitch.server Set 2833 dtmf send payload to 101 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:7173 sofia/internal/1000@freeswitch.server Set 2833 dtmf receive payload to 101 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:7196 sofia/internal/1000@freeswitch.server Set rtp dtmf delay to 40 2017-09-23 15:08:52.917044 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/1000@freeswitch.server! 2017-09-23 15:08:52.917044 [DEBUG] switch_channel.c:3473 (sofia/internal/1000@freeswitch.server) Callstate Change RINGING -> EARLY 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:6848 Audio params are unchanged for sofia/internal/1000@freeswitch.server.
Chris
I haven't had a chance to dig into this further. That sounds like a logical place to begin investigating. Thanks.
Chris
On Thu, Sep 21, 2017 at 02:29:25PM +0000, Scott Granados wrote:
Sounds like it could be a problem in Jitter buffer management. What do you think? Have you tried adjusting the type and size of buffer negotiated by the PBX?
On Sep 21, 2017, at 8:56 AM, Chris Nestrud <ccn@chrisnestrud.com> wrote:
Express Talk has a strange issue for me where several seconds of delay will build up after about an hour of a SIP call being connected. I haven't had this delay issue when using Bria on iOS. Express Talk is very accessible. Let me know if you try it and have
On Thu, Sep 21, 2017 at 09:50:13AM -0500, Chris Nestrud wrote: the delay issue.
Chris
On Tue, Sep 19, 2017 at 08:21:34PM -0500, Chris Nestrud wrote:
Express Talk from NCH looks accessible.
http://www.nch.com.au/talk/index.html
The command-line version of Linphone may also be worth a look; the Windows GUI version is not accessible.
Please let me know if you find anything better.
Chris
On Tue, Sep 19, 2017 at 04:30:22PM -0400, Katherine M. Moss wrote: > Hello all, > > I forget what the concensus was when we discussed this before, > but I seem to be finding difficulties finding a softphone that > has both accessibility as well as a good feature set; > particularly, I need the ability to pull contacts and what not > from Active Directory since we use that; I've heard of Microsip, > though I know that it does not support this feature. If > X-Lite/Bria used to be accessible, they aren't now; unlabeled > fields everywhere which cannot be seemingly deciphered via > object navigation with NVDA (Haven't tried JAWS, but I will.) We're
using Asterisk for the server if that matters. Thanks.
> > _______________________________________________ > Blind-sysadmins mailing list > Blind-sysadmins@lists.hodgsonfamily.org > https://lists.hodgsonfamily.org/listinfo/blind-sysadmins
_______________________________________________ Blind-sysadmins mailing list Blind-sysadmins@lists.hodgsonfamily.org https://lists.hodgsonfamily.org/listinfo/blind-sysadmins
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-- Can Kırca
Hello, Unfortunately Microsip does not provide above features, but you can the following application, it has ability to use IPV6 support: http://www.linphone.org/ 2017-11-01 16:23 GMT+03:00, Bill Dengler <codeofdusk@gmail.com>:
Yeah, and also how about Opus at 48 kHz?
Bill
On Nov 1, 2017, at 12:52 AM, Jason White via Blind-sysadmins <blind-sysadmins@lists.hodgsonfamily.org> wrote:
I'm reviving an old thread here. Are there any SIP clients mentioned in this discussion as offering good screen reader accessibility that support IPv6?
-----Original Message----- From: Blind-sysadmins [mailto:blind-sysadmins-bounces@lists.hodgsonfamily.org] On Behalf Of Chris Nestrud Sent: Saturday, September 23, 2017 4:30 PM To: Blind sysadmins list <blind-sysadmins@lists.hodgsonfamily.org> Subject: Re: [Blind-sysadmins] softphone recommendations
My recommendation changes from Express Talk to Phoner Lite. Phoner Lite has some unlabeled buttons and is not as accessible as Express Talk., but Phoner Lite does not have this strange delay issue.
Using same hardware and two softphones with same codec, looks like it might be an Express Talk issue.
Express Talk:
Dial 9196, immediately start saying numbers quickly, stopping when numbers are heard, 1-12 is heard from echo. The echo application is working and sends back pretty much all data but with this 1.5 second delay.
Free Switch log of Express Talk connection:
2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:4352 Set telephone-event payload to 101@8000 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:3056 Set Codec sofia/internal/1000@freeswitch.server PCMU/8000 20 ms 160 samples 64000 bits 1 channels 2017-09-23 14:44:06.997035 [DEBUG] switch_core_codec.c:111 sofia/internal/1000@freeswitch.server Original read codec set to PCMU:0 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:4695 Set telephone-event payload to 101@8000 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:4754 sofia/internal/1000@freeswitch.server Set 2833 dtmf send payload to 101 recv payload to 101 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:6865 AUDIO RTP [sofia/internal/1000@freeswitch.server] 72.14.x.x port 29276 -> 71.238.x.x port 8000 codec: 0 ms: 20 2017-09-23 14:44:06.997035 [DEBUG] switch_rtp.c:4096 Starting timer [soft] 160 bytes per 20ms 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:7166 sofia/internal/1000@freeswitch.server Set 2833 dtmf send payload to 101 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:7173 sofia/internal/1000@freeswitch.server Set 2833 dtmf receive payload to 101 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:7196 sofia/internal/1000@freeswitch.server Set rtp dtmf delay to 40 2017-09-23 14:44:06.997035 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/1000@freeswitch.server! 2017-09-23 14:44:06.997035 [DEBUG] switch_channel.c:3473 (sofia/internal/1000@freeswitch.server) Callstate Change RINGING -> EARLY 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:6848 Audio params are unchanged for sofia/internal/1000@freeswitch.server.
Phoner Lite:
Same computer and audio device as Express Talk. I set the codec to G.711 u-Law, 64 kbps which matches Express Talk. It was originally set to Opus, 50kbps.
Dial 9196, immediately start saying numbers quickly, stopping when numbers are heard, 2 is heard from echo. The echo application is working and sends back data with minimal delay.
Free Switch log of Phoner Lite connection:
2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:4352 Set telephone-event payload to 101@8000 2017-09-23 15:08:52.917044 [WARNING] switch_core_media.c:4606 Invalid SDP for opus. Don't ask.. but it needs a /2 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:3056 Set Codec sofia/internal/1000@freeswitch.server PCMU/8000 20 ms 160 samples 64000 bits 1 channels 2017-09-23 15:08:52.917044 [DEBUG] switch_core_codec.c:111 sofia/internal/1000@freeswitch.server Original read codec set to PCMU:0 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:4695 Set telephone-event payload to 101@8000 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:4754 sofia/internal/1000@freeswitch.server Set 2833 dtmf send payload to 101 recv payload to 101 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:6865 AUDIO RTP [sofia/internal/1000@freeswitch.server] 72.14.x.x port 23668 -> 71.238.x.x port 5062 codec: 0 ms: 20 2017-09-23 15:08:52.917044 [DEBUG] switch_rtp.c:4096 Starting timer [soft] 160 bytes per 20ms 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:7166 sofia/internal/1000@freeswitch.server Set 2833 dtmf send payload to 101 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:7173 sofia/internal/1000@freeswitch.server Set 2833 dtmf receive payload to 101 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:7196 sofia/internal/1000@freeswitch.server Set rtp dtmf delay to 40 2017-09-23 15:08:52.917044 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/1000@freeswitch.server! 2017-09-23 15:08:52.917044 [DEBUG] switch_channel.c:3473 (sofia/internal/1000@freeswitch.server) Callstate Change RINGING -> EARLY 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:6848 Audio params are unchanged for sofia/internal/1000@freeswitch.server.
Chris
I haven't had a chance to dig into this further. That sounds like a logical place to begin investigating. Thanks.
Chris
On Thu, Sep 21, 2017 at 02:29:25PM +0000, Scott Granados wrote:
Sounds like it could be a problem in Jitter buffer management. What do you think? Have you tried adjusting the type and size of buffer negotiated by the PBX?
On Sep 21, 2017, at 8:56 AM, Chris Nestrud <ccn@chrisnestrud.com> wrote:
Express Talk has a strange issue for me where several seconds of delay will build up after about an hour of a SIP call being connected. I haven't had this delay issue when using Bria on iOS. Express Talk is very accessible. Let me know if you try it and have
On Thu, Sep 21, 2017 at 09:50:13AM -0500, Chris Nestrud wrote: the delay issue.
Chris
On Tue, Sep 19, 2017 at 08:21:34PM -0500, Chris Nestrud wrote:
Express Talk from NCH looks accessible.
http://www.nch.com.au/talk/index.html
The command-line version of Linphone may also be worth a look; the Windows GUI version is not accessible.
Please let me know if you find anything better.
Chris
On Tue, Sep 19, 2017 at 04:30:22PM -0400, Katherine M. Moss wrote: > Hello all, > > I forget what the concensus was when we discussed this before, > but I seem to be finding difficulties finding a softphone that > has both accessibility as well as a good feature set; > particularly, I need the ability to pull contacts and what not > from Active Directory since we use that; I've heard of Microsip, > though I know that it does not support this feature. If > X-Lite/Bria used to be accessible, they aren't now; unlabeled > fields everywhere which cannot be seemingly deciphered via > object navigation with NVDA (Haven't tried JAWS, but I will.) We're
using Asterisk for the server if that matters. Thanks.
> > _______________________________________________ > Blind-sysadmins mailing list > Blind-sysadmins@lists.hodgsonfamily.org > https://lists.hodgsonfamily.org/listinfo/blind-sysadmins
_______________________________________________ Blind-sysadmins mailing list Blind-sysadmins@lists.hodgsonfamily.org https://lists.hodgsonfamily.org/listinfo/blind-sysadmins
_______________________________________________ Blind-sysadmins mailing list Blind-sysadmins@lists.hodgsonfamily.org https://lists.hodgsonfamily.org/listinfo/blind-sysadmins
_______________________________________________ Blind-sysadmins mailing list Blind-sysadmins@lists.hodgsonfamily.org https://lists.hodgsonfamily.org/listinfo/blind-sysadmins
_______________________________________________ Blind-sysadmins mailing list Blind-sysadmins@lists.hodgsonfamily.org https://lists.hodgsonfamily.org/listinfo/blind-sysadmins
_______________________________________________ Blind-sysadmins mailing list Blind-sysadmins@lists.hodgsonfamily.org https://lists.hodgsonfamily.org/listinfo/blind-sysadmins
-- Can Kırca
Phoner Lite may be able to do this. I know that it supports Opus. www.phonerlite.de/index_en.htm I did not find the Linphone Windows GUI to be accessible. Chris On Wed, Nov 01, 2017 at 04:45:43PM +0300, Can K??rca wrote:
Hello,
Unfortunately Microsip does not provide above features, but you can the following application, it has ability to use IPV6 support: http://www.linphone.org/
2017-11-01 16:23 GMT+03:00, Bill Dengler <codeofdusk@gmail.com>:
Yeah, and also how about Opus at 48 kHz?
Bill
On Nov 1, 2017, at 12:52 AM, Jason White via Blind-sysadmins <blind-sysadmins@lists.hodgsonfamily.org> wrote:
I'm reviving an old thread here. Are there any SIP clients mentioned in this discussion as offering good screen reader accessibility that support IPv6?
-----Original Message----- From: Blind-sysadmins [mailto:blind-sysadmins-bounces@lists.hodgsonfamily.org] On Behalf Of Chris Nestrud Sent: Saturday, September 23, 2017 4:30 PM To: Blind sysadmins list <blind-sysadmins@lists.hodgsonfamily.org> Subject: Re: [Blind-sysadmins] softphone recommendations
My recommendation changes from Express Talk to Phoner Lite. Phoner Lite has some unlabeled buttons and is not as accessible as Express Talk., but Phoner Lite does not have this strange delay issue.
Using same hardware and two softphones with same codec, looks like it might be an Express Talk issue.
Express Talk:
Dial 9196, immediately start saying numbers quickly, stopping when numbers are heard, 1-12 is heard from echo. The echo application is working and sends back pretty much all data but with this 1.5 second delay.
Free Switch log of Express Talk connection:
2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:4352 Set telephone-event payload to 101@8000 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:3056 Set Codec sofia/internal/1000@freeswitch.server PCMU/8000 20 ms 160 samples 64000 bits 1 channels 2017-09-23 14:44:06.997035 [DEBUG] switch_core_codec.c:111 sofia/internal/1000@freeswitch.server Original read codec set to PCMU:0 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:4695 Set telephone-event payload to 101@8000 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:4754 sofia/internal/1000@freeswitch.server Set 2833 dtmf send payload to 101 recv payload to 101 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:6865 AUDIO RTP [sofia/internal/1000@freeswitch.server] 72.14.x.x port 29276 -> 71.238.x.x port 8000 codec: 0 ms: 20 2017-09-23 14:44:06.997035 [DEBUG] switch_rtp.c:4096 Starting timer [soft] 160 bytes per 20ms 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:7166 sofia/internal/1000@freeswitch.server Set 2833 dtmf send payload to 101 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:7173 sofia/internal/1000@freeswitch.server Set 2833 dtmf receive payload to 101 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:7196 sofia/internal/1000@freeswitch.server Set rtp dtmf delay to 40 2017-09-23 14:44:06.997035 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/1000@freeswitch.server! 2017-09-23 14:44:06.997035 [DEBUG] switch_channel.c:3473 (sofia/internal/1000@freeswitch.server) Callstate Change RINGING -> EARLY 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:6848 Audio params are unchanged for sofia/internal/1000@freeswitch.server.
Phoner Lite:
Same computer and audio device as Express Talk. I set the codec to G.711 u-Law, 64 kbps which matches Express Talk. It was originally set to Opus, 50kbps.
Dial 9196, immediately start saying numbers quickly, stopping when numbers are heard, 2 is heard from echo. The echo application is working and sends back data with minimal delay.
Free Switch log of Phoner Lite connection:
2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:4352 Set telephone-event payload to 101@8000 2017-09-23 15:08:52.917044 [WARNING] switch_core_media.c:4606 Invalid SDP for opus. Don't ask.. but it needs a /2 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:3056 Set Codec sofia/internal/1000@freeswitch.server PCMU/8000 20 ms 160 samples 64000 bits 1 channels 2017-09-23 15:08:52.917044 [DEBUG] switch_core_codec.c:111 sofia/internal/1000@freeswitch.server Original read codec set to PCMU:0 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:4695 Set telephone-event payload to 101@8000 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:4754 sofia/internal/1000@freeswitch.server Set 2833 dtmf send payload to 101 recv payload to 101 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:6865 AUDIO RTP [sofia/internal/1000@freeswitch.server] 72.14.x.x port 23668 -> 71.238.x.x port 5062 codec: 0 ms: 20 2017-09-23 15:08:52.917044 [DEBUG] switch_rtp.c:4096 Starting timer [soft] 160 bytes per 20ms 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:7166 sofia/internal/1000@freeswitch.server Set 2833 dtmf send payload to 101 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:7173 sofia/internal/1000@freeswitch.server Set 2833 dtmf receive payload to 101 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:7196 sofia/internal/1000@freeswitch.server Set rtp dtmf delay to 40 2017-09-23 15:08:52.917044 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/1000@freeswitch.server! 2017-09-23 15:08:52.917044 [DEBUG] switch_channel.c:3473 (sofia/internal/1000@freeswitch.server) Callstate Change RINGING -> EARLY 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:6848 Audio params are unchanged for sofia/internal/1000@freeswitch.server.
Chris
I haven't had a chance to dig into this further. That sounds like a logical place to begin investigating. Thanks.
Chris
On Thu, Sep 21, 2017 at 02:29:25PM +0000, Scott Granados wrote:
Sounds like it could be a problem in Jitter buffer management. What do you think? Have you tried adjusting the type and size of buffer negotiated by the PBX?
On Sep 21, 2017, at 8:56 AM, Chris Nestrud <ccn@chrisnestrud.com> wrote:
Express Talk has a strange issue for me where several seconds of delay will build up after about an hour of a SIP call being connected. I haven't had this delay issue when using Bria on iOS. Express Talk is very accessible. Let me know if you try it and have
On Thu, Sep 21, 2017 at 09:50:13AM -0500, Chris Nestrud wrote: the delay issue.
Chris
On Tue, Sep 19, 2017 at 08:21:34PM -0500, Chris Nestrud wrote: > Express Talk from NCH looks accessible. > > http://www.nch.com.au/talk/index.html > > The command-line version of Linphone may also be worth a look; > the Windows GUI version is not accessible. > > http://www.linphone.org/ > > Please let me know if you find anything better. > > Chris > > On Tue, Sep 19, 2017 at 04:30:22PM -0400, Katherine M. Moss wrote: >> Hello all, >> >> I forget what the concensus was when we discussed this before, >> but I seem to be finding difficulties finding a softphone that >> has both accessibility as well as a good feature set; >> particularly, I need the ability to pull contacts and what not >> from Active Directory since we use that; I've heard of Microsip, >> though I know that it does not support this feature. If >> X-Lite/Bria used to be accessible, they aren't now; unlabeled >> fields everywhere which cannot be seemingly deciphered via >> object navigation with NVDA (Haven't tried JAWS, but I will.) We're
using Asterisk for the server if that matters. Thanks.
>> >> _______________________________________________ >> Blind-sysadmins mailing list >> Blind-sysadmins@lists.hodgsonfamily.org >> https://lists.hodgsonfamily.org/listinfo/blind-sysadmins
_______________________________________________ Blind-sysadmins mailing list Blind-sysadmins@lists.hodgsonfamily.org https://lists.hodgsonfamily.org/listinfo/blind-sysadmins
_______________________________________________ Blind-sysadmins mailing list Blind-sysadmins@lists.hodgsonfamily.org https://lists.hodgsonfamily.org/listinfo/blind-sysadmins
_______________________________________________ Blind-sysadmins mailing list Blind-sysadmins@lists.hodgsonfamily.org https://lists.hodgsonfamily.org/listinfo/blind-sysadmins
_______________________________________________ Blind-sysadmins mailing list Blind-sysadmins@lists.hodgsonfamily.org https://lists.hodgsonfamily.org/listinfo/blind-sysadmins
_______________________________________________ Blind-sysadmins mailing list Blind-sysadmins@lists.hodgsonfamily.org https://lists.hodgsonfamily.org/listinfo/blind-sysadmins
-- Can K??rca
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Yes but does it do 48 k opus? Bill
On Nov 1, 2017, at 3:05 PM, Chris Nestrud <ccn@chrisnestrud.com> wrote:
Phoner Lite may be able to do this. I know that it supports Opus. www.phonerlite.de/index_en.htm <http://www.phonerlite.de/index_en.htm>
I did not find the Linphone Windows GUI to be accessible.
Chris
On Wed, Nov 01, 2017 at 04:45:43PM +0300, Can K??rca wrote:
Hello,
Unfortunately Microsip does not provide above features, but you can the following application, it has ability to use IPV6 support: http://www.linphone.org/
2017-11-01 16:23 GMT+03:00, Bill Dengler <codeofdusk@gmail.com>:
Yeah, and also how about Opus at 48 kHz?
Bill
On Nov 1, 2017, at 12:52 AM, Jason White via Blind-sysadmins <blind-sysadmins@lists.hodgsonfamily.org> wrote:
I'm reviving an old thread here. Are there any SIP clients mentioned in this discussion as offering good screen reader accessibility that support IPv6?
-----Original Message----- From: Blind-sysadmins [mailto:blind-sysadmins-bounces@lists.hodgsonfamily.org] On Behalf Of Chris Nestrud Sent: Saturday, September 23, 2017 4:30 PM To: Blind sysadmins list <blind-sysadmins@lists.hodgsonfamily.org> Subject: Re: [Blind-sysadmins] softphone recommendations
My recommendation changes from Express Talk to Phoner Lite. Phoner Lite has some unlabeled buttons and is not as accessible as Express Talk., but Phoner Lite does not have this strange delay issue.
Using same hardware and two softphones with same codec, looks like it might be an Express Talk issue.
Express Talk:
Dial 9196, immediately start saying numbers quickly, stopping when numbers are heard, 1-12 is heard from echo. The echo application is working and sends back pretty much all data but with this 1.5 second delay.
Free Switch log of Express Talk connection:
2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:4352 Set telephone-event payload to 101@8000 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:3056 Set Codec sofia/internal/1000@freeswitch.server PCMU/8000 20 ms 160 samples 64000 bits 1 channels 2017-09-23 14:44:06.997035 [DEBUG] switch_core_codec.c:111 sofia/internal/1000@freeswitch.server Original read codec set to PCMU:0 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:4695 Set telephone-event payload to 101@8000 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:4754 sofia/internal/1000@freeswitch.server Set 2833 dtmf send payload to 101 recv payload to 101 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:6865 AUDIO RTP [sofia/internal/1000@freeswitch.server] 72.14.x.x port 29276 -> 71.238.x.x port 8000 codec: 0 ms: 20 2017-09-23 14:44:06.997035 [DEBUG] switch_rtp.c:4096 Starting timer [soft] 160 bytes per 20ms 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:7166 sofia/internal/1000@freeswitch.server Set 2833 dtmf send payload to 101 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:7173 sofia/internal/1000@freeswitch.server Set 2833 dtmf receive payload to 101 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:7196 sofia/internal/1000@freeswitch.server Set rtp dtmf delay to 40 2017-09-23 14:44:06.997035 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/1000@freeswitch.server! 2017-09-23 14:44:06.997035 [DEBUG] switch_channel.c:3473 (sofia/internal/1000@freeswitch.server) Callstate Change RINGING -> EARLY 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:6848 Audio params are unchanged for sofia/internal/1000@freeswitch.server.
Phoner Lite:
Same computer and audio device as Express Talk. I set the codec to G.711 u-Law, 64 kbps which matches Express Talk. It was originally set to Opus, 50kbps.
Dial 9196, immediately start saying numbers quickly, stopping when numbers are heard, 2 is heard from echo. The echo application is working and sends back data with minimal delay.
Free Switch log of Phoner Lite connection:
2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:4352 Set telephone-event payload to 101@8000 2017-09-23 15:08:52.917044 [WARNING] switch_core_media.c:4606 Invalid SDP for opus. Don't ask.. but it needs a /2 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:3056 Set Codec sofia/internal/1000@freeswitch.server PCMU/8000 20 ms 160 samples 64000 bits 1 channels 2017-09-23 15:08:52.917044 [DEBUG] switch_core_codec.c:111 sofia/internal/1000@freeswitch.server Original read codec set to PCMU:0 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:4695 Set telephone-event payload to 101@8000 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:4754 sofia/internal/1000@freeswitch.server Set 2833 dtmf send payload to 101 recv payload to 101 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:6865 AUDIO RTP [sofia/internal/1000@freeswitch.server] 72.14.x.x port 23668 -> 71.238.x.x port 5062 codec: 0 ms: 20 2017-09-23 15:08:52.917044 [DEBUG] switch_rtp.c:4096 Starting timer [soft] 160 bytes per 20ms 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:7166 sofia/internal/1000@freeswitch.server Set 2833 dtmf send payload to 101 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:7173 sofia/internal/1000@freeswitch.server Set 2833 dtmf receive payload to 101 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:7196 sofia/internal/1000@freeswitch.server Set rtp dtmf delay to 40 2017-09-23 15:08:52.917044 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/1000@freeswitch.server! 2017-09-23 15:08:52.917044 [DEBUG] switch_channel.c:3473 (sofia/internal/1000@freeswitch.server) Callstate Change RINGING -> EARLY 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:6848 Audio params are unchanged for sofia/internal/1000@freeswitch.server.
Chris
I haven't had a chance to dig into this further. That sounds like a logical place to begin investigating. Thanks.
Chris
On Thu, Sep 21, 2017 at 02:29:25PM +0000, Scott Granados wrote:
Sounds like it could be a problem in Jitter buffer management. What do you think? Have you tried adjusting the type and size of buffer negotiated by the PBX?
> On Sep 21, 2017, at 8:56 AM, Chris Nestrud <ccn@chrisnestrud.com> wrote: > > Express Talk has a strange issue for me where several seconds of > delay will build up after about an hour of a SIP call being > connected. I haven't had this delay issue when using Bria on iOS. > Express Talk is very accessible. Let me know if you try it and have
On Thu, Sep 21, 2017 at 09:50:13AM -0500, Chris Nestrud wrote: the delay issue.
> > Chris > > On Tue, Sep 19, 2017 at 08:21:34PM -0500, Chris Nestrud wrote: >> Express Talk from NCH looks accessible. >> >> http://www.nch.com.au/talk/index.html >> >> The command-line version of Linphone may also be worth a look; >> the Windows GUI version is not accessible. >> >> http://www.linphone.org/ >> >> Please let me know if you find anything better. >> >> Chris >> >> On Tue, Sep 19, 2017 at 04:30:22PM -0400, Katherine M. Moss wrote: >>> Hello all, >>> >>> I forget what the concensus was when we discussed this before, >>> but I seem to be finding difficulties finding a softphone that >>> has both accessibility as well as a good feature set; >>> particularly, I need the ability to pull contacts and what not >>> from Active Directory since we use that; I've heard of Microsip, >>> though I know that it does not support this feature. If >>> X-Lite/Bria used to be accessible, they aren't now; unlabeled >>> fields everywhere which cannot be seemingly deciphered via >>> object navigation with NVDA (Haven't tried JAWS, but I will.) We're using Asterisk for the server if that matters. Thanks. >>> >>> _______________________________________________ >>> Blind-sysadmins mailing list >>> Blind-sysadmins@lists.hodgsonfamily.org >>> https://lists.hodgsonfamily.org/listinfo/blind-sysadmins > > _______________________________________________ > Blind-sysadmins mailing list > Blind-sysadmins@lists.hodgsonfamily.org > https://lists.hodgsonfamily.org/listinfo/blind-sysadmins
_______________________________________________ Blind-sysadmins mailing list Blind-sysadmins@lists.hodgsonfamily.org https://lists.hodgsonfamily.org/listinfo/blind-sysadmins
_______________________________________________ Blind-sysadmins mailing list Blind-sysadmins@lists.hodgsonfamily.org https://lists.hodgsonfamily.org/listinfo/blind-sysadmins
_______________________________________________ Blind-sysadmins mailing list Blind-sysadmins@lists.hodgsonfamily.org https://lists.hodgsonfamily.org/listinfo/blind-sysadmins
_______________________________________________ Blind-sysadmins mailing list Blind-sysadmins@lists.hodgsonfamily.org https://lists.hodgsonfamily.org/listinfo/blind-sysadmins
-- Can K??rca
_______________________________________________ Blind-sysadmins mailing list Blind-sysadmins@lists.hodgsonfamily.org <mailto:Blind-sysadmins@lists.hodgsonfamily.org> https://lists.hodgsonfamily.org/listinfo/blind-sysadmins <https://lists.hodgsonfamily.org/listinfo/blind-sysadmins>
_______________________________________________ Blind-sysadmins mailing list Blind-sysadmins@lists.hodgsonfamily.org <mailto:Blind-sysadmins@lists.hodgsonfamily.org> https://lists.hodgsonfamily.org/listinfo/blind-sysadmins <https://lists.hodgsonfamily.org/listinfo/blind-sysadmins>
I believe so, but I don't have it at the moment to test. CODEC information is here: http://www.phonerlite.de/codecs_en.htm Chris On Wed, Nov 01, 2017 at 03:07:28PM +0000, Bill Dengler wrote:
Yes but does it do 48 k opus?
Bill
On Nov 1, 2017, at 3:05 PM, Chris Nestrud <ccn@chrisnestrud.com> wrote:
Phoner Lite may be able to do this. I know that it supports Opus. www.phonerlite.de/index_en.htm <http://www.phonerlite.de/index_en.htm>
I did not find the Linphone Windows GUI to be accessible.
Chris
On Wed, Nov 01, 2017 at 04:45:43PM +0300, Can K??rca wrote:
Hello,
Unfortunately Microsip does not provide above features, but you can the following application, it has ability to use IPV6 support: http://www.linphone.org/
2017-11-01 16:23 GMT+03:00, Bill Dengler <codeofdusk@gmail.com>:
Yeah, and also how about Opus at 48 kHz?
Bill
On Nov 1, 2017, at 12:52 AM, Jason White via Blind-sysadmins <blind-sysadmins@lists.hodgsonfamily.org> wrote:
I'm reviving an old thread here. Are there any SIP clients mentioned in this discussion as offering good screen reader accessibility that support IPv6?
-----Original Message----- From: Blind-sysadmins [mailto:blind-sysadmins-bounces@lists.hodgsonfamily.org] On Behalf Of Chris Nestrud Sent: Saturday, September 23, 2017 4:30 PM To: Blind sysadmins list <blind-sysadmins@lists.hodgsonfamily.org> Subject: Re: [Blind-sysadmins] softphone recommendations
My recommendation changes from Express Talk to Phoner Lite. Phoner Lite has some unlabeled buttons and is not as accessible as Express Talk., but Phoner Lite does not have this strange delay issue.
Using same hardware and two softphones with same codec, looks like it might be an Express Talk issue.
Express Talk:
Dial 9196, immediately start saying numbers quickly, stopping when numbers are heard, 1-12 is heard from echo. The echo application is working and sends back pretty much all data but with this 1.5 second delay.
Free Switch log of Express Talk connection:
2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:4352 Set telephone-event payload to 101@8000 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:3056 Set Codec sofia/internal/1000@freeswitch.server PCMU/8000 20 ms 160 samples 64000 bits 1 channels 2017-09-23 14:44:06.997035 [DEBUG] switch_core_codec.c:111 sofia/internal/1000@freeswitch.server Original read codec set to PCMU:0 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:4695 Set telephone-event payload to 101@8000 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:4754 sofia/internal/1000@freeswitch.server Set 2833 dtmf send payload to 101 recv payload to 101 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:6865 AUDIO RTP [sofia/internal/1000@freeswitch.server] 72.14.x.x port 29276 -> 71.238.x.x port 8000 codec: 0 ms: 20 2017-09-23 14:44:06.997035 [DEBUG] switch_rtp.c:4096 Starting timer [soft] 160 bytes per 20ms 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:7166 sofia/internal/1000@freeswitch.server Set 2833 dtmf send payload to 101 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:7173 sofia/internal/1000@freeswitch.server Set 2833 dtmf receive payload to 101 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:7196 sofia/internal/1000@freeswitch.server Set rtp dtmf delay to 40 2017-09-23 14:44:06.997035 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/1000@freeswitch.server! 2017-09-23 14:44:06.997035 [DEBUG] switch_channel.c:3473 (sofia/internal/1000@freeswitch.server) Callstate Change RINGING -> EARLY 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:6848 Audio params are unchanged for sofia/internal/1000@freeswitch.server.
Phoner Lite:
Same computer and audio device as Express Talk. I set the codec to G.711 u-Law, 64 kbps which matches Express Talk. It was originally set to Opus, 50kbps.
Dial 9196, immediately start saying numbers quickly, stopping when numbers are heard, 2 is heard from echo. The echo application is working and sends back data with minimal delay.
Free Switch log of Phoner Lite connection:
2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:4352 Set telephone-event payload to 101@8000 2017-09-23 15:08:52.917044 [WARNING] switch_core_media.c:4606 Invalid SDP for opus. Don't ask.. but it needs a /2 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:3056 Set Codec sofia/internal/1000@freeswitch.server PCMU/8000 20 ms 160 samples 64000 bits 1 channels 2017-09-23 15:08:52.917044 [DEBUG] switch_core_codec.c:111 sofia/internal/1000@freeswitch.server Original read codec set to PCMU:0 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:4695 Set telephone-event payload to 101@8000 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:4754 sofia/internal/1000@freeswitch.server Set 2833 dtmf send payload to 101 recv payload to 101 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:6865 AUDIO RTP [sofia/internal/1000@freeswitch.server] 72.14.x.x port 23668 -> 71.238.x.x port 5062 codec: 0 ms: 20 2017-09-23 15:08:52.917044 [DEBUG] switch_rtp.c:4096 Starting timer [soft] 160 bytes per 20ms 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:7166 sofia/internal/1000@freeswitch.server Set 2833 dtmf send payload to 101 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:7173 sofia/internal/1000@freeswitch.server Set 2833 dtmf receive payload to 101 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:7196 sofia/internal/1000@freeswitch.server Set rtp dtmf delay to 40 2017-09-23 15:08:52.917044 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/1000@freeswitch.server! 2017-09-23 15:08:52.917044 [DEBUG] switch_channel.c:3473 (sofia/internal/1000@freeswitch.server) Callstate Change RINGING -> EARLY 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:6848 Audio params are unchanged for sofia/internal/1000@freeswitch.server.
Chris
I haven't had a chance to dig into this further. That sounds like a logical place to begin investigating. Thanks.
Chris
On Thu, Sep 21, 2017 at 02:29:25PM +0000, Scott Granados wrote: > Sounds like it could be a problem in Jitter buffer management. What do you think? Have you tried adjusting the type and size of buffer negotiated by the PBX? > >> On Sep 21, 2017, at 8:56 AM, Chris Nestrud <ccn@chrisnestrud.com> wrote: >> >> Express Talk has a strange issue for me where several seconds of >> delay will build up after about an hour of a SIP call being >> connected. I haven't had this delay issue when using Bria on iOS. >> Express Talk is very accessible. Let me know if you try it and have
On Thu, Sep 21, 2017 at 09:50:13AM -0500, Chris Nestrud wrote: the delay issue.
>> >> Chris >> >> On Tue, Sep 19, 2017 at 08:21:34PM -0500, Chris Nestrud wrote: >>> Express Talk from NCH looks accessible. >>> >>> http://www.nch.com.au/talk/index.html >>> >>> The command-line version of Linphone may also be worth a look; >>> the Windows GUI version is not accessible. >>> >>> http://www.linphone.org/ >>> >>> Please let me know if you find anything better. >>> >>> Chris >>> >>> On Tue, Sep 19, 2017 at 04:30:22PM -0400, Katherine M. Moss wrote: >>>> Hello all, >>>> >>>> I forget what the concensus was when we discussed this before, >>>> but I seem to be finding difficulties finding a softphone that >>>> has both accessibility as well as a good feature set; >>>> particularly, I need the ability to pull contacts and what not >>>> from Active Directory since we use that; I've heard of Microsip, >>>> though I know that it does not support this feature. If >>>> X-Lite/Bria used to be accessible, they aren't now; unlabeled >>>> fields everywhere which cannot be seemingly deciphered via >>>> object navigation with NVDA (Haven't tried JAWS, but I will.) We're using Asterisk for the server if that matters. Thanks. >>>> >>>> _______________________________________________ >>>> Blind-sysadmins mailing list >>>> Blind-sysadmins@lists.hodgsonfamily.org >>>> https://lists.hodgsonfamily.org/listinfo/blind-sysadmins >> >> _______________________________________________ >> Blind-sysadmins mailing list >> Blind-sysadmins@lists.hodgsonfamily.org >> https://lists.hodgsonfamily.org/listinfo/blind-sysadmins > > > _______________________________________________ > Blind-sysadmins mailing list > Blind-sysadmins@lists.hodgsonfamily.org > https://lists.hodgsonfamily.org/listinfo/blind-sysadmins
_______________________________________________ Blind-sysadmins mailing list Blind-sysadmins@lists.hodgsonfamily.org https://lists.hodgsonfamily.org/listinfo/blind-sysadmins
_______________________________________________ Blind-sysadmins mailing list Blind-sysadmins@lists.hodgsonfamily.org https://lists.hodgsonfamily.org/listinfo/blind-sysadmins
_______________________________________________ Blind-sysadmins mailing list Blind-sysadmins@lists.hodgsonfamily.org https://lists.hodgsonfamily.org/listinfo/blind-sysadmins
-- Can K??rca
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That’s a tough one. IPv6 isn’t that common with SIP products yet. We run in to this a lot on our corporate systems. I use XLite and Bria but I don’t think these have the specific features you need.
On Oct 31, 2017, at 8:52 PM, Jason White via Blind-sysadmins <blind-sysadmins@lists.hodgsonfamily.org> wrote:
I'm reviving an old thread here. Are there any SIP clients mentioned in this discussion as offering good screen reader accessibility that support IPv6?
-----Original Message----- From: Blind-sysadmins [mailto:blind-sysadmins-bounces@lists.hodgsonfamily.org] On Behalf Of Chris Nestrud Sent: Saturday, September 23, 2017 4:30 PM To: Blind sysadmins list <blind-sysadmins@lists.hodgsonfamily.org> Subject: Re: [Blind-sysadmins] softphone recommendations
My recommendation changes from Express Talk to Phoner Lite. Phoner Lite has some unlabeled buttons and is not as accessible as Express Talk., but Phoner Lite does not have this strange delay issue.
Using same hardware and two softphones with same codec, looks like it might be an Express Talk issue.
Express Talk:
Dial 9196, immediately start saying numbers quickly, stopping when numbers are heard, 1-12 is heard from echo. The echo application is working and sends back pretty much all data but with this 1.5 second delay.
Free Switch log of Express Talk connection:
2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:4352 Set telephone-event payload to 101@8000 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:3056 Set Codec sofia/internal/1000@freeswitch.server PCMU/8000 20 ms 160 samples 64000 bits 1 channels 2017-09-23 14:44:06.997035 [DEBUG] switch_core_codec.c:111 sofia/internal/1000@freeswitch.server Original read codec set to PCMU:0 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:4695 Set telephone-event payload to 101@8000 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:4754 sofia/internal/1000@freeswitch.server Set 2833 dtmf send payload to 101 recv payload to 101 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:6865 AUDIO RTP [sofia/internal/1000@freeswitch.server] 72.14.x.x port 29276 -> 71.238.x.x port 8000 codec: 0 ms: 20 2017-09-23 14:44:06.997035 [DEBUG] switch_rtp.c:4096 Starting timer [soft] 160 bytes per 20ms 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:7166 sofia/internal/1000@freeswitch.server Set 2833 dtmf send payload to 101 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:7173 sofia/internal/1000@freeswitch.server Set 2833 dtmf receive payload to 101 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:7196 sofia/internal/1000@freeswitch.server Set rtp dtmf delay to 40 2017-09-23 14:44:06.997035 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/1000@freeswitch.server! 2017-09-23 14:44:06.997035 [DEBUG] switch_channel.c:3473 (sofia/internal/1000@freeswitch.server) Callstate Change RINGING -> EARLY 2017-09-23 14:44:06.997035 [DEBUG] switch_core_media.c:6848 Audio params are unchanged for sofia/internal/1000@freeswitch.server.
Phoner Lite:
Same computer and audio device as Express Talk. I set the codec to G.711 u-Law, 64 kbps which matches Express Talk. It was originally set to Opus, 50kbps.
Dial 9196, immediately start saying numbers quickly, stopping when numbers are heard, 2 is heard from echo. The echo application is working and sends back data with minimal delay.
Free Switch log of Phoner Lite connection:
2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:4352 Set telephone-event payload to 101@8000 2017-09-23 15:08:52.917044 [WARNING] switch_core_media.c:4606 Invalid SDP for opus. Don't ask.. but it needs a /2 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:3056 Set Codec sofia/internal/1000@freeswitch.server PCMU/8000 20 ms 160 samples 64000 bits 1 channels 2017-09-23 15:08:52.917044 [DEBUG] switch_core_codec.c:111 sofia/internal/1000@freeswitch.server Original read codec set to PCMU:0 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:4695 Set telephone-event payload to 101@8000 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:4754 sofia/internal/1000@freeswitch.server Set 2833 dtmf send payload to 101 recv payload to 101 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:6865 AUDIO RTP [sofia/internal/1000@freeswitch.server] 72.14.x.x port 23668 -> 71.238.x.x port 5062 codec: 0 ms: 20 2017-09-23 15:08:52.917044 [DEBUG] switch_rtp.c:4096 Starting timer [soft] 160 bytes per 20ms 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:7166 sofia/internal/1000@freeswitch.server Set 2833 dtmf send payload to 101 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:7173 sofia/internal/1000@freeswitch.server Set 2833 dtmf receive payload to 101 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:7196 sofia/internal/1000@freeswitch.server Set rtp dtmf delay to 40 2017-09-23 15:08:52.917044 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/1000@freeswitch.server! 2017-09-23 15:08:52.917044 [DEBUG] switch_channel.c:3473 (sofia/internal/1000@freeswitch.server) Callstate Change RINGING -> EARLY 2017-09-23 15:08:52.917044 [DEBUG] switch_core_media.c:6848 Audio params are unchanged for sofia/internal/1000@freeswitch.server.
Chris
I haven't had a chance to dig into this further. That sounds like a logical place to begin investigating. Thanks.
Chris
On Thu, Sep 21, 2017 at 02:29:25PM +0000, Scott Granados wrote:
Sounds like it could be a problem in Jitter buffer management. What do you think? Have you tried adjusting the type and size of buffer negotiated by the PBX?
On Sep 21, 2017, at 8:56 AM, Chris Nestrud <ccn@chrisnestrud.com> wrote:
Express Talk has a strange issue for me where several seconds of delay will build up after about an hour of a SIP call being connected. I haven't had this delay issue when using Bria on iOS. Express Talk is very accessible. Let me know if you try it and have
On Thu, Sep 21, 2017 at 09:50:13AM -0500, Chris Nestrud wrote: the delay issue.
Chris
On Tue, Sep 19, 2017 at 08:21:34PM -0500, Chris Nestrud wrote:
Express Talk from NCH looks accessible.
http://www.nch.com.au/talk/index.html
The command-line version of Linphone may also be worth a look; the Windows GUI version is not accessible.
Please let me know if you find anything better.
Chris
On Tue, Sep 19, 2017 at 04:30:22PM -0400, Katherine M. Moss wrote:
Hello all,
I forget what the concensus was when we discussed this before, but I seem to be finding difficulties finding a softphone that has both accessibility as well as a good feature set; particularly, I need the ability to pull contacts and what not from Active Directory since we use that; I've heard of Microsip, though I know that it does not support this feature. If X-Lite/Bria used to be accessible, they aren't now; unlabeled fields everywhere which cannot be seemingly deciphered via object navigation with NVDA (Haven't tried JAWS, but I will.) We're
using Asterisk for the server if that matters. Thanks.
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Scott Granados <scott@granados-llc.net> wrote:
That’s a tough one. IPv6 isn’t that common with SIP products yet. We run in to this a lot on our corporate systems. I use XLite and Bria but I don’t think these have the specific features you need.
SIP is also a good use case for IPv6, as it has real problems with NAT. My home network and virtual private server both have IPv6 natively, as do several of the SIP destinations (friends and family, mostly) taht I call regularly. I can configure a Stun server and work around the NAT issues, but IPv6 is the simpler solution - also supported by FreeSWITCH and Asterisk at the server end.
Absolutely agreed. IPv6 solves a lot of problems, this is a big one. A lot of the providers don’t support V6 yet, at least none that I know of. (SIP providers I am speaking of not Network). Not having to fight with NAT or ALG inspection and such would be a good thing. I wasn’t aware Asterisk supported it. I was working with PBX in a Flash and didn’t see that was a possibility. Good to know and thanks for that tidbit.
On Nov 1, 2017, at 11:30 AM, Jason White <jason@jasonjgw.net> wrote:
Scott Granados <scott@granados-llc.net> wrote:
That’s a tough one. IPv6 isn’t that common with SIP products yet. We run in to this a lot on our corporate systems. I use XLite and Bria but I don’t think these have the specific features you need.
SIP is also a good use case for IPv6, as it has real problems with NAT. My home network and virtual private server both have IPv6 natively, as do several of the SIP destinations (friends and family, mostly) taht I call regularly. I can configure a Stun server and work around the NAT issues, but IPv6 is the simpler solution - also supported by FreeSWITCH and Asterisk at the server end.
Returning to an old thread, again: I reproduced the reported audio delay issue with ExpressTalk and FreeSWITCH. Phoner Lite appears to be good, but it needs additional work on its accessibility for screen reader users. As earlier noted, MicroSIP doesn't support IPv6. It does, however, support the Opus codec. There's what appears to be a drag and drop interface for codec selection, which I wasn't able to use, so I configured the codec preference order by editing the configuration file. I haven't found an accessible means of issuing DTMF tones in MicroSIP after a call has started. As noted in the documentation, you can append a sequence of commas and digits to the number to be called, resulting in pauses (the commas) and DTMF tones (the digits) being issued as soon as the connection is made. However, this does not satisfy the need to use the dial pad while the call is in progress. I hope these notes are useful. Comments and questions are welcome. -----Original Message----- From: Scott Granados [mailto:scott@granados-llc.net] Sent: Wednesday, November 1, 2017 3:36 PM To: Jason White <jason@jasonjgw.net> Cc: Blind sysadmins list <blind-sysadmins@lists.hodgsonfamily.org> Subject: Re: [Blind-sysadmins] softphone recommendations Absolutely agreed. IPv6 solves a lot of problems, this is a big one. A lot of the providers don’t support V6 yet, at least none that I know of. (SIP providers I am speaking of not Network). Not having to fight with NAT or ALG inspection and such would be a good thing. I wasn’t aware Asterisk supported it. I was working with PBX in a Flash and didn’t see that was a possibility. Good to know and thanks for that tidbit.
On Nov 1, 2017, at 11:30 AM, Jason White <jason@jasonjgw.net> wrote:
Scott Granados <scott@granados-llc.net> wrote:
That’s a tough one. IPv6 isn’t that common with SIP products yet. We run in to this a lot on our corporate systems. I use XLite and Bria but I don’t think these have the specific features you need.
SIP is also a good use case for IPv6, as it has real problems with NAT. My home network and virtual private server both have IPv6 natively, as do several of the SIP destinations (friends and family, mostly) taht I call regularly. I can configure a Stun server and work around the NAT issues, but IPv6 is the simpler solution - also supported by FreeSWITCH and Asterisk at the server end.
It is, but I don't know if it can read contact card type information from outside sources. On Tue, 19 Sep 2017 20:21:34 -0500, you wrote:
Express Talk from NCH looks accessible.
http://www.nch.com.au/talk/index.html
The command-line version of Linphone may also be worth a look; the Windows GUI version is not accessible.
Please let me know if you find anything better.
Chris
On Tue, Sep 19, 2017 at 04:30:22PM -0400, Katherine M. Moss wrote:
Hello all,
I forget what the concensus was when we discussed this before, but I seem to be finding difficulties finding a softphone that has both accessibility as well as a good feature set; particularly, I need the ability to pull contacts and what not from Active Directory since we use that; I've heard of Microsip, though I know that it does not support this feature. If X-Lite/Bria used to be accessible, they aren't now; unlabeled fields everywhere which cannot be seemingly deciphered via object navigation with NVDA (Haven't tried JAWS, but I will.) We're using Asterisk for the server if that matters. Thanks.
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I think MicroSIP is more accessible instead of others, you can download it from www.microsip.org Also, If you need you can find jaws scripts for above application written by Dlee from the following link: http://dlee.org/MicroSIP/ 2017-10-14 17:31 GMT+03:00, Steve Matzura <sm@noisynotes.com>:
It is, but I don't know if it can read contact card type information from outside sources.
On Tue, 19 Sep 2017 20:21:34 -0500, you wrote:
Express Talk from NCH looks accessible.
http://www.nch.com.au/talk/index.html
The command-line version of Linphone may also be worth a look; the Windows GUI version is not accessible.
Please let me know if you find anything better.
Chris
On Tue, Sep 19, 2017 at 04:30:22PM -0400, Katherine M. Moss wrote:
Hello all,
I forget what the concensus was when we discussed this before, but I seem to be finding difficulties finding a softphone that has both accessibility as well as a good feature set; particularly, I need the ability to pull contacts and what not from Active Directory since we use that; I've heard of Microsip, though I know that it does not support this feature. If X-Lite/Bria used to be accessible, they aren't now; unlabeled fields everywhere which cannot be seemingly deciphered via object navigation with NVDA (Haven't tried JAWS, but I will.) We're using Asterisk for the server if that matters. Thanks.
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-- Can Kırca
I think MicroSIP is more accessible instead of others, you can download it from www.microsip.org Also, If you need you can find jaws scripts for above application written by Dlee from the following link: http://dlee.org/MicroSIP/ 2017-10-14 17:31 GMT+03:00, Steve Matzura <sm@noisynotes.com>:
It is, but I don't know if it can read contact card type information from outside sources.
On Tue, 19 Sep 2017 20:21:34 -0500, you wrote:
Express Talk from NCH looks accessible.
http://www.nch.com.au/talk/index.html
The command-line version of Linphone may also be worth a look; the Windows GUI version is not accessible.
Please let me know if you find anything better.
Chris
On Tue, Sep 19, 2017 at 04:30:22PM -0400, Katherine M. Moss wrote:
Hello all,
I forget what the concensus was when we discussed this before, but I seem to be finding difficulties finding a softphone that has both accessibility as well as a good feature set; particularly, I need the ability to pull contacts and what not from Active Directory since we use that; I've heard of Microsip, though I know that it does not support this feature. If X-Lite/Bria used to be accessible, they aren't now; unlabeled fields everywhere which cannot be seemingly deciphered via object navigation with NVDA (Haven't tried JAWS, but I will.) We're using Asterisk for the server if that matters. Thanks.
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-- Can Kırca
Hi, I often used phoner. though i guess it has no possibility to access the AD but you might import the data using a csv file. Greetings, Simon Mit freundlichen Grüßen Simon Eigeldinger Informatik Nebengebäude 1, OG1 ------------------------- Stadt Hohenems Kaiser-Franz-Josef-Straße 4 6845 Hohenems Österreich Tel.: +43 5576 7101-1143 Fax: +43 5576 7101-1149 E-Mail: simon.eigeldinger@hohenems.at Web: www.hohenems.at Diese Nachricht und allfällige angehängte Dokumente sind vertraulich und nur für den/die Adressaten bestimmt. -----Ursprüngliche Nachricht----- Von: Blind-sysadmins [mailto:blind-sysadmins-bounces@lists.hodgsonfamily.org] Im Auftrag von Katherine M. Moss Gesendet: Dienstag, 19. September 2017 22:30 An: blind-sysadmins@lists.hodgsonfamily.org Betreff: [Blind-sysadmins] softphone recommendations Hello all, I forget what the concensus was when we discussed this before, but I seem to be finding difficulties finding a softphone that has both accessibility as well as a good feature set; particularly, I need the ability to pull contacts and what not from Active Directory since we use that; I've heard of Microsip, though I know that it does not support this feature. If X-Lite/Bria used to be accessible, they aren't now; unlabeled fields everywhere which cannot be seemingly deciphered via object navigation with NVDA (Haven't tried JAWS, but I will.) We're using Asterisk for the server if that matters. Thanks. _______________________________________________ Blind-sysadmins mailing list Blind-sysadmins@lists.hodgsonfamily.org https://lists.hodgsonfamily.org/listinfo/blind-sysadmins
I've never heard of that one. I think I've got 3CX going, but now I just need to find my SIP ID, though at least their fields are labeled. If Bria used to be accessible, then there is certainly a regression there ... ----- Original Message ----- From: "Eigeldinger Simon" <simon.eigeldinger@hohenems.at> To: "Blind sysadmins list" <blind-sysadmins@lists.hodgsonfamily.org> Sent: Wednesday, September 20, 2017 1:22:34 AM Subject: Re: [Blind-sysadmins] softphone recommendations Hi, I often used phoner. though i guess it has no possibility to access the AD but you might import the data using a csv file. Greetings, Simon Mit freundlichen Grüßen Simon Eigeldinger Informatik Nebengebäude 1, OG1 ------------------------- Stadt Hohenems Kaiser-Franz-Josef-Straße 4 6845 Hohenems Österreich Tel.: +43 5576 7101-1143 Fax: +43 5576 7101-1149 E-Mail: simon.eigeldinger@hohenems.at Web: www.hohenems.at Diese Nachricht und allfällige angehängte Dokumente sind vertraulich und nur für den/die Adressaten bestimmt. -----Ursprüngliche Nachricht----- Von: Blind-sysadmins [mailto:blind-sysadmins-bounces@lists.hodgsonfamily.org] Im Auftrag von Katherine M. Moss Gesendet: Dienstag, 19. September 2017 22:30 An: blind-sysadmins@lists.hodgsonfamily.org Betreff: [Blind-sysadmins] softphone recommendations Hello all, I forget what the concensus was when we discussed this before, but I seem to be finding difficulties finding a softphone that has both accessibility as well as a good feature set; particularly, I need the ability to pull contacts and what not from Active Directory since we use that; I've heard of Microsip, though I know that it does not support this feature. If X-Lite/Bria used to be accessible, they aren't now; unlabeled fields everywhere which cannot be seemingly deciphered via object navigation with NVDA (Haven't tried JAWS, but I will.) We're using Asterisk for the server if that matters. Thanks. _______________________________________________ Blind-sysadmins mailing list Blind-sysadmins@lists.hodgsonfamily.org https://lists.hodgsonfamily.org/listinfo/blind-sysadmins _______________________________________________ Blind-sysadmins mailing list Blind-sysadmins@lists.hodgsonfamily.org https://lists.hodgsonfamily.org/listinfo/blind-sysadmins
I've never heard of that one. I think I've got 3CX going, but now I just need to find my SIP ID, though at least their fields are labeled. If Bria used to be accessible, then there is certainly a regression there ... ----- Original Message ----- From: "Eigeldinger Simon" <simon.eigeldinger@hohenems.at> To: "Blind sysadmins list" <blind-sysadmins@lists.hodgsonfamily.org> Sent: Wednesday, September 20, 2017 1:22:34 AM Subject: Re: [Blind-sysadmins] softphone recommendations Hi, I often used phoner. though i guess it has no possibility to access the AD but you might import the data using a csv file. Greetings, Simon Mit freundlichen Grüßen Simon Eigeldinger Informatik Nebengebäude 1, OG1 ------------------------- Stadt Hohenems Kaiser-Franz-Josef-Straße 4 6845 Hohenems Österreich Tel.: +43 5576 7101-1143 Fax: +43 5576 7101-1149 E-Mail: simon.eigeldinger@hohenems.at Web: www.hohenems.at Diese Nachricht und allfällige angehängte Dokumente sind vertraulich und nur für den/die Adressaten bestimmt. -----Ursprüngliche Nachricht----- Von: Blind-sysadmins [mailto:blind-sysadmins-bounces@lists.hodgsonfamily.org] Im Auftrag von Katherine M. Moss Gesendet: Dienstag, 19. September 2017 22:30 An: blind-sysadmins@lists.hodgsonfamily.org Betreff: [Blind-sysadmins] softphone recommendations Hello all, I forget what the concensus was when we discussed this before, but I seem to be finding difficulties finding a softphone that has both accessibility as well as a good feature set; particularly, I need the ability to pull contacts and what not from Active Directory since we use that; I've heard of Microsip, though I know that it does not support this feature. If X-Lite/Bria used to be accessible, they aren't now; unlabeled fields everywhere which cannot be seemingly deciphered via object navigation with NVDA (Haven't tried JAWS, but I will.) We're using Asterisk for the server if that matters. Thanks. _______________________________________________ Blind-sysadmins mailing list Blind-sysadmins@lists.hodgsonfamily.org https://lists.hodgsonfamily.org/listinfo/blind-sysadmins _______________________________________________ Blind-sysadmins mailing list Blind-sysadmins@lists.hodgsonfamily.org https://lists.hodgsonfamily.org/listinfo/blind-sysadmins
Can’t speak for windows but XLite and Bria are accessible on the Mac and on the iPhone and also Android (Bria that is). I will have to try under windows to see if I can think of any ideas to help there but you’ve probably exhausted the options.
On Sep 19, 2017, at 4:30 PM, Katherine M. Moss <kmoss@winterhillsolutions.com> wrote:
Hello all,
I forget what the concensus was when we discussed this before, but I seem to be finding difficulties finding a softphone that has both accessibility as well as a good feature set; particularly, I need the ability to pull contacts and what not from Active Directory since we use that; I've heard of Microsip, though I know that it does not support this feature. If X-Lite/Bria used to be accessible, they aren't now; unlabeled fields everywhere which cannot be seemingly deciphered via object navigation with NVDA (Haven't tried JAWS, but I will.) We're using Asterisk for the server if that matters. Thanks.
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participants (9)
-
Bill Dengler -
Billy Irwin -
Can Kırca -
Chris Nestrud -
Eigeldinger Simon -
Jason White -
Katherine M. Moss -
Scott Granados -
Steve Matzura