Fusion PBX: how to get feature codes working in calls?
Hello Everyone. I have installed fusion PBX as recommended and it seems to be working fine. What I can't seem to get working is the feature codes during a call. Non of the star codes referenced in the user guide work, and I can find no reference within the various sections of fusion PBX to a feature codes module or app. The most important code I need to work is toggling call recordings during a call. Sometimes, it's necessary to pause the recording when someone provides credit card info for example. star 2 sounds like it would do the trick. Does anyone have any idea what I need to do to get in call codes working? This is a stock install and very few settings have been changed. Many thanks, Mo.
Mobeen Iqbal <mobeeniqbal@gmail.com> wrote:
Does anyone have any idea what I need to do to get in call codes working?
First question: is your Asterisk installation recognizing DTMF tones? If you try voicemail or another feature that requires them, are they properly detected? There are different settings for sending/receiving DTMF if you're using a SIP phone, so make sure that the client side is properly configured. As noted on the following Web page, DTMF can be distorted by audio compression, which is why you should use RFC2833 or one of the other standards mentioned: https://www.voip-info.org/wiki/view/Asterisk+sip+dtmfmode If DTMF is working, then the problem will be somewhere in the Asterisk configuration. It's a long time since I've dealt with Asterisk configuration, so I'm probably not going to be of much help.
Hi. Many thanks for your email. I'm using fusion PBX. Is that still based on asterisk's method of configuration? Very best wishes, Mo. On 07/06/2017 02:05, Jason White via Blind-sysadmins wrote:
Mobeen Iqbal <mobeeniqbal@gmail.com> wrote:
Does anyone have any idea what I need to do to get in call codes working?
First question: is your Asterisk installation recognizing DTMF tones? If you try voicemail or another feature that requires them, are they properly detected?
There are different settings for sending/receiving DTMF if you're using a SIP phone, so make sure that the client side is properly configured. As noted on the following Web page, DTMF can be distorted by audio compression, which is why you should use RFC2833 or one of the other standards mentioned: https://www.voip-info.org/wiki/view/Asterisk+sip+dtmfmode
If DTMF is working, then the problem will be somewhere in the Asterisk configuration. It's a long time since I've dealt with Asterisk configuration, so I'm probably not going to be of much help.
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No, it uses FreeSWITCH. Go on #fusionpbx on Freenode. The main developer is there often, and is quite helpful. He will try to sell you his classes, but otherwise the community frequents that channel. Bill
On Jun 7, 2017, at 2:34 AM, Mobeen Iqbal <mobeeniqbal@gmail.com> wrote:
Hi. Many thanks for your email. I'm using fusion PBX. Is that still based on asterisk's method of configuration? Very best wishes, Mo.
On 07/06/2017 02:05, Jason White via Blind-sysadmins wrote:
Mobeen Iqbal <mobeeniqbal@gmail.com> wrote:
Does anyone have any idea what I need to do to get in call codes working?
First question: is your Asterisk installation recognizing DTMF tones? If you try voicemail or another feature that requires them, are they properly detected?
There are different settings for sending/receiving DTMF if you're using a SIP phone, so make sure that the client side is properly configured. As noted on the following Web page, DTMF can be distorted by audio compression, which is why you should use RFC2833 or one of the other standards mentioned: https://www.voip-info.org/wiki/view/Asterisk+sip+dtmfmode
If DTMF is working, then the problem will be somewhere in the Asterisk configuration. It's a long time since I've dealt with Asterisk configuration, so I'm probably not going to be of much help.
_______________________________________________ Blind-sysadmins mailing list Blind-sysadmins@lists.hodgsonfamily.org https://lists.hodgsonfamily.org/listinfo/blind-sysadmins
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Bill Dengler <codeofdusk@gmail.com> wrote:
No, it uses FreeSWITCH.
That's good - I can help a little with FreeSWITCH, if necessary. However, I suggest following Bill's advice and trying the IRC channel first. The default FreeSWITCH configuration (the one supplied with FreeSWITCH itself) does include the commands that you want, configured using the bind_meta facility - see dialplan/default.xml and dialplan/features.xml for the actual feature configurations.
participants (3)
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Bill Dengler -
Jason White -
Mobeen Iqbal