Devileyezz
join:2005-09-19
Toronto, ON
|  FreePBX - PIAF - Voicenetwork - Aastra DTMF issues
Hi guys,
I recently installed a PBX in a Flash 1.3 distro and set it up. I have a Linksys SPA 941 (fw: 5.1.8) and Aastra 57i (fw: 2.3.1.26) as my phones.
My VoIP provider is Voicenetwork.ca. On there, they give me an option to use the dtmf mode I like and it was set to rfc2833.
Asterisk version: 1.4.21.2 (I can't upgrade because PIAF has decided to freeze updates till the dahdi implementation is complete or something like that)
Zaptel: 1.4.12.1
Libpri: 1.4.7
I call 800-387-2524 and it looks like an Asterisk PBX, I press 1 for english and it doesn't work!
Many other IVRs my dtmf doesn't work.
Aastra calls rfc2833 as RTP.
Linksys calls it AVT.
So every instance has rfc2833 in some form or another. Yet, my DTMF simply doesn't work.
How do I fix it?
Please help! |
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celtic
join:2001-02-08
USA | I find dtmf=info works best. Give it a try. |
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Devileyezz
join:2005-09-19
Toronto, ON | Nothing helps.  |
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psevge
Premium
join:2004-08-28
Santa Clara, CA | reply to Devileyezz
There is an asterisk bug because of which RFC2833 does not work - »bugs.digium.com/view.php?id=13209
It's possible that your provider does not support dtmf 'info'. |
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celtic
join:2001-02-08
USA
| reply to Devileyezz
Did you check the setting in both trunk and extension?
»www.voipinfo.org/wiki/view/Asterisk+DTMF
»www.voip-info.org/wiki-Asterisk+sip+dtmfmode |
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Devileyezz
join:2005-09-19
Toronto, ON | Yes, ofcourse.
I wouldn't post here if I didn't look everywhere. Nothing in there helps. |
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Devileyezz
join:2005-09-19
Toronto, ON | reply to psevge
Oh, just great! I added to it too, thanks for the pointer! |
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Devileyezz
join:2005-09-19
Toronto, ON | reply to Devileyezz
No one else? |
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baysoor
join:2002-03-12
San Jose, CA | I have the exact same problem with elastix 1.3 and voicepulse. future-nine (sip info) and gizmoproject seem to be immune from this problem when I call out from them. |
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kaila
join:2000-10-11
Lincolnshire, IL
clubs: 
| said by baysoor  :I have the exact same problem with elastix 1.3 and voicepulse. future-nine (sip info) and gizmoproject seem to be immune from this problem when I call out from them. Try switching to inband on the voicepulse trunk. Place dtmfmode=inband in your peer details of the voicepulse trunk. |
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psevge
Premium
join:2004-08-28
Santa Clara, CA | Voicepulse does NOT support inband. It's either rfc2833 or info. |
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Devileyezz
join:2005-09-19
Toronto, ON | reply to Devileyezz
Atleast you guys can update and be done with it. I can't even do that. |
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kaila
join:2000-10-11
Lincolnshire, IL
clubs:
| reply to psevge
said by psevge :Voicepulse does NOT support inband. It's either rfc2833 or info. I'm using inband now. VoicePulse also allows rfc2833compensate=yes in the peer details to combat tone loops, but it's not a 100% fix. |
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psevge
Premium
join:2004-08-28
Santa Clara, CA | Maybe they changed it recently. During the first week of Sept when I switch my Voicepulse trunks to their sjc and nyc servers, inband wouldn't work. |
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baysoor
join:2002-03-12
San Jose, CA
 ·VoicePulse for Bus..
 ·AT&T Yahoo
| reply to kaila
Try switching to inband on the voicepulse trunk. Place dtmfmode=inband in your peer details of the voicepulse trunk.
I had asked voicepulse for help. Voicepulse does not support sip info. They also told me to try inband. It was a disaster. Nothing worked.
I have also tried relaxdtmf and rfc2833compensate=yes without any success. |
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Devileyezz
join:2005-09-19
Toronto, ON | Same here. It's weird.
Anyway, info all over seems to work for me now. |
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