[Asterisk-Users] Polycom DTMF after connection not working
On a polycom 600 which is working perfectly otherwise, I am unable to use DTMF with IVR or such - not even to dialout of a Sipura setup elsewhere. Other phones (analogue connected to ATA) are accepted. I suspectthe phone is not using rfc2833 but I don't know how to specify that it should useit (not available on the http configuration). ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom DTMF
Problem: Polycom SoundPoint IP phones (running SIP) ceases to send DTMF that Asterisk can detect and use. It worked in 1.0.5, but has not worked since. This has been verified on SoundPoint IP 300's and SoundPoint IP 600's. Workaround: It used to be that for DTMF to work, I had to set the mode in sip.conf to inband. Without making any configuration changes on the phones, I changed the DTMF mode to rfc2833. The DTMF is recognized. No reboot to the phone is necessary, and remember that you can reload the sip configuration with a reload in Asterisk, meaning your PBX doesn't have to be restarted either. Discussion: This is probably not the right way to fix this, as Polycom's configurations, by default, will encode DTMF in the active RTP stream. There may have been a change in the sip channel's code that is causing this. Others on the list have indicated that they worked around the problem by reverting the version of the sip app to an older version. As the new code usually fixes other problems, the solution of reverting seemed to be counter-productive, so I tried other DTMF signalling modes. Thankfully, the stock Polycom configs will work with Asterisk's sip.conf rfc2833 DTMF mode, at least as of CVS-v1-0-03/23/05-21:40:48. When I get more time, or if someone else has the time, an examination of what changed to cause this could enable us to fix the heart of the matter. Other users on the Asterisk list (see thread *-1.0.7 DTFM = Not working from 03/23/2005) have reported other UAs not working. Therefore, there may be a bigger problem with the fundamental issue at hand: when do we change DTMF in channels, to ensure compliance with standards, as well as compatibility with older UAs. Hope this helps someone. Sincerely, David Gomillion ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom DTMF
On Thu, 24 Mar 2005, David Gomillion wrote: Problem: Polycom SoundPoint IP phones (running SIP) ceases to send DTMF that Asterisk can detect and use. It worked in 1.0.5, but has not worked since. This has been verified on SoundPoint IP 300's and SoundPoint IP 600's. Workaround: It used to be that for DTMF to work, I had to set the mode in sip.conf to inband. Without making any configuration changes on the phones, I changed the DTMF mode to rfc2833. The DTMF is recognized. No reboot to the phone is necessary, and remember that you can reload the sip configuration with a reload in Asterisk, meaning your PBX doesn't have to be restarted either. Discussion: This is probably not the right way to fix this, as Polycom's configurations, by default, will encode DTMF in the active RTP stream. There may have been a change in the sip channel's code that is causing this. Others on the list have indicated that they worked around the problem by reverting the version of the sip app to an older version. As the new code usually fixes other problems, the solution of reverting seemed to be counter-productive, so I tried other DTMF signalling modes. Thankfully, the stock Polycom configs will work with Asterisk's sip.conf rfc2833 DTMF mode, at least as of CVS-v1-0-03/23/05-21:40:48. When I get more time, or if someone else has the time, an examination of what changed to cause this could enable us to fix the heart of the matter. Other users on the Asterisk list (see thread *-1.0.7 DTFM = Not working from 03/23/2005) have reported other UAs not working. Therefore, there may be a bigger problem with the fundamental issue at hand: when do we change DTMF in channels, to ensure compliance with standards, as well as compatibility with older UAs. Hope this helps someone. Sincerely, David Gomillion David, I noticed this in testing the 1.0.7 Release Candidate w/ my Polycom phones. I posted the following in http://bugs.digium.com/bug_view_page.php?bug_id=0003746 03-16-05 16:05 Alright. I updated my Development and Home PBXs to 1.0.7 w/ Slepp's Dundi 1.0.2_diff-4 patch. Both are running solid. One issue that I noticed is that my Polycom SP IP phones had to be changed to use RFC2833 signalling instead of the Inband signalling I had been using earlier. I simply modified the sip.conf to have dtmfmode=rfc2833. This proved to be a slight gotcha for a couple of clients when I updated THEIR boxes. This could be just a fluke, and I may be an idiot for having used inband DTMF in the first place, but it is something to be concious of. Can anyone pinpoint a specific SIP patch that may have been applied where this may have been affected? How should this be handled? Should I add it as a new bug-note? Or should we just chalk it up and slap a notice in the release notes? Should inband signaling be broken in 1.0.7 for Polycom phones? Or should it work? I guess this is the question that needs to be asked.. is it normal behaviour or a bug? I'm not sure if this is related, but I also enabled MMX math routines for Zaptel. I never got any followup on it before 1.0.7 was dropped. I'm going to open up a new bug report on this in Mantis and see what we can get as far as comments. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom DTMF
On Thu, 24 Mar 2005 09:36:21 -0600, David Gomillion [EMAIL PROTECTED] wrote: Problem: Polycom SoundPoint IP phones (running SIP) ceases to send DTMF that Asterisk can detect and use. It worked in 1.0.5, but has not worked since. This has been verified on SoundPoint IP 300's and SoundPoint IP 600's. Workaround: It used to be that for DTMF to work, I had to set the mode in sip.conf to inband. Without making any configuration changes on the phones, I changed the DTMF mode to rfc2833. The DTMF is recognized. No reboot to the phone is necessary, and remember that you can reload the sip configuration with a reload in Asterisk, meaning your PBX doesn't have to be restarted either. Discussion: This is probably not the right way to fix this, as Polycom's configurations, by default, will encode DTMF in the active RTP stream. ... No, it is probably the right way to fix it. RFC2833 is the preferred way of transporting DTMF. RFC2833 encodes the DTMF into the RTP packets as distinct messages. In fact, RFC2833 is an enhancement to RTP to carry DTMF. Inband signalling also puts the DTMF into the RTP too, but into the media itself. That means the DTMF is encoded as sound. Inband DTMF should be avoided whenever possible. On low-bandwidth codes, it is very easy for the frequency of the sounds to be altered slightly so the tones are not recognized. Not to mention all of the issues around the lenght of the tones, leading to missed digits, or doubled digits. Tom ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom DTMF
This is problem with the SIP chanell in asterisk; my SPA-3000 inband is not working either so I stay with 1.0.5 #Joseph [snip] I noticed this in testing the 1.0.7 Release Candidate w/ my Polycom phones. I posted the following in http://bugs.digium.com/bug_view_page.php?bug_id=0003746 03-16-05 16:05 Alright. I updated my Development and Home PBXs to 1.0.7 w/ Slepp's Dundi 1.0.2_diff-4 patch. Both are running solid. One issue that I noticed is that my Polycom SP IP phones had to be changed to use RFC2833 signalling instead of the Inband signalling I had been using earlier. I simply modified the sip.conf to have dtmfmode=rfc2833. This proved to be a slight gotcha for a couple of clients when I updated THEIR boxes. This could be just a fluke, and I may be an idiot for having used inband DTMF in the first place, but it is something to be concious of. Can anyone pinpoint a specific SIP patch that may have been applied where this may have been affected? How should this be handled? Should I add it as a new bug-note? Or should we just chalk it up and slap a notice in the release notes? Should inband signaling be broken in 1.0.7 for Polycom phones? Or should it work? I guess this is the question that needs to be asked.. is it normal behaviour or a bug? I'm not sure if this is related, but I also enabled MMX math routines for Zaptel. I never got any followup on it before 1.0.7 was dropped. I'm going to open up a new bug report on this in Mantis and see what we can get as far as comments. -- #Joseph ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
