[asterisk-users] No DTMF on Sip Trunk?
Hi All - For the first time, I'm setting up SIP trunking between two asterisk boxes. The calls themselves work fine, but I'm not able to get DTMF working. I've tried using inband, rfc2833 and auto, and none of them work. Maybe I'm missing something obvious? Here's my config: Asterisk1 (1.2.18): sip.conf [129trunk551] type=friend secret= username=129trunk551 host=xxx.xxx.xxx.xxx context=phones dtmfmode=auto qualify=1000 disallow=all allow=ulaw insecure=very Asterisk2 (ABE revC): sip.conf [129trunk551] type=friend secret=*** username=129trunk551 host=yyy.yyy.yyy.yyy context=default dtmfmode=auto qualify=1000 disallow=all allow=ulaw insecure=very Thanks, Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No DTMF on Sip Trunk?
On Thu, 2008-04-24 at 12:02 -0400, Noah Miller wrote: For the first time, I'm setting up SIP trunking between two asterisk boxes. The calls themselves work fine, but I'm not able to get DTMF working. If you're connecting an Asterisk 1.2 box to an Asterisk 1.4 box (as it appears that you are), you'll need to set rfc2833compensate=yes in the peer or friend section of sip.conf on the Asterisk 1.4 box. This tells Asterisk to send RFC2833 DTMF the way that Asterisk 1.2 expects it, instead of the newer (read: more standards compliant) way that Asterisk 1.4 now handles RFC2833 DTMF tones. In a nutshell, try adding rfc2833compensate=yes to your section named [129trunk551] on the box you're calling Asterisk2. -- Jared Smith Community Relations Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No DTMF on Sip Trunk?
Hi Jared - For the first time, I'm setting up SIP trunking between two asterisk boxes. The calls themselves work fine, but I'm not able to get DTMF working. If you're connecting an Asterisk 1.2 box to an Asterisk 1.4 box (as it appears that you are), you'll need to set rfc2833compensate=yes in the peer or friend section of sip.conf on the Asterisk 1.4 box. Unfortunately, this didn't work. Maybe rfc2833compensate isn't available in ABE? I think this may require inband signalling anyway, as we'll require non-sip (zap) devices to be able to use these sip trunks and enter DTMF. Any other ideas? Thanks! Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No DTMF on Sip Trunk?
For ABE support you really should contact Digium. BTW, there is no such thing as a sip trunk. It's a sip peer or connection or account. Noah Miller wrote: Hi Jared - For the first time, I'm setting up SIP trunking between two asterisk boxes. The calls themselves work fine, but I'm not able to get DTMF working. If you're connecting an Asterisk 1.2 box to an Asterisk 1.4 box (as it appears that you are), you'll need to set rfc2833compensate=yes in the peer or friend section of sip.conf on the Asterisk 1.4 box. Unfortunately, this didn't work. Maybe rfc2833compensate isn't available in ABE? I think this may require inband signalling anyway, as we'll require non-sip (zap) devices to be able to use these sip trunks and enter DTMF. Any other ideas? Thanks! Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No DTMF on Sip Trunk?
For ABE support you really should contact Digium. BTW, there is no such thing as a sip trunk. It's a sip peer or connection or account. shrug Semantics. IAX connections between two asterisk boxes are often called IAX trunks. This is the same thing in SIP flavor./shrug Also, no offense against Digium support, but the list actually responds more quickly at this point. I think the Digium support staff are in a situation of high demand and short staffing. - Noah Noah Miller wrote: Hi Jared - For the first time, I'm setting up SIP trunking between two asterisk boxes. The calls themselves work fine, but I'm not able to get DTMF working. If you're connecting an Asterisk 1.2 box to an Asterisk 1.4 box (as it appears that you are), you'll need to set rfc2833compensate=yes in the peer or friend section of sip.conf on the Asterisk 1.4 box. Unfortunately, this didn't work. Maybe rfc2833compensate isn't available in ABE? I think this may require inband signalling anyway, as we'll require non-sip (zap) devices to be able to use these sip trunks and enter DTMF. Any other ideas? Thanks! Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, T-1, PRI, Frame Relay, Linux, and network design. Based near Birmingham, AL. Now accepting clients worldwide. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No DTMF on Sip Trunk?
24 apr 2008 kl. 23.01 skrev Noah Miller: For ABE support you really should contact Digium. BTW, there is no such thing as a sip trunk. It's a sip peer or connection or account. shrug Semantics. IAX connections between two asterisk boxes are often called IAX trunks. This is the same thing in SIP flavor./shrug Also, no offense against Digium support, but the list actually responds more quickly at this point. I think the Digium support staff are in a situation of high demand and short staffing. Actually, there's a large difference between an IAX2 trunk and an IAX2 connection. The IAX2 trunk multiplexes multiple media streams in one UDP packet, therefore you can call it trunking. In order for this to work, you need to enable a zaptel timer source in your system. As Eric say, there's no trunking support similar to IAX2 trunks in the SIP channel driver. Semantics, but important in this case. :-) /O --- * Olle E. Johansson - [EMAIL PROTECTED] * Asterisk Training http://edvina.net/training/ * The Asterisk SIP Masterclass in Barcelona, May 5-9 - REGISTER now! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No DTMF on Sip Trunk?
Forwarded Message From: Noah Miller [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] No DTMF on Sip Trunk? Date: Thu, 24 Apr 2008 17:01:18 -0400 For ABE support you really should contact Digium. BTW, there is no such thing as a sip trunk. It's a sip peer or connection or account. shrug Semantics. IAX connections between two asterisk boxes are often called IAX trunks. This is the same thing in SIP flavor./shrug Also, no offense against Digium support, but the list actually responds more quickly at this point. I think the Digium support staff are in a situation of high demand and short staffing. - Noah Actually, Digium Support has been quite responsive in recent weeks, as noted on this list 2 weeks ago: http://lists.digium.com/pipermail/asterisk-users/2008-April/209457.html We strive to be as responsive as we can, and have had some success on this front recently. Please give us a chance! Noah, if you have a specific support experience where we weren't as responsive as we could have been, please contact me off-list to discuss. I want to hear about it! ~Kenny Shumard Digium Technical Support Manager ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No DTMF on Sip Trunk?
Actually, Digium Support has been quite responsive in recent weeks, as noted on this list 2 weeks ago: http://lists.digium.com/pipermail/asterisk-users/2008-April/209457.html We strive to be as responsive as we can, and have had some success on this front recently. Please give us a chance! Thanks Kenny! I don't mean to disparage you folks. You've always been extremely knowledgeable and courteous. Glad to see you get some praise. I just had a simple little question, and I thought I'd ask on the list to see if anyone else had seen this before. - Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No DTMF on Sip Trunk?
No, it is not the same thing. An IAX2 Trunk is a version of IAX2 that puts audio from multiple calls between the same two servers into a single UDP packet. Fewer packets need to be sent so you use the bandwidth much more efficiency because you don't have the packet header overhead. SIP does nothing similar. Noah Miller wrote: For ABE support you really should contact Digium. BTW, there is no such thing as a sip trunk. It's a sip peer or connection or account. shrug Semantics. IAX connections between two asterisk boxes are often called IAX trunks. This is the same thing in SIP flavor./shrug Also, no offense against Digium support, but the list actually responds more quickly at this point. I think the Digium support staff are in a situation of high demand and short staffing. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
