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
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.
> -------- Forwarded Message -------- > From: Noah Miller <noahisaacmiller at gmail.com> > Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users at lists.digium.com> > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users at 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