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[asterisk-users] No DTMF on Sip Trunk?

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noahisaacmiller at gma...
Guest

Posted: Thu Apr 24, 2008 11:02 am Post subject: [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 (1.2.1: 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

jsmith at digium.com
Guest

Posted: Thu Apr 24, 2008 11:18 am Post subject: [asterisk-users] No DTMF on Sip Trunk?

On Thu, 2008-04-24 at 12:02 -0400, Noah Miller wrote:

Quote:

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.

noahisaacmiller at gma...
Guest

Posted: Thu Apr 24, 2008 12:33 pm Post subject: [asterisk-users] No DTMF on Sip Trunk?

Hi Jared -

Quote:

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.

eric at fnords.org
Guest

Posted: Thu Apr 24, 2008 3:39 pm Post subject: [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:

Quote:

Hi Jared -

Quote:

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.

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

_______________________________________________
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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--
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.

noahisaacmiller at gma...
Guest

Posted: Thu Apr 24, 2008 4:01 pm Post subject: [asterisk-users] No DTMF on Sip Trunk?

Quote:

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

Quote:

Noah Miller wrote:

Quote:

Hi Jared -

Quote:

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.

Quote:

_______________________________________________
-- 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

oej at edvina.net
Guest

Posted: Thu Apr 24, 2008 4:49 pm Post subject: [asterisk-users] No DTMF on Sip Trunk?

24 apr 2008 kl. 23.01 skrev Noah Miller:

Quote:

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.

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. Smile

/O

---
* Olle E. Johansson - oej at edvina.net
* Asterisk Training http://edvina.net/training/
* The Asterisk SIP Masterclass in Barcelona, May 5-9 - REGISTER now!

kshumard at digium.com
Guest

Posted: Thu Apr 24, 2008 5:16 pm Post subject: [asterisk-users] No DTMF on Sip Trunk?

Quote:

-------- 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

Quote:

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.

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

noahisaacmiller at gma...
Guest

Posted: Thu Apr 24, 2008 5:43 pm Post subject: [asterisk-users] No DTMF on Sip Trunk?

Quote:

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

eric at fnords.org
Guest

Posted: Thu Apr 24, 2008 7:30 pm Post subject: [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:

Quote:

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.

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