[asterisk-users] No DTMF on Sip Connection between two asterisk boxes?

[Noah Miller]

Hi Olle -

>  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. :-)

Well, I stand corrected, and straight from the SIP-Lord's* fingers.  I
have adjusted the subject of this thread accordingly.  I guess I was
thinking of word "trunk" colloquially, as in a something that connects
calls from multiple devices to another location.

Anyhoo, I'll go ahead and ask Digium support, but if anyone here has
any insight, please let me know.  Since I changed the thread subject,
I'll repost the original question:

For the first time, I'm setting up SIP connections 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


* In the asterisk universe, SIP-Lords are the good guys ;-)