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Have you tried the using the "SIPDtmfMode" function in your dial plan?
It can be used to change the DTMF mode between two points in a call.
The problem, I would think, would be if your phones are set up to ONLY
send inband audio then you have to find someway to get audio to
transcode the DTMF from inband to info. I'm not familiar enough with
the specifics of Asterisk's behavior to know whether that "just works"
or if it needs some special setup. Try putting SipDtmfMode(info) just
before the dial command and see what happens.<br>
<br>
Good Luck,<br>
Brent<br>
<br>
<br>
Brian J. Murrell wrote:
<blockquote cite="mid:1207778497.3303.94.camel@pc.ilinx" type="cite">
<pre wrap="">On Wed, 2008-04-09 at 16:18 -0500, Tilghman Lesher wrote:
</pre>
<blockquote type="cite">
<pre wrap="">No, that's correct. The problem is that you aren't using the peer definition
when you dial (as you said, you've never needed it before).
Use
Dial(SIP/1234@voipmich)
NOT
Dial(<a class="moz-txt-link-abbreviated" href="mailto:SIP/1234@tf.voipmich.com">SIP/1234@tf.voipmich.com</a>)
</pre>
</blockquote>
<pre wrap=""><!---->
OK. Trying exactly as you describe above, it does dial:
-- Executing [s@macro-ringingdial:2] Dial("SIP/1011002206-b631f650", "SIP/18668398145@voipmich") in new stack
With "sip set debug peer voipmich" I'd expect to see SIP packets for
every digit I press on my phone, right? I don't. I don't see anything
beyond the initial call establishment:
Audio is at 67.193.45.68 port 11724
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Reliably Transmitting (NAT) to 69.41.0.50:5060:
INVITE <a class="moz-txt-link-abbreviated" href="mailto:sip:18668398145@tf.voipmich.com">sip:18668398145@tf.voipmich.com</a> SIP/2.0
Via: SIP/2.0/UDP 67.193.45.68:5060;branch=z9hG4bK2a84f89b;rport
From: "2003" <sip:nobody@nodomain>;tag=as5c70ce0e
To: <a class="moz-txt-link-rfc2396E" href="mailto:sip:18668398145@tf.voipmich.com"><sip:18668398145@tf.voipmich.com></a>
Contact: <a class="moz-txt-link-rfc2396E" href="mailto:sip:nobody@67.193.45.68"><sip:nobody@67.193.45.68></a>
Call-ID: 1090b17e076edb94740dfd9c4f436590@nodomain
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 09 Apr 2008 21:55:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 207
v=0
o=root 11375 11375 IN IP4 67.193.45.68
s=session
c=IN IP4 67.193.45.68
t=0 0
m=audio 11724 RTP/AVP 0 3
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
-- Called 18668398145@voipmich
-- SIP/voipmich-084a5500 is making progress passing it to SIP/1011002206-b631f650
== Spawn extension (macro-ringingdial, s, 2) exited non-zero on 'SIP/1011002206-b631f650' in macro 'ringingdial'
== Spawn extension (macro-ringingdial, s, 2) exited non-zero on 'SIP/1011002206-b631f650'
Of course, in there between the call being established and torn down, I
did hit lots of digits on my phone.
b.
</pre>
<pre wrap="">
<hr size="4" width="90%">
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