[asterisk-users] setting dtmf mode for a particular peer

[Brian J. Murrell]

On Wed, 2008-04-09 at 16:18 -0500, Tilghman Lesher wrote:
> 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 at voipmich)
> NOT
> Dial(SIP/1234 at tf.voipmich.com)

OK.  Trying exactly as you describe above, it does dial:

    -- Executing [s at macro-ringingdial:2] Dial("SIP/1011002206-b631f650", "SIP/18668398145 at 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 sip:18668398145 at tf.voipmich.com SIP/2.0
Via: SIP/2.0/UDP 67.193.45.68:5060;branch=z9hG4bK2a84f89b;rport
From: "2003" <sip:nobody at nodomain>;tag=as5c70ce0e
To: <sip:18668398145 at tf.voipmich.com>
Contact: <sip:nobody at 67.193.45.68>
Call-ID: 1090b17e076edb94740dfd9c4f436590 at 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 at 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.

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