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/[EMAIL PROTECTED]) > NOT > Dial(SIP/[EMAIL PROTECTED])
OK. Trying exactly as you describe above, it does dial:
-- Executing [EMAIL PROTECTED]:2] Dial("SIP/1011002206-b631f650",
"SIP/[EMAIL PROTECTED]") 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:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 67.193.45.68:5060;branch=z9hG4bK2a84f89b;rport
From: "2003" <sip:[EMAIL PROTECTED]>;tag=as5c70ce0e
To: <sip:[EMAIL PROTECTED]>
Contact: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
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 [EMAIL PROTECTED]
-- 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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