It works now - It seems I had a space after extension# and that was causing a
problem.
The client is a Cirpack (www.cirpack.com) softswitch. The sip debug output (AS
REQUESTED) is:
<-------------------------------- SIP debug output
------------------------------------>
*CLI> sip debug
SIP Debugging Enabled
*CLI> Sip read:
INVITE sip:[EMAIL PROTECTED]:5060;user=phone SIP/2.0
Allow: UPDATE
Call-ID: [EMAIL PROTECTED]
Contact: <sip:[EMAIL PROTECTED]:5061;user=phone>
Content-Type: multipart/mixed;boundary="unique-boundary-1"
CSeq: 220 INVITE
From:
<sip:[EMAIL PROTECTED];user=phone>;tag=000000000000000000004aa44670Max-Fo
rwards: 31
MIME-Version: 1.0
To: <sip:[EMAIL PROTECTED];user=phone>
User-Agent: Cirpack/v4.3o (gw_sip)
Via: SIP/2.0/UDP 172.31.128.11:5061;branch=z9hG4bK-DAA
Content-Length: 520
--unique-boundary-1
Content-Type: application/ISUP;version=cp10isup;base=etsi121
Content-Disposition: signal;handling=optional
01 07 02 70 00 02 01 03 09 02 0a 00 0a 07 03 13 15 44 12 01 20 04 08 83 10 15 74
77 11 11 0f 06 01 10 00
--unique-boundary-1
Content-Type: application/SDP
v=0
o=cp10 1068206724 1068206724 IN IP4 172.31.128.12
s=SIP Call
c=IN IP4 172.31.128.12
t=0 0
m=audio 16636 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=ptime:20
a=maxptime:30
--unique-boundary-1--
13 headers, 21 lines
Using latest request as basis request
Sending to 172.31.128.11 : 5061 (non-NAT)
NOTICE[1133718080]: File chan_sip.c, Line 1768 (process_sdp): Content is
'multipart/mixed;boundary="unique-boundary-1"', not 'application/sdp'
Sip read:
INVITE sip:[EMAIL PROTECTED]:5060;user=phone SIP/2.0
Allow: UPDATE
Call-ID: [EMAIL PROTECTED]
Contact: <sip:[EMAIL PROTECTED]:5061;user=phone>
Content-Type: multipart/mixed;boundary="unique-boundary-1"
CSeq: 220 INVITE
From:
<sip:[EMAIL PROTECTED];user=phone>;tag=000000000000000000004aa44670Max-Fo
rwards: 31
MIME-Version: 1.0
To: <sip:[EMAIL PROTECTED];user=phone>
User-Agent: Cirpack/v4.3o (gw_sip)
Via: SIP/2.0/UDP 172.31.128.11:5061;branch=z9hG4bK-DAA
Content-Length: 520
--unique-boundary-1
Content-Type: application/ISUP;version=cp10isup;base=etsi121
Content-Disposition: signal;handling=optional
01 07 02 70 00 02 01 03 09 02 0a 00 0a 07 03 13 15 44 12 01 20 04 08 83 10 15 74
77 11 11 0f 06 01 10 00
--unique-boundary-1
Content-Type: application/SDP
v=0
o=cp10 1068206724 1068206724 IN IP4 172.31.128.12
s=SIP Call
c=IN IP4 172.31.128.12
t=0 0
m=audio 16636 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=ptime:20
a=maxptime:30
--unique-boundary-1--
13 headers, 21 lines
Ignoring this request
Looking for 5147771111 in incoming
list_route: hop: <sip:[EMAIL PROTECTED]:5061;user=phone>
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.31.128.11:5061;branch=z9hG4bK-DAA
From:
<sip:[EMAIL PROTECTED];user=phone>;tag=000000000000000000004aa44670To:
<sip:[EMAIL PROTECTED];user=phone>;tag=as56dff219
Call-ID: [EMAIL PROTECTED]
CSeq: 220 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[EMAIL PROTECTED]>
Content-Length: 0
to 172.31.128.11:5061
-- Executing Dial("SIP/-08114370", "Zap/2|10") in new stack
-- Called 2
-- Zap/2-1 is ringing
Transmitting (no NAT):
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.31.128.11:5061;branch=z9hG4bK-DAA
From:
<sip:[EMAIL PROTECTED];user=phone>;tag=000000000000000000004aa44670To:
<sip:[EMAIL PROTECTED];user=phone>;tag=as56dff219
Call-ID: [EMAIL PROTECTED]
CSeq: 220 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[EMAIL PROTECTED]>
Content-Length: 0
to 172.31.128.11:5061
-- Zap/2-1 is ringing
-- Zap/2-1 is ringing
-- Nobody picked up in 10000 ms
-- Hungup 'Zap/2-1'
<--------------------- end of SIP debug ------------------------------>
-----Original Message-----
From: Olle E. Johansson
To: [EMAIL PROTECTED]
Sent: 11/7/03 7:02 AM
Subject: Re: [Asterisk-Users] Error in Incoming SIP call
Lal, Deepak (Contractor) wrote:
> When I get a SIP call, I get the following error:
>
> *CLI> NOTICE[1133718080]: File chan_sip.c, Line 1768 (process_sdp):
Content is
> 'multipart/mixed;boundary="unique-boundary-1"', not 'application/sdp'
Which client is used to place the call? I haven't seen multipart/mixed
used, even
though it is not incorrect at all. Could you do a SIP debug and capture
the whole
SIP invite?
This might not be related to your problem, I'm just curious of what the
other
part of the payload can be.
/O
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