[Asterisk-Users] Error in Incoming SIP call

Lal, Deepak (Contractor) dlal at harris.com
Fri Nov 7 06:18:59 MST 2003


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:5147771111 at 137.237.233.155:5060;user=phone SIP/2.0
Allow: UPDATE
Call-ID: 000000000000000000004aa4466f at HARRIS3.HARRIS.COM
Contact: <sip:5144211002 at 172.31.128.11:5061;user=phone>
Content-Type: multipart/mixed;boundary="unique-boundary-1"
CSeq: 220 INVITE
From:
<sip:5144211002 at 172.31.128.11;user=phone>;tag=000000000000000000004aa44670Max-Fo
rwards: 31
MIME-Version: 1.0
To: <sip:5147771111 at 137.237.233.155;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:5147771111 at 137.237.233.155:5060;user=phone SIP/2.0
Allow: UPDATE
Call-ID: 000000000000000000004aa4466f at HARRIS3.HARRIS.COM
Contact: <sip:5144211002 at 172.31.128.11:5061;user=phone>
Content-Type: multipart/mixed;boundary="unique-boundary-1"
CSeq: 220 INVITE
From:
<sip:5144211002 at 172.31.128.11;user=phone>;tag=000000000000000000004aa44670Max-Fo
rwards: 31
MIME-Version: 1.0
To: <sip:5147771111 at 137.237.233.155;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:5144211002 at 172.31.128.11:5061;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:5144211002 at 172.31.128.11;user=phone>;tag=000000000000000000004aa44670To:
<sip:5147771111 at 137.237.233.155;user=phone>;tag=as56dff219
Call-ID: 000000000000000000004aa4466f at HARRIS3.HARRIS.COM
CSeq: 220 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:5147771111 at 137.237.233.155>
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:5144211002 at 172.31.128.11;user=phone>;tag=000000000000000000004aa44670To:
<sip:5147771111 at 137.237.233.155;user=phone>;tag=as56dff219
Call-ID: 000000000000000000004aa4466f at HARRIS3.HARRIS.COM
CSeq: 220 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:5147771111 at 137.237.233.155>
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: asterisk-users at lists.digium.com
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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