Lal, Deepak (Contractor)
2003-Nov-06 15:22 UTC
[Asterisk-Users] Error in Incoming SIP call
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' WARNING[1225991360]: File pbx.c, Line 1154 (pbx_extension_helper): No application '' for extension (incoming, 5147771111, 1) == Spawn extension (incoming, 5147771111, 1) exited non-zero on 'SIP/-08114358' WARNING[1225991360]: File pbx.c, Line 1154 (pbx_extension_helper): No application '' for extension (incoming, h, 1) == Spawn extension (incoming, h, 1) exited non-zero on 'SIP/-08114358' In my extensions file, I have the following defined: [incoming] exten => 5147771111,1,Dial,Zap/2|10 Any suggestions will be appreciated!
what does "show dialplan incoming" show ? Also try using Dial(Zap/bla,10) instead Maritn On Thu, 6 Nov 2003, 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' > WARNING[1225991360]: File pbx.c, Line 1154 (pbx_extension_helper): No > application '' for extension (incoming, 5147771111, 1) > == Spawn extension (incoming, 5147771111, 1) exited non-zero on > 'SIP/-08114358' > WARNING[1225991360]: File pbx.c, Line 1154 (pbx_extension_helper): No > application '' for extension (incoming, h, 1) > == Spawn extension (incoming, h, 1) exited non-zero on 'SIP/-08114358' > > > In my extensions file, I have the following defined: > > [incoming] > > exten => 5147771111,1,Dial,Zap/2|10 > > > > Any suggestions will be appreciated! > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users >
>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' >WARNING[1225991360]: File pbx.c, Line 1154 (pbx_extension_helper): No >application '' for extension (incoming, 5147771111, 1) > == Spawn extension (incoming, 5147771111, 1) exited non-zero on >'SIP/-08114358' >WARNING[1225991360]: File pbx.c, Line 1154 (pbx_extension_helper): No >application '' for extension (incoming, h, 1) > == Spawn extension (incoming, h, 1) exited non-zero on 'SIP/-08114358' > > >In my extensions file, I have the following defined: > >[incoming] > >exten => 5147771111,1,Dial,Zap/2|10 > > >Any suggestions will be appreciated!Try: exten => 5147771111,1,Dial(Zap/2,10) JT
On Thu, 6 Nov 2003, Lal, Deepak (Contractor) wrote:> When I get a SIP call, I get the following error: > WARNING[1225991360]: File pbx.c, Line 1154 (pbx_extension_helper): No > application '' for extension (incoming, 5147771111, 1) > == Spawn extension (incoming, 5147771111, 1) exited non-zero on > 'SIP/-08114358' > WARNING[1225991360]: File pbx.c, Line 1154 (pbx_extension_helper): No > application '' for extension (incoming, h, 1) > == Spawn extension (incoming, h, 1) exited non-zero on 'SIP/-08114358' > In my extensions file, I have the following defined: > > [incoming] > > exten => 5147771111,1,Dial,Zap/2|10Is this copied and pasted or retyped? I've seen this before when you have a space in the extension line. There shouldn't be any. -- Dave Weis "I believe there are more instances of the abridgment djweis@sjdjweis.com of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations."- James Madison
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
Lal, Deepak (Contractor)
2003-Nov-07 06:18 UTC
[Asterisk-Users] Error in Incoming SIP call
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@137.237.233.155:5060;user=phone SIP/2.0
Allow: UPDATE
Call-ID: 000000000000000000004aa4466f@HARRIS3.HARRIS.COM
Contact: <sip:5144211002@172.31.128.11:5061;user=phone>
Content-Type: multipart/mixed;boundary="unique-boundary-1"
CSeq: 220 INVITE
From:
<sip:5144211002@172.31.128.11;user=phone>;tag=000000000000000000004aa44670Max-Fo
rwards: 31
MIME-Version: 1.0
To: <sip:5147771111@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@137.237.233.155:5060;user=phone SIP/2.0
Allow: UPDATE
Call-ID: 000000000000000000004aa4466f@HARRIS3.HARRIS.COM
Contact: <sip:5144211002@172.31.128.11:5061;user=phone>
Content-Type: multipart/mixed;boundary="unique-boundary-1"
CSeq: 220 INVITE
From:
<sip:5144211002@172.31.128.11;user=phone>;tag=000000000000000000004aa44670Max-Fo
rwards: 31
MIME-Version: 1.0
To: <sip:5147771111@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@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@172.31.128.11;user=phone>;tag=000000000000000000004aa44670To:
<sip:5147771111@137.237.233.155;user=phone>;tag=as56dff219
Call-ID: 000000000000000000004aa4466f@HARRIS3.HARRIS.COM
CSeq: 220 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:5147771111@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@172.31.128.11;user=phone>;tag=000000000000000000004aa44670To:
<sip:5147771111@137.237.233.155;user=phone>;tag=as56dff219
Call-ID: 000000000000000000004aa4466f@HARRIS3.HARRIS.COM
CSeq: 220 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:5147771111@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@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
_______________________________________________
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
John Todd wrote:>> exten => 5147771111,1,Dial,Zap/2|10 > Try: > exten => 5147771111,1,Dial(Zap/2,10)I think these two versions of giving arguments are confusing. Reading docs and "show application xxxx" texts, both variants are used, sometimes even in the same text. Is the first syntax old, to be replaced by the more easy to understand second syntax? If so, we have to remove the confusing old syntax from documentation. /Olle