What does your log say?
On 6/14/06, Greg Fausak <lgfausak(a)gmail.com> wrote:
Before I start debugging I thought I'd ask.
I have a sip message from a gateway (Cisco IOS) that
is being sliently dropped. Should ser be able to handle
a message that looks like this? I haven't done a multipart
message before.
Thanks,
---greg
INVITE sip:+19194724170@sn-sip-in.ca-sn1.cisco.com:5060 SIP/2.0
Via: SIP/2.0/UDP 172.18.109.91:5060;branch=z9hG4bK4F0109C
Remote-Party-ID: <sip:
+19199915651(a)172.18.109.91>;party=calling;screen=yes;privacy=off
From: <sip:+19199915651@172.18.109.91>;tag=249E4980-FFC
To: <sip:+19194724170@sn-sip-in.ca-sn1.cisco.com>
Date: Tue, 13 Jun 2006 19:37:55 GMT
Call-ID: F241878C-FA4A11DA-81EFC5C8-2F1D7951(a)172.18.109.91
Supported: 100rel,timer,resource-priority,replaces
Min-SE: 1800
Cisco-Guid: 4064260884-4199158234-2157445126-1397050494
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 10
Timestamp: 1150227475
Contact: <sip:+19199915651@172.18.109.91:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: multipart/mixed;boundary=uniqueBoundary
Mime-Version: 1.0
Content-Length: 797
--uniqueBoundary
Content-Type: application/sdp
Content-Disposition: session;handling=required
v=0
o=CiscoSystemsSIP-GW-UserAgent 8340 2382 IN IP4 172.18.109.91
s=SIP Call
c=IN IP4 172.18.109.91
t=0 0
m=audio 19122 RTP/AVP 18 3 0 4 100 101
c=IN IP4 172.18.109.91
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=yes
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
--uniqueBoundary
Content-Type: application/gtd
Content-Disposition: signal;handling=optional
IAM,
PRN,isdn*,,NI***,
USI,rate,c,s,c,1
USI,lay1,ulaw
TMR,00
CPN,02,,1,4724170
CGN,04,,1,y,2,9199915651
CPC,09
FCI,,,,,,,y,
GCI,f23fb314fa4a11da8098000653454c7e
--uniqueBoundary--
--
Greg Fausak
greg(a)thursday.com
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