[asterisk-users] Asterisk hangs up SIP call after 6 200 retransmits

[Raj Jain]

 One potential reason could be that the ACK request being sent to
Asterisk is malformed. Notice "branch=0" in the top Via. This should start
with "z9hG4bK" magic cookie since the INVITE was an RFC 3261 transaction.

While "branch=0" is valid in RFC 2543, I don't think an INVITE can start-off
as RFC 3261 and then the ACK can switch over to RFC 2543 in the middle of
the transaction. Clearly, Asterisk is dropping this ACK on the floor.

Raj


<-- SIP read from 147.202.nnn.nnn:5060:
ACK sip:6499777777 at 203.89.nnn.nnn SIP/2.0
Record-Route: <sip:147.202.nnn.nnn;ftag=as4917b107;lr=on>
Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=0
Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK61752efe;rport=5060
From: "6494444444" < sip:6494444444 at 202.180.nnn.nnn>;tag=as4917b107
To: <sip:6499777777 at domain.co.nz>;tag=as7cefaa53
Contact: < sip:6494444444 at 202.180.nnn.nnn>
Call-ID: 79620dc1382184b64681b2e85584ca4d at 202.180.nnn.nnn
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 69
Content-Length: 0



--- (12 headers 0 lines) ---
Retransmitting #6 (no NAT) to 147.202.nnn.nnn:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bKd0e3.48331fd3.0;received=
147.202.nnn.nnn
Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK3757e55c;rport=5060
Record-Route: <sip:147.202.nnn.nnn;ftag=as4917b107;lr=on>
From: "6494444444" <sip:6494444444 at 202.180.nnn.nnn>;tag=as4917b107
To: <sip:6499777777 at domain.co.nz>;tag=as7cefaa53
Call-ID: 79620dc1382184b64681b2e85584ca4d at 202.180.nnn.nnn
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:6499777777 at 203.89.nnn.nnn>
Content-Type: application/sdp
Content-Length: 244

v=0
o=root 16300 16300 IN IP4 203.89.nnn.nnn
s=session
c=IN IP4 203.89.nnn.nnn
t=0 0
m=audio 11648 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
capetown*CLI>
<-- SIP read from 147.202.nnn.nnn:5060:
ACK sip:6499777777 at 203.89.nnn.nnn SIP/2.0
Record-Route: <sip:147.202.nnn.nnn;ftag=as4917b107;lr=on>
Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=0
Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK0c397910;rport=5060
From: "6494444444" <sip:6494444444 at 202.180.nnn.nnn>;tag=as4917b107
To: <sip:6499777777 at domain.co.nz>;tag=as7cefaa53
Contact: <sip:6494444444 at 202.180.nnn.nnn>
Call-ID: 79620dc1382184b64681b2e85584ca4d at 202.180.nnn.nnn
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 69
Content-Length: 0


--- (12 headers 0 lines) ---
== Spawn extension (ivr-3, s, 7) exited non-zero on
'SIP/6499777777-b791bb60'
   -- Executing Hangup("SIP/6499777777-b791bb60", "") in new stack
== Spawn extension (ivr-3, h, 1) exited non-zero on
'SIP/6499777777-b791bb60'
Destroying call '79620dc1382184b64681b2e85584ca4d at 202.180.nnn.nnn'
capetown*CLI>

Any advice in resolving this issue would be greatly appreciated.

Regards

Cameron



___________________________________________________________
What kind of emailer are you? Find out today - get a free analysis of your
email personality. Take the quiz at the Yahoo! Mail Championship.
http://uk.rd.yahoo.com/evt=44106/*http://mail.yahoo.net/uk
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070329/ad932af3/attachment.htm