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

kjcsb kjcsb at yahoo.com
Sun Apr 1 20:42:57 MST 2007 

>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. 

OK. But in the calls that don't get dropped, the "branch=0" is present also. See below for an example:

<-- SIP read from 147.202.nnn.nnn:5060: 
INVITE sip:6499777777 at 203.89.nnn.nnn SIP/2.0
Record-Route: <sip:147.202.nnn.nnn;ftag=as1370b1ab;lr=on>
Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bK22ab.697375a4.0
Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK4cf2bb78;rport=5060
From: "6494444444" <sip:6494444444 at 202.180.nnn.nnn>;tag=as1370b1ab
To: <sip:6499777777 at domain.co.nz>
Contact: <sip:6494444444 at 202.180.nnn.nnn>
Call-ID: 1fd7e9c847bada25357102fc6173f7f8 at 202.180.nnn.nnn
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 69
Date: Mon, 02 Apr 2007 03:37:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 338
v=0
o=root 11402 11402 IN IP4 202.180.nnn.nnn
s=session
c=IN IP4 202.180.nnn.nnn
t=0 0
m=audio 39686 RTP/AVP 18 97 3 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:97 iLBC/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
--- (15 headers 15 lines) ---
Using INVITE request as basis request - 1fd7e9c847bada25357102fc6173f7f8 at 202.180.nnn.nnn
Sending to 147.202.nnn.nnn : 5060 (non-NAT)
Found peer 'DLS'
Found RTP audio format 18
Found RTP audio format 97
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 202.180.nnn.nnn:39686
Found description format G729
Found description format iLBC
Found description format GSM
Found description format PCMU
Found description format PCMA
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x50e (gsm|ulaw|alaw|g729|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 6499777777 in from-trunk (domain 203.89.nnn.nnn)
list_route: hop: <sip:147.202.nnn.nnn;ftag=as1370b1ab;lr=on>
Transmitting (no NAT) to 147.202.nnn.nnn:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bK22ab.697375a4.0;received=147.202.nnn.nnn
Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK4cf2bb78;rport=5060
From: "6494444444" <sip:6494444444 at 202.180.nnn.nnn>;tag=as1370b1ab
To: <sip:6499777777 at domain.co.nz>
Call-ID: 1fd7e9c847bada25357102fc6173f7f8 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-Length: 0

---
    -- Goto (ivr-3,s,1)
    -- Executing Set("SIP/6499777777-b7908550", "LOOPCOUNT=0") in new stack
    -- Executing Set("SIP/6499777777-b7908550", "__DIR-CONTEXT=11000111000") in new stack
    -- Executing Answer("SIP/6499777777-b7908550", "") in new stack
We're at 203.89.nnn.nnn port 15804
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 147.202.nnn.nnn:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bK22ab.697375a4.0;received=147.202.nnn.nnn
Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK4cf2bb78;rport=5060
Record-Route: <sip:147.202.nnn.nnn;ftag=as1370b1ab;lr=on>
From: "6494444444" <sip:6494444444 at 202.180.nnn.nnn>;tag=as1370b1ab
To: <sip:6499777777 at domain.co.nz>;tag=as7ecf44d1
Call-ID: 1fd7e9c847bada25357102fc6173f7f8 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 15804 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 - - - -
---
    -- Executing Wait("SIP/6499777777-b7908550", "1") in new stack
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=as1370b1ab;lr=on>
Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=0
Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK5ba4f251;rport=5060
From: "6494444444" <sip:6494444444 at 202.180.nnn.nnn>;tag=as1370b1ab
To: <sip:6499777777 at domain.co.nz>;tag=as7ecf44d1
Contact: <sip:6494444444 at 202.180.nnn.nnn>
Call-ID: 1fd7e9c847bada25357102fc6173f7f8 at 202.180.nnn.nnn
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 69
Content-Length: 0

--- (12 headers 0 lines) ---
    -- Executing Set("SIP/6499777777-b7908550", "TIMEOUT(digit)=3") in new stack
    -- Digit timeout set to 3
    -- Executing Set("SIP/6499777777-b7908550", "TIMEOUT(response)=10") in new stack
    -- Response timeout set to 10
    -- Executing BackGround("SIP/6499777777-b7908550", "custom/11000111000-welcome") in new stack
    -- Playing 'custom/11000111000-welcome' (language 'nz')
capetown*CLI> 
<-- SIP read from 147.202.nnn.nnn:5060: 
BYE sip:6499777777 at 203.89.nnn.nnn SIP/2.0
Record-Route: <sip:147.202.nnn.nnn;ftag=as1370b1ab;lr=on>
Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bK32ab.feee6b67.0
Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK7916f637;rport=5060
From: "6494444444" <sip:6494444444 at 202.180.nnn.nnn>;tag=as1370b1ab
To: <sip:6499777777 at domain.co.nz>;tag=as7ecf44d1
Contact: <sip:6494444444 at 202.180.nnn.nnn>
Call-ID: 1fd7e9c847bada25357102fc6173f7f8 at 202.180.nnn.nnn
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 69
Content-Length: 0

--- (12 headers 0 lines) ---
Sending to 147.202.nnn.nnn : 5060 (non-NAT)
Transmitting (no NAT) to 147.202.nnn.nnn:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 147.202.nnn.nnn;branch=z9hG4bK32ab.feee6b67.0;received=147.202.nnn.nnn
Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK7916f637;rport=5060
Record-Route: <sip:147.202.nnn.nnn;ftag=as1370b1ab;lr=on>
From: "6494444444" <sip:6494444444 at 202.180.nnn.nnn>;tag=as1370b1ab
To: <sip:6499777777 at domain.co.nz>;tag=as7ecf44d1
Call-ID: 1fd7e9c847bada25357102fc6173f7f8 at 202.180.nnn.nnn
CSeq: 103 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:6499777777 at 203.89.nnn.nnn>
Content-Length: 0

---
  == Spawn extension (ivr-3, s, 7) exited non-zero on 'SIP/6499777777-b7908550'
    -- Executing Hangup("SIP/6499777777-b7908550", "") in new stack
  == Spawn extension (ivr-3, h, 1) exited non-zero on 'SIP/6499777777-b7908550'
Destroying call '1fd7e9c847bada25357102fc6173f7f8 at 202.180.nnn.nnn'
capetown*CLI> sip no debug
SIP Debugging Disabled
capetown*CLI>


Cameron


	
	
		
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