thr3ads.net - asterisk users - [asterisk-users] Asterisk hangs up SIP call after 6 200 retransmits [Mar 2007]

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2007-Mar-29 10:20 UTC

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

I have the following scenario:
PSTN gateway (202.180.nnn.nnn) -> OpenSER 1.0.1 (147.202.nnn.nnn) ->
Asterisk 1.2.16
(203.89.nnn.nnn)
When an incoming call is received, often (but not always) Asterisk repeatedly
sends a SIP 200 OK message and eventually hangs up the call.

sip.conf
[general]
port = 5060           ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0    ; Address to bind to (all addresses on machine)
externip = 203.89.nnn.nnn
disallow=all
allow=ulaw
allow=alaw
language=nz

[DLS]
username=6499777777
type=peer
host=domain.co.nz
context=from-trunk
canreinvite=no

Note that Asterisk registers with proxy:
6499777777:password@conversant.co.nz/6499777777

sip debug peer DLS
<-- SIP read from 147.202.nnn.nnn:5060: 
INVITE sip:6499777777@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=z9hG4bKd0e3.48331fd3.0
Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK3757e55c;rport=5060
From: "6494444444"
<sip:6494444444@202.180.nnn.nnn>;tag=as4917b107
To: <sip:6499777777@domain.co.nz>
Contact: <sip:6494444444@202.180.nnn.nnn>
Call-ID: 79620dc1382184b64681b2e85584ca4d@202.180.nnn.nnn
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 69
Date: Thu, 29 Mar 2007 17:00:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 338
 
v=0
o=root 13636 13636 IN IP4 202.180.nnn.nnn
s=session
c=IN IP4 202.180.nnn.nnn
t=0 0
m=audio 36274 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 -
79620dc1382184b64681b2e85584ca4d@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:36274
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=as4917b107;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=z9hG4bKd0e3.48331fd3.0;received=147.202.nnn.nnn
Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK3757e55c;rport=5060
From: "6494444444"
<sip:6494444444@202.180.nnn.nnn>;tag=as4917b107
To: <sip:6499777777@domain.co.nz>
Call-ID: 79620dc1382184b64681b2e85584ca4d@202.180.nnn.nnn
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:6499777777@203.89.nnn.nnn>
Content-Length: 0
 
 
---
    -- Executing Goto("SIP/6499777777-b791bb60",
"ivr-3|s|1") in new stack
    -- Goto (ivr-3,s,1)
    -- Executing Set("SIP/6499777777-b791bb60",
"LOOPCOUNT=0") in new stack
    -- Executing Set("SIP/6499777777-b791bb60",
"__DIR-CONTEXT=11000111000") in new stack
    -- Executing Answer("SIP/6499777777-b791bb60", "") in
new stack
We're at 203.89.nnn.nnn port 11648
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=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@202.180.nnn.nnn>;tag=as4917b107
To: <sip:6499777777@domain.co.nz>;tag=as7cefaa53
Call-ID: 79620dc1382184b64681b2e85584ca4d@202.180.nnn.nnn
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:6499777777@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 - - - -
 
---
    -- Executing Wait("SIP/6499777777-b791bb60", "1") in new
stack
capetown*CLI> 
<-- SIP read from 147.202.nnn.nnn:5060: 
INVITE sip:6499777777@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=z9hG4bKd0e3.48331fd3.0
Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK3757e55c;rport=5060
From: "6494444444"
<sip:6494444444@202.180.nnn.nnn>;tag=as4917b107
To: <sip:6499777777@domain.co.nz>
Contact: <sip:6494444444@202.180.nnn.nnn>
Call-ID: 79620dc1382184b64681b2e85584ca4d@202.180.nnn.nnn
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 69
Date: Thu, 29 Mar 2007 17:00:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 338
 
v=0
o=root 13636 13636 IN IP4 202.180.nnn.nnn
s=session
c=IN IP4 202.180.nnn.nnn
t=0 0
m=audio 36274 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) ---
Ignoring this INVITE request
We're at 203.89.nnn.nnn port 11648
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=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@202.180.nnn.nnn>;tag=as4917b107
To: <sip:6499777777@domain.co.nz>;tag=as7cefaa53
Call-ID: 79620dc1382184b64681b2e85584ca4d@202.180.nnn.nnn
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:6499777777@203.89.nnn.nnn>
Content-Type: application/sdp
Content-Length: 244
 
v=0
o=root 16300 16301 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@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@202.180.nnn.nnn>;tag=as4917b107
To: <sip:6499777777@domain.co.nz>;tag=as7cefaa53
Contact: <sip:6494444444@202.180.nnn.nnn>
Call-ID: 79620dc1382184b64681b2e85584ca4d@202.180.nnn.nnn
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 69
Content-Length: 0
 
 
--- (12 headers 0 lines) ---
capetown*CLI> 
<-- SIP read from 147.202.nnn.nnn:5060: 
ACK sip:6499777777@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=z9hG4bK31c6be29;rport=5060
From: "6494444444"
<sip:6494444444@202.180.nnn.nnn>;tag=as4917b107
To: <sip:6499777777@domain.co.nz>;tag=as7cefaa53
Contact: <sip:6494444444@202.180.nnn.nnn>
Call-ID: 79620dc1382184b64681b2e85584ca4d@202.180.nnn.nnn
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 69
Content-Length: 0
 
 
--- (12 headers 0 lines) ---
Retransmitting #1 (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@202.180.nnn.nnn>;tag=as4917b107
To: <sip:6499777777@domain.co.nz>;tag=as7cefaa53
Call-ID: 79620dc1382184b64681b2e85584ca4d@202.180.nnn.nnn
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:6499777777@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 - - - -
 
---
    -- Executing Set("SIP/6499777777-b791bb60",
"TIMEOUT(digit)=3") in new stack
    -- Digit timeout set to 3
    -- Executing Set("SIP/6499777777-b791bb60",
"TIMEOUT(response)=10") in new stack
    -- Response timeout set to 10
    -- Executing BackGround("SIP/6499777777-b791bb60",
"custom/11000111000-welcome") in new stack
    -- Playing 'custom/11000111000-welcome' (language 'nz')
capetown*CLI> 
<-- SIP read from 147.202.nnn.nnn:5060: 
ACK sip:6499777777@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=z9hG4bK39ff738b;rport=5060
From: "6494444444"
<sip:6494444444@202.180.nnn.nnn>;tag=as4917b107
To: <sip:6499777777@domain.co.nz>;tag=as7cefaa53
Contact: <sip:6494444444@202.180.nnn.nnn>
Call-ID: 79620dc1382184b64681b2e85584ca4d@202.180.nnn.nnn
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 69
Content-Length: 0
 
 
--- (12 headers 0 lines) ---
Retransmitting #2 (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@202.180.nnn.nnn>;tag=as4917b107
To: <sip:6499777777@domain.co.nz>;tag=as7cefaa53
Call-ID: 79620dc1382184b64681b2e85584ca4d@202.180.nnn.nnn
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:6499777777@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@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=z9hG4bK0a33b51e;rport=5060
From: "6494444444"
<sip:6494444444@202.180.nnn.nnn>;tag=as4917b107
To: <sip:6499777777@domain.co.nz>;tag=as7cefaa53
Contact: <sip:6494444444@202.180.nnn.nnn>
Call-ID: 79620dc1382184b64681b2e85584ca4d@202.180.nnn.nnn
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 69
Content-Length: 0
 
 
--- (12 headers 0 lines) ---
Retransmitting #3 (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@202.180.nnn.nnn>;tag=as4917b107
To: <sip:6499777777@domain.co.nz>;tag=as7cefaa53
Call-ID: 79620dc1382184b64681b2e85584ca4d@202.180.nnn.nnn
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:6499777777@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@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=z9hG4bK69eeeace;rport=5060
From: "6494444444"
<sip:6494444444@202.180.nnn.nnn>;tag=as4917b107
To: <sip:6499777777@domain.co.nz>;tag=as7cefaa53
Contact: <sip:6494444444@202.180.nnn.nnn>
Call-ID: 79620dc1382184b64681b2e85584ca4d@202.180.nnn.nnn
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 69
Content-Length: 0
 
 
--- (12 headers 0 lines) ---
Retransmitting #4 (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@202.180.nnn.nnn>;tag=as4917b107
To: <sip:6499777777@domain.co.nz>;tag=as7cefaa53
Call-ID: 79620dc1382184b64681b2e85584ca4d@202.180.nnn.nnn
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:6499777777@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@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=z9hG4bK277865be;rport=5060
From: "6494444444"
<sip:6494444444@202.180.nnn.nnn>;tag=as4917b107
To: <sip:6499777777@domain.co.nz>;tag=as7cefaa53
Contact: <sip:6494444444@202.180.nnn.nnn>
Call-ID: 79620dc1382184b64681b2e85584ca4d@202.180.nnn.nnn
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 69
Content-Length: 0
 
 
--- (12 headers 0 lines) ---
Retransmitting #5 (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@202.180.nnn.nnn>;tag=as4917b107
To: <sip:6499777777@domain.co.nz>;tag=as7cefaa53
Call-ID: 79620dc1382184b64681b2e85584ca4d@202.180.nnn.nnn
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:6499777777@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@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=z9hG4bK388ee2ac;rport=5060
From: "6494444444"
<sip:6494444444@202.180.nnn.nnn>;tag=as4917b107
To: <sip:6499777777@domain.co.nz>;tag=as7cefaa53
Contact: <sip:6494444444@202.180.nnn.nnn>
Call-ID: 79620dc1382184b64681b2e85584ca4d@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@202.180.nnn.nnn>;tag=as4917b107
To: <sip:6499777777@domain.co.nz>;tag=as7cefaa53
Call-ID: 79620dc1382184b64681b2e85584ca4d@202.180.nnn.nnn
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:6499777777@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@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@202.180.nnn.nnn>;tag=as4917b107
To: <sip:6499777777@domain.co.nz>;tag=as7cefaa53
Contact: <sip:6494444444@202.180.nnn.nnn>
Call-ID: 79620dc1382184b64681b2e85584ca4d@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@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.
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Raj Jain

2007-Mar-29 11:26 UTC

head link

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

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@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@202.180.nnn.nnn>;tag=as4917b107
To: <sip:6499777777@domain.co.nz>;tag=as7cefaa53
Contact: < sip:6494444444@202.180.nnn.nnn>
Call-ID: 79620dc1382184b64681b2e85584ca4d@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;received147.202.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@202.180.nnn.nnn>;tag=as4917b107
To: <sip:6499777777@domain.co.nz>;tag=as7cefaa53
Call-ID: 79620dc1382184b64681b2e85584ca4d@202.180.nnn.nnn
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:6499777777@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@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@202.180.nnn.nnn>;tag=as4917b107
To: <sip:6499777777@domain.co.nz>;tag=as7cefaa53
Contact: <sip:6494444444@202.180.nnn.nnn>
Call-ID: 79620dc1382184b64681b2e85584ca4d@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@202.180.nnn.nnn'
capetown*CLI>

Any advice in resolving this issue would be greatly appreciated.

Regards

Cameron



___________________________________________________________
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http://uk.rd.yahoo.com/evt=44106/*http://mail.yahoo.net/uk
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kjcsb

2007-Mar-29 11:26 UTC

head link

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

I have the following scenario:
PSTN gateway (202.180.nnn.nnn) -> OpenSER 1.0.1 (147.202.nnn.nnn) ->
Asterisk 1.2.16
(203.89.nnn.nnn)
When an incoming call is received, often (but not always) Asterisk repeatedly
sends a SIP 200 OK message and eventually hangs up the call.
sip.conf
[general]
port = 5060           ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0    ; Address to bind to (all addresses on machine)
externip = 203.89.nnn.nnn
disallow=all
allow=ulaw
allow=alaw
language=nz
[DLS]
username=6499777777
type=peer
host=domain.co.nz
context=from-trunk
canreinvite=no
Note that Asterisk registers with proxy:
6499777777:password@conversant.co.nz/6499777777
sip debug peer DLS
<-- SIP read from 147.202.nnn.nnn:5060: 
INVITE sip:6499777777@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=z9hG4bKd0e3.48331fd3.0
Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK3757e55c;rport=5060
From: "6494444444"
<sip:6494444444@202.180.nnn.nnn>;tag=as4917b107
To: <sip:6499777777@domain.co.nz>
Contact: <sip:6494444444@202.180.nnn.nnn>
Call-ID: 79620dc1382184b64681b2e85584ca4d@202.180.nnn.nnn
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 69
Date: Thu, 29 Mar 2007 17:00:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 338
v=0
o=root 13636 13636 IN IP4 202.180.nnn.nnn
s=session
c=IN IP4 202.180.nnn.nnn
t=0 0
m=audio 36274 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 -
79620dc1382184b64681b2e85584ca4d@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:36274
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=as4917b107;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=z9hG4bKd0e3.48331fd3.0;received=147.202.nnn.nnn
Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK3757e55c;rport=5060
From: "6494444444"
<sip:6494444444@202.180.nnn.nnn>;tag=as4917b107
To: <sip:6499777777@domain.co.nz>
Call-ID: 79620dc1382184b64681b2e85584ca4d@202.180.nnn.nnn
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:6499777777@203.89.nnn.nnn>
Content-Length: 0

---
    -- Executing Goto("SIP/6499777777-b791bb60",
"ivr-3|s|1") in new stack
    -- Goto (ivr-3,s,1)
    -- Executing Set("SIP/6499777777-b791bb60",
"LOOPCOUNT=0") in new stack
    -- Executing Set("SIP/6499777777-b791bb60",
"__DIR-CONTEXT=11000111000") in new stack
    -- Executing Answer("SIP/6499777777-b791bb60", "") in
new stack
We're at 203.89.nnn.nnn port 11648
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=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@202.180.nnn.nnn>;tag=as4917b107
To: <sip:6499777777@domain.co.nz>;tag=as7cefaa53
Call-ID: 79620dc1382184b64681b2e85584ca4d@202.180.nnn.nnn
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:6499777777@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 - - - -
---
    -- Executing Wait("SIP/6499777777-b791bb60", "1") in new
stack
capetown*CLI> 
<-- SIP read from 147.202.nnn.nnn:5060: 
INVITE sip:6499777777@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=z9hG4bKd0e3.48331fd3.0
Via: SIP/2.0/UDP 202.180.nnn.nnn:5060;branch=z9hG4bK3757e55c;rport=5060
From: "6494444444"
<sip:6494444444@202.180.nnn.nnn>;tag=as4917b107
To: <sip:6499777777@domain.co.nz>
Contact: <sip:6494444444@202.180.nnn.nnn>
Call-ID: 79620dc1382184b64681b2e85584ca4d@202.180.nnn.nnn
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 69
Date: Thu, 29 Mar 2007 17:00:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 338
v=0
o=root 13636 13636 IN IP4 202.180.nnn.nnn
s=session
c=IN IP4 202.180.nnn.nnn
t=0 0
m=audio 36274 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) ---
Ignoring this INVITE request
We're at 203.89.nnn.nnn port 11648
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=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@202.180.nnn.nnn>;tag=as4917b107
To: <sip:6499777777@domain.co.nz>;tag=as7cefaa53
Call-ID: 79620dc1382184b64681b2e85584ca4d@202.180.nnn.nnn
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:6499777777@203.89.nnn.nnn>
Content-Type: application/sdp
Content-Length: 244
v=0
o=root 16300 16301 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@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@202.180.nnn.nnn>;tag=as4917b107
To: <sip:6499777777@domain.co.nz>;tag=as7cefaa53
Contact: <sip:6494444444@202.180.nnn.nnn>
Call-ID: 79620dc1382184b64681b2e85584ca4d@202.180.nnn.nnn
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 69
Content-Length: 0

--- (12 headers 0 lines) ---
capetown*CLI> 
<-- SIP read from 147.202.nnn.nnn:5060: 
ACK sip:6499777777@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=z9hG4bK31c6be29;rport=5060
From: "6494444444"
<sip:6494444444@202.180.nnn.nnn>;tag=as4917b107
To: <sip:6499777777@domain.co.nz>;tag=as7cefaa53
Contact: <sip:6494444444@202.180.nnn.nnn>
Call-ID: 79620dc1382184b64681b2e85584ca4d@202.180.nnn.nnn
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 69
Content-Length: 0

--- (12 headers 0 lines) ---
Retransmitting #1 (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@202.180.nnn.nnn>;tag=as4917b107
To: <sip:6499777777@domain.co.nz>;tag=as7cefaa53
Call-ID: 79620dc1382184b64681b2e85584ca4d@202.180.nnn.nnn
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:6499777777@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 - - - -
---
    -- Executing Set("SIP/6499777777-b791bb60",
"TIMEOUT(digit)=3") in new stack
    -- Digit timeout set to 3
    -- Executing Set("SIP/6499777777-b791bb60",
"TIMEOUT(response)=10") in new stack
    -- Response timeout set to 10
    -- Executing BackGround("SIP/6499777777-b791bb60",
"custom/11000111000-welcome") in new stack
    -- Playing 'custom/11000111000-welcome' (language 'nz')
capetown*CLI> 
<-- SIP read from 147.202.nnn.nnn:5060: 
ACK sip:6499777777@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=z9hG4bK39ff738b;rport=5060
From: "6494444444"
<sip:6494444444@202.180.nnn.nnn>;tag=as4917b107
To: <sip:6499777777@domain.co.nz>;tag=as7cefaa53
Contact: <sip:6494444444@202.180.nnn.nnn>
Call-ID: 79620dc1382184b64681b2e85584ca4d@202.180.nnn.nnn
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 69
Content-Length: 0

--- (12 headers 0 lines) ---
Retransmitting #2 (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@202.180.nnn.nnn>;tag=as4917b107
To: <sip:6499777777@domain.co.nz>;tag=as7cefaa53
Call-ID: 79620dc1382184b64681b2e85584ca4d@202.180.nnn.nnn
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:6499777777@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@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=z9hG4bK0a33b51e;rport=5060
From: "6494444444"
<sip:6494444444@202.180.nnn.nnn>;tag=as4917b107
To: <sip:6499777777@domain.co.nz>;tag=as7cefaa53
Contact: <sip:6494444444@202.180.nnn.nnn>
Call-ID: 79620dc1382184b64681b2e85584ca4d@202.180.nnn.nnn
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 69
Content-Length: 0

--- (12 headers 0 lines) ---
Retransmitting #3 (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@202.180.nnn.nnn>;tag=as4917b107
To: <sip:6499777777@domain.co.nz>;tag=as7cefaa53
Call-ID: 79620dc1382184b64681b2e85584ca4d@202.180.nnn.nnn
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:6499777777@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@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=z9hG4bK69eeeace;rport=5060
From: "6494444444"
<sip:6494444444@202.180.nnn.nnn>;tag=as4917b107
To: <sip:6499777777@domain.co.nz>;tag=as7cefaa53
Contact: <sip:6494444444@202.180.nnn.nnn>
Call-ID: 79620dc1382184b64681b2e85584ca4d@202.180.nnn.nnn
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 69
Content-Length: 0

--- (12 headers 0 lines) ---
Retransmitting #4 (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@202.180.nnn.nnn>;tag=as4917b107
To: <sip:6499777777@domain.co.nz>;tag=as7cefaa53
Call-ID: 79620dc1382184b64681b2e85584ca4d@202.180.nnn.nnn
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:6499777777@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@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=z9hG4bK277865be;rport=5060
From: "6494444444"
<sip:6494444444@202.180.nnn.nnn>;tag=as4917b107
To: <sip:6499777777@domain.co.nz>;tag=as7cefaa53
Contact: <sip:6494444444@202.180.nnn.nnn>
Call-ID: 79620dc1382184b64681b2e85584ca4d@202.180.nnn.nnn
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 69
Content-Length: 0

--- (12 headers 0 lines) ---
Retransmitting #5 (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@202.180.nnn.nnn>;tag=as4917b107
To: <sip:6499777777@domain.co.nz>;tag=as7cefaa53
Call-ID: 79620dc1382184b64681b2e85584ca4d@202.180.nnn.nnn
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:6499777777@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@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=z9hG4bK388ee2ac;rport=5060
From: "6494444444"
<sip:6494444444@202.180.nnn.nnn>;tag=as4917b107
To: <sip:6499777777@domain.co.nz>;tag=as7cefaa53
Contact: <sip:6494444444@202.180.nnn.nnn>
Call-ID: 79620dc1382184b64681b2e85584ca4d@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@202.180.nnn.nnn>;tag=as4917b107
To: <sip:6499777777@domain.co.nz>;tag=as7cefaa53
Call-ID: 79620dc1382184b64681b2e85584ca4d@202.180.nnn.nnn
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:6499777777@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@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@202.180.nnn.nnn>;tag=as4917b107
To: <sip:6499777777@domain.co.nz>;tag=as7cefaa53
Contact: <sip:6494444444@202.180.nnn.nnn>
Call-ID: 79620dc1382184b64681b2e85584ca4d@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@202.180.nnn.nnn'
capetown*CLI>
Any advice in resolving this issue would be greatly appreciated.
Regards
Cameron


	
	
		
___________________________________________________________ 
New Yahoo! Mail is the ultimate force in competitive emailing. Find out more at
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kjcsb

2007-Apr-01 20:42 UTC

head link

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

>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@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@202.180.nnn.nnn>;tag=as1370b1ab
To: <sip:6499777777@domain.co.nz>
Contact: <sip:6494444444@202.180.nnn.nnn>
Call-ID: 1fd7e9c847bada25357102fc6173f7f8@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@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@202.180.nnn.nnn>;tag=as1370b1ab
To: <sip:6499777777@domain.co.nz>
Call-ID: 1fd7e9c847bada25357102fc6173f7f8@202.180.nnn.nnn
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:6499777777@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@202.180.nnn.nnn>;tag=as1370b1ab
To: <sip:6499777777@domain.co.nz>;tag=as7ecf44d1
Call-ID: 1fd7e9c847bada25357102fc6173f7f8@202.180.nnn.nnn
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:6499777777@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@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@202.180.nnn.nnn>;tag=as1370b1ab
To: <sip:6499777777@domain.co.nz>;tag=as7ecf44d1
Contact: <sip:6494444444@202.180.nnn.nnn>
Call-ID: 1fd7e9c847bada25357102fc6173f7f8@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@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@202.180.nnn.nnn>;tag=as1370b1ab
To: <sip:6499777777@domain.co.nz>;tag=as7ecf44d1
Contact: <sip:6494444444@202.180.nnn.nnn>
Call-ID: 1fd7e9c847bada25357102fc6173f7f8@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@202.180.nnn.nnn>;tag=as1370b1ab
To: <sip:6499777777@domain.co.nz>;tag=as7ecf44d1
Call-ID: 1fd7e9c847bada25357102fc6173f7f8@202.180.nnn.nnn
CSeq: 103 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:6499777777@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@202.180.nnn.nnn'
capetown*CLI> sip no debug
SIP Debugging Disabled
capetown*CLI>


Cameron


	
	
		
___________________________________________________________ 
New Yahoo! Mail is the ultimate force in competitive emailing. Find out more at
the Yahoo! Mail Championships. Plus: play games and win prizes.
http://uk.rd.yahoo.com/evt=44106/*http://mail.yahoo.net/uk

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