ASTERISK-12874: Additional codecs are added to the SDP after a "Moved Temporarily" mesage

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Summary: ASTERISK-12874: Additional codecs are added to the SDP after a "Moved Temporarily" mesage - SIP TCP

Reporter: mattdarnell (mattdarnell) Labels:

Date Opened: 2008-10-10 20:25:43 Date Closed: 2011-06-07 14:02:43

Priority: Major Regression? No

Status: Closed/Complete Components: Channels/chan_sip/CodecHandling

Versions: Frequency of
Occurrence

Related
Issues:

Environment: Attachments: ( 0) full.tar.gz

Description: After receiving a 302 SIP response Asterisk adds additional codecs to the SPD when they are not allowed in sip.conf. The only codec allowed in this example is ulaw.

This is a 1.6.0 system with "0013523: Trouble with Temporarily Moved" applied

-- Got SIP response 302 "Moved Temporarily" back from 10.10.20.31
Transmitting (no NAT) to 10.10.20.31:5060:
ACK sip:3451@10.10.20.31 SIP/2.0
Via: SIP/2.0/TCP 10.10.20.50:5060;branch=z9hG4bK5cdde719;rport
Max-Forwards: 70
From: "4000" <sip:3451@wikitelcom.com>;tag=as43e0f2af
To: <sip:3451@10.10.20.31>;tag=f4177796d8
Contact: <sip:3451@10.10.20.50:5060;transport=TCP>
Call-ID: 3deb3e4b3f6dc02c709391dd7d73f566@wikitelcom.com
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.0
Content-Length: 0

---
-- Now forwarding SIP/4000-b7d5ac38 to 'SIP/3451::::TCP@10.10.20.31:5065' (thanks to SIP/sip-tcp-0831ed70)
== Using SIP RTP CoS mark 5
Audio is at 10.10.20.50 port 15016
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP

****** ADDITIONAL INFORMATION ******

<------------>
Scheduling destruction of SIP dialog 'B4318815@meta.plnium.com' in 32000 ms (Method: OPTIONS)
Really destroying SIP dialog 'D665BE88@meta.plnium.com' Method: OPTIONS
sbc1*CLI>
sbc1*CLI>
sbc1*CLI>
sbc1*CLI>
sbc1*CLI>
sbc1*CLI>
sbc1*CLI>
sbc1*CLI>
sbc1*CLI>
sbc1*CLI>
sbc1*CLI>
sbc1*CLI>
sbc1*CLI>
sbc1*CLI>
sbc1*CLI>
sbc1*CLI>
sbc1*CLI>
sbc1*CLI>
sbc1*CLI>
sbc1*CLI>
sbc1*CLI>
sbc1*CLI>
sbc1*CLI>
sbc1*CLI>
sbc1*CLI>
sbc1*CLI>
sbc1*CLI>
sbc1*CLI>
sbc1*CLI>
sbc1*CLI>
sbc1*CLI>
sbc1*CLI>
sbc1*CLI>
sbc1*CLI>
sbc1*CLI>
sbc1*CLI>
sbc1*CLI>
<--- SIP read from UDP://67.53.192.138:40658 --->
INVITE sip:7000@vm.akamaitel.com SIP/2.0
Via: SIP/2.0/UDP 67.53.192.138:40658;branch=z9hG4bK-d8754z-e916892f8c40c069-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:4000@67.53.192.138:40658>
To: "7000"<sip:7000@vm.akamaitel.com>
From: "4000"<sip:4000@vm.akamaitel.com>;tag=ba50db02
Call-ID: OTNmZmNmZGUxMjc3ZWI4ZjE5MTBhNWRkZWFkNTJlNDA.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1100l stamp 47546
Content-Length: 186

v=0
o=- 0 2 IN IP4 192.168.200.48
s=CounterPath X-Lite 3.0
c=IN IP4 67.53.192.138
t=0 0
m=audio 32640 RTP/AVP 0 101
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv

<------------->
--- (12 headers 9 lines) ---
== Using SIP RTP CoS mark 5
Sending to 67.53.192.138 : 40658 (NAT)
Using INVITE request as basis request - OTNmZmNmZGUxMjc3ZWI4ZjE5MTBhNWRkZWFkNTJlNDA.
Found user '4000' for '4000'

<--- Reliably Transmitting (NAT) to 67.53.192.138:40658 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 67.53.192.138:40658;branch=z9hG4bK-d8754z-e916892f8c40c069-1---d8754z-;received=67.53.192.138;rport=40658
From: "4000"<sip:4000@vm.akamaitel.com>;tag=ba50db02
To: "7000"<sip:7000@vm.akamaitel.com>;tag=as05585d2f
Call-ID: OTNmZmNmZGUxMjc3ZWI4ZjE5MTBhNWRkZWFkNTJlNDA.
CSeq: 1 INVITE
User-Agent: Asterisk PBX 1.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0526f311"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog 'OTNmZmNmZGUxMjc3ZWI4ZjE5MTBhNWRkZWFkNTJlNDA.' in 32000 ms (Method: INVITE)
sbc1*CLI>
<--- SIP read from UDP://67.53.192.138:40658 --->
ACK sip:7000@vm.akamaitel.com SIP/2.0
Via: SIP/2.0/UDP 67.53.192.138:40658;branch=z9hG4bK-d8754z-e916892f8c40c069-1---d8754z-;rport
To: "7000"<sip:7000@vm.akamaitel.com>;tag=as05585d2f
From: "4000"<sip:4000@vm.akamaitel.com>;tag=ba50db02
Call-ID: OTNmZmNmZGUxMjc3ZWI4ZjE5MTBhNWRkZWFkNTJlNDA.
CSeq: 1 ACK
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---

<--- SIP read from UDP://67.53.192.138:40658 --->
INVITE sip:7000@vm.akamaitel.com SIP/2.0
Via: SIP/2.0/UDP 67.53.192.138:40658;branch=z9hG4bK-d8754z-b7584f22e62d164e-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:4000@67.53.192.138:40658>
To: "7000"<sip:7000@vm.akamaitel.com>
From: "4000"<sip:4000@vm.akamaitel.com>;tag=ba50db02
Call-ID: OTNmZmNmZGUxMjc3ZWI4ZjE5MTBhNWRkZWFkNTJlNDA.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1100l stamp 47546
Authorization: Digest username="4000",realm="asterisk",nonce="0526f311",uri="sip:7000@vm.akamaitel.com",response="07bb819aa9db97e6f6b11c9fb26a8bf8",algorithm=MD5
Content-Length: 186

v=0
o=- 0 2 IN IP4 192.168.200.48
s=CounterPath X-Lite 3.0
c=IN IP4 67.53.192.138
t=0 0
m=audio 32640 RTP/AVP 0 101
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv

<------------->
--- (13 headers 9 lines) ---
Sending to 67.53.192.138 : 40658 (NAT)
Using INVITE request as basis request - OTNmZmNmZGUxMjc3ZWI4ZjE5MTBhNWRkZWFkNTJlNDA.
Found user '4000' for '4000'
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 67.53.192.138:32640
Found audio description format telephone-event for ID 101
Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 67.53.192.138:32640
Looking for 7000 in default (domain vm.akamaitel.com)
list_route: hop: <sip:4000@67.53.192.138:40658>

<--- Transmitting (NAT) to 67.53.192.138:40658 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 67.53.192.138:40658;branch=z9hG4bK-d8754z-b7584f22e62d164e-1---d8754z-;received=67.53.192.138;rport=40658
From: "4000"<sip:4000@vm.akamaitel.com>;tag=ba50db02
To: "7000"<sip:7000@vm.akamaitel.com>
Call-ID: OTNmZmNmZGUxMjc3ZWI4ZjE5MTBhNWRkZWFkNTJlNDA.
CSeq: 2 INVITE
User-Agent: Asterisk PBX 1.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:7000@64.75.215.160>
Content-Length: 0

<------------>
-- Executing [7000@default:1] Dial("SIP/4000-b7d5ac38", "SIP/3451@sip-tcp") in new stack
== Using SIP RTP CoS mark 5
Audio is at 10.10.20.50 port 11178
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.10.20.31:5060:
INVITE sip:3451@10.10.20.31 SIP/2.0
Via: SIP/2.0/TCP 10.10.20.50:5060;branch=z9hG4bK5cdde719;rport
Max-Forwards: 70
From: "4000" <sip:3451@wikitelcom.com>;tag=as43e0f2af
To: <sip:3451@10.10.20.31>
Contact: <sip:3451@10.10.20.50:5060;transport=TCP>
Call-ID: 3deb3e4b3f6dc02c709391dd7d73f566@wikitelcom.com
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0
Date: Fri, 10 Oct 2008 21:07:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 257

v=0
o=root 878237844 878237844 IN IP4 10.10.20.50
s=Asterisk PBX 1.6.0
c=IN IP4 10.10.20.50
t=0 0
m=audio 11178 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
-- Called 3451@sip-tcp
sbc1*CLI>
<--- SIP read from TCP://10.10.20.31:5060 --->
SIP/2.0 100 Trying
FROM: "4000"<sip:3451@wikitelcom.com>;tag=as43e0f2af
TO: <sip:3451@10.10.20.31>
CSEQ: 102 INVITE
CALL-ID: 3deb3e4b3f6dc02c709391dd7d73f566@wikitelcom.com
VIA: SIP/2.0/TCP 10.10.20.50:5060;branch=z9hG4bK5cdde719;rport
CONTENT-LENGTH: 0

<------------->
--- (7 headers 0 lines) ---

<--- SIP read from TCP://10.10.20.31:5060 --->
SIP/2.0 302 Moved Temporarily
FROM: "4000"<sip:3451@wikitelcom.com>;tag=as43e0f2af
TO: <sip:3451@10.10.20.31>;tag=f4177796d8
CSEQ: 102 INVITE
CALL-ID: 3deb3e4b3f6dc02c709391dd7d73f566@wikitelcom.com
VIA: SIP/2.0/TCP 10.10.20.50:5060;branch=z9hG4bK5cdde719;rport
CONTACT: <sip:3451@10.10.20.31:5065;transport=TCP>
CONTENT-LENGTH: 0
SERVER: RTCC/3.0.0.0

<------------->
--- (9 headers 0 lines) ---
-- Got SIP response 302 "Moved Temporarily" back from 10.10.20.31
Transmitting (no NAT) to 10.10.20.31:5060:
ACK sip:3451@10.10.20.31 SIP/2.0
Via: SIP/2.0/TCP 10.10.20.50:5060;branch=z9hG4bK5cdde719;rport
Max-Forwards: 70
From: "4000" <sip:3451@wikitelcom.com>;tag=as43e0f2af
To: <sip:3451@10.10.20.31>;tag=f4177796d8
Contact: <sip:3451@10.10.20.50:5060;transport=TCP>
Call-ID: 3deb3e4b3f6dc02c709391dd7d73f566@wikitelcom.com
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.0
Content-Length: 0

---
-- Now forwarding SIP/4000-b7d5ac38 to 'SIP/3451::::TCP@10.10.20.31:5065' (thanks to SIP/sip-tcp-0831ed70)
== Using SIP RTP CoS mark 5
Audio is at 10.10.20.50 port 15016
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.10.20.31:5065:
INVITE sip:3451@10.10.20.31:5065 SIP/2.0
Via: SIP/2.0/TCP 10.10.20.50:5060;branch=z9hG4bK0ec8d3a2;rport
Max-Forwards: 70
From: "4000" <sip:4000@10.10.20.50>;tag=as7a6e267c
To: <sip:3451@10.10.20.31:5065>
Contact: <sip:4000@10.10.20.50:5060;transport=TCP>
Call-ID: 638538aa57bb3ad537820ce46e2e4dce@10.10.20.50
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0
Date: Fri, 10 Oct 2008 21:07:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Type: application/sdp

Content-Length: 304

v=0
o=root 999779971 999779971 IN IP4 10.10.20.50
s=Asterisk PBX 1.6.0
c=IN IP4 10.10.20.50
t=0 0
m=audio 15016 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
sbc1*CLI>
<--- SIP read from TCP://10.10.20.31:5065 --->
SIP/2.0 100 Trying
FROM: "4000"<sip:4000@10.10.20.50>;tag=as7a6e267c
TO: <sip:3451@10.10.20.31:5065>
CSEQ: 102 INVITE
CALL-ID: 638538aa57bb3ad537820ce46e2e4dce@10.10.20.50
VIA: SIP/2.0/TCP 10.10.20.50:5060;branch=z9hG4bK0ec8d3a2;rport
CONTENT-LENGTH: 0

<------------->
--- (7 headers 0 lines) ---

<--- SIP read from TCP://10.10.20.31:5065 --->
SIP/2.0 180 Ringing
FROM: "4000"<sip:4000@10.10.20.50>;tag=as7a6e267c
TO: <sip:3451@10.10.20.31:5065>;epid=86977CFD96;tag=bcc3a7707c
CSEQ: 102 INVITE
CALL-ID: 638538aa57bb3ad537820ce46e2e4dce@10.10.20.50
VIA: SIP/2.0/TCP 10.10.20.50:5060;branch=z9hG4bK0ec8d3a2;rport
CONTENT-LENGTH: 0
SERVER: RTCC/3.0.0.0

<------------->
--- (8 headers 0 lines) ---
-- SIP/10.10.20.31:5065-083068a8 is ringing

<--- Transmitting (NAT) to 67.53.192.138:40658 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 67.53.192.138:40658;branch=z9hG4bK-d8754z-b7584f22e62d164e-1---d8754z-;received=67.53.192.138;rport=40658
From: "4000"<sip:4000@vm.akamaitel.com>;tag=ba50db02
To: "7000"<sip:7000@vm.akamaitel.com>;tag=as7b9966b3
Call-ID: OTNmZmNmZGUxMjc3ZWI4ZjE5MTBhNWRkZWFkNTJlNDA.
CSeq: 2 INVITE
User-Agent: Asterisk PBX 1.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:7000@64.75.215.160>
Content-Length: 0

<------------>
sbc1*CLI>
<--- SIP read from TCP://10.10.20.31:5065 --->
SIP/2.0 200 OK
FROM: "4000"<sip:4000@10.10.20.50>;tag=as7a6e267c
TO: <sip:3451@10.10.20.31:5065>;epid=86977CFD96;tag=bcc3a7707c
CSEQ: 102 INVITE
CALL-ID: 638538aa57bb3ad537820ce46e2e4dce@10.10.20.50
VIA: SIP/2.0/TCP 10.10.20.50:5060;branch=z9hG4bK0ec8d3a2;rport
CONTACT: <sip:UM2.Makai.local:5065;transport=Tcp;maddr=10.10.20.31>;automata
CONTENT-LENGTH: 193
CONTENT-TYPE: application/sdp
ALLOW: UPDATE
SERVER: RTCC/3.0.0.0
ALLOW: Ack, Cancel, Bye,Invite,Message,Info,Service,Options,BeNotify

v=0
o=- 0 0 IN IP4 10.10.20.31
s=Microsoft Exchange Speech Engine
c=IN IP4 10.10.20.31
t=0 0
m=audio 41728 RTP/AVP 0 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

<------------->
--- (12 headers 9 lines) ---
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 10.10.20.31:41728
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.10.20.31:41728
list_route: hop: <sip:UM2.Makai.local:5065;transport=Tcp;maddr=10.10.20.31>
set_destination: Parsing <sip:UM2.Makai.local:5065;transport=Tcp;maddr=10.10.20.31> for address/port to send to
set_destination: set destination to 10.10.20.31, port 5065
Transmitting (no NAT) to 10.10.20.31:5065:
ACK sip:UM2.Makai.local:5065;transport=Tcp;maddr=10.10.20.31 SIP/2.0
Via: SIP/2.0/TCP 10.10.20.50:5060;branch=z9hG4bK67edb323;rport
Max-Forwards: 70
From: "4000" <sip:4000@10.10.20.50>;tag=as7a6e267c
To: <sip:3451@10.10.20.31:5065>;tag=bcc3a7707c
Contact: <sip:4000@10.10.20.50:5060;transport=TCP>
Call-ID: 638538aa57bb3ad537820ce46e2e4dce@10.10.20.50
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.0
Content-Length: 0

---
-- SIP/10.10.20.31:5065-083068a8 answered SIP/4000-b7d5ac38
Audio is at 64.75.215.160 port 10914
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 67.53.192.138:40658 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 67.53.192.138:40658;branch=z9hG4bK-d8754z-b7584f22e62d164e-1---d8754z-;received=67.53.192.138;rport=40658
From: "4000"<sip:4000@vm.akamaitel.com>;tag=ba50db02
To: "7000"<sip:7000@vm.akamaitel.com>;tag=as7b9966b3
Call-ID: OTNmZmNmZGUxMjc3ZWI4ZjE5MTBhNWRkZWFkNTJlNDA.
CSeq: 2 INVITE
User-Agent: Asterisk PBX 1.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:7000@64.75.215.160>
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 1356450974 1356450974 IN IP4 64.75.215.160
s=Asterisk PBX 1.6.0
c=IN IP4 64.75.215.160
t=0 0
m=audio 10914 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
-- Native bridging SIP/4000-b7d5ac38 and SIP/10.10.20.31:5065-083068a8
set_destination: Parsing <sip:UM2.Makai.local:5065;transport=Tcp;maddr=10.10.20.31> for address/port to send to
set_destination: set destination to 10.10.20.31, port 5065
Audio is at 10.10.20.50 port 15016
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 10.10.20.31:5065:
INVITE sip:UM2.Makai.local:5065;transport=Tcp;maddr=10.10.20.31 SIP/2.0
Via: SIP/2.0/TCP 10.10.20.50:5060;branch=z9hG4bK61091df4;rport
Max-Forwards: 70
From: "4000" <sip:4000@10.10.20.50>;tag=as7a6e267c
To: <sip:3451@10.10.20.31:5065>;tag=bcc3a7707c
Contact: <sip:4000@10.10.20.50:5060;transport=TCP>
Call-ID: 638538aa57bb3ad537820ce46e2e4dce@10.10.20.50
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 285

v=0
o=root 999779971 999779972 IN IP4 67.53.192.138
s=Asterisk PBX 1.6.0
c=IN IP4 67.53.192.138
t=0 0
m=audio 32640 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 - - - -
a=ptime:20
a=sendrecv

---
sbc1*CLI>
<--- SIP read from TCP://10.10.20.31:5065 --->
SIP/2.0 100 Trying
FROM: "4000"<sip:4000@10.10.20.50>;tag=as7a6e267c
TO: <sip:3451@10.10.20.31:5065>;tag=bcc3a7707c
CSEQ: 103 INVITE
CALL-ID: 638538aa57bb3ad537820ce46e2e4dce@10.10.20.50
VIA: SIP/2.0/TCP 10.10.20.50:5060;branch=z9hG4bK61091df4;rport
CONTENT-LENGTH: 0

<------------->
--- (7 headers 0 lines) ---

<--- SIP read from UDP://67.53.192.138:40658 --->
ACK sip:7000@64.75.215.160 SIP/2.0
Via: SIP/2.0/UDP 67.53.192.138:40658;branch=z9hG4bK-d8754z-5f00562d761d8762-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:4000@67.53.192.138:40658>
To: "7000"<sip:7000@vm.akamaitel.com>;tag=as7b9966b3
From: "4000"<sip:4000@vm.akamaitel.com>;tag=ba50db02
Call-ID: OTNmZmNmZGUxMjc3ZWI4ZjE5MTBhNWRkZWFkNTJlNDA.
CSeq: 2 ACK
User-Agent: X-Lite release 1100l stamp 47546
Authorization: Digest username="4000",realm="asterisk",nonce="0526f311",uri="sip:7000@vm.akamaitel.com",response="07bb819aa9db97e6f6b11c9fb26a8bf8",algorithm=MD5
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
set_destination: Parsing <sip:4000@67.53.192.138:40658> for address/port to send to
set_destination: set destination to 67.53.192.138, port 40658
Audio is at 64.75.215.160 port 10914
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 67.53.192.138:40658:
INVITE sip:4000@67.53.192.138:40658 SIP/2.0
Via: SIP/2.0/UDP 64.75.215.160:5060;branch=z9hG4bK20dd91c6;rport
Max-Forwards: 70
From: "7000"<sip:7000@vm.akamaitel.com>;tag=as7b9966b3
To: "4000"<sip:4000@vm.akamaitel.com>;tag=ba50db02
Contact: <sip:7000@64.75.215.160>
Call-ID: OTNmZmNmZGUxMjc3ZWI4ZjE5MTBhNWRkZWFkNTJlNDA.
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 259

v=0
o=root 1356450974 1356450975 IN IP4 10.10.20.31
s=Asterisk PBX 1.6.0
c=IN IP4 10.10.20.31
t=0 0
m=audio 41728 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
Really destroying SIP dialog '3deb3e4b3f6dc02c709391dd7d73f566@wikitelcom.com' Method: INVITE
sbc1*CLI>
<--- SIP read from TCP://10.10.20.31:5065 --->
SIP/2.0 415 Unsupported Media Type
FROM: "4000"<sip:4000@10.10.20.50>;tag=as7a6e267c
TO: <sip:3451@10.10.20.31:5065>;tag=bcc3a7707c;epid=86977CFD96
CSEQ: 103 INVITE
CALL-ID: 638538aa57bb3ad537820ce46e2e4dce@10.10.20.50
VIA: SIP/2.0/TCP 10.10.20.50:5060;branch=z9hG4bK61091df4;rport
CONTENT-LENGTH: 0
SERVER: RTCC/3.0.0.0

<------------->
--- (8 headers 0 lines) ---
-- Got SIP response 415 "Unsupported Media Type" back from 10.10.20.31
set_destination: Parsing <sip:UM2.Makai.local:5065;transport=Tcp;maddr=10.10.20.31> for address/port to send to
set_destination: set destination to 10.10.20.31, port 5065
Transmitting (no NAT) to 10.10.20.31:5065:
ACK sip:UM2.Makai.local:5065;transport=Tcp;maddr=10.10.20.31 SIP/2.0
Via: SIP/2.0/TCP 10.10.20.50:5060;branch=z9hG4bK61091df4;rport
Max-Forwards: 70
From: "4000" <sip:4000@10.10.20.50>;tag=as7a6e267c
To: <sip:3451@10.10.20.31:5065>;tag=bcc3a7707c
Contact: <sip:4000@10.10.20.50:5060;transport=TCP>
Call-ID: 638538aa57bb3ad537820ce46e2e4dce@10.10.20.50
CSeq: 103 ACK
User-Agent: Asterisk PBX 1.6.0
Content-Length: 0

---
== Spawn extension (default, 7000, 1) exited non-zero on 'SIP/4000-b7d5ac38'
Scheduling destruction of SIP dialog 'OTNmZmNmZGUxMjc3ZWI4ZjE5MTBhNWRkZWFkNTJlNDA.' in 32000 ms (Method: ACK)
sbc1*CLI>
<--- SIP read from UDP://67.53.192.138:40658 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 64.75.215.160:5060;branch=z9hG4bK20dd91c6;rport=5060
Contact: <sip:4000@67.53.192.138:40658>
To: "4000"<sip:4000@vm.akamaitel.com>;tag=ba50db02
From: "7000"<sip:7000@vm.akamaitel.com>;tag=as7b9966b3
Call-ID: OTNmZmNmZGUxMjc3ZWI4ZjE5MTBhNWRkZWFkNTJlNDA.
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1100l stamp 47546
Content-Length: 186

v=0
o=- 0 3 IN IP4 192.168.200.48
s=CounterPath X-Lite 3.0
c=IN IP4 67.53.192.138
t=0 0
m=audio 32640 RTP/AVP 0 101
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv

<------------->
--- (11 headers 9 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 67.53.192.138:32640
Found audio description format telephone-event for ID 101
Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 67.53.192.138:32640
set_destination: Parsing <sip:4000@67.53.192.138:40658> for address/port to send to
set_destination: set destination to 67.53.192.138, port 40658
Transmitting (NAT) to 67.53.192.138:40658:
ACK sip:4000@67.53.192.138:40658 SIP/2.0
Via: SIP/2.0/UDP 64.75.215.160:5060;branch=z9hG4bK5913ff87;rport
Max-Forwards: 70
From: "7000"<sip:7000@vm.akamaitel.com>;tag=as7b9966b3
To: "4000"<sip:4000@vm.akamaitel.com>;tag=ba50db02
Contact: <sip:7000@64.75.215.160>
Call-ID: OTNmZmNmZGUxMjc3ZWI4ZjE5MTBhNWRkZWFkNTJlNDA.
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.0
Content-Length: 0

---
set_destination: Parsing <sip:4000@67.53.192.138:40658> for address/port to send to
set_destination: set destination to 67.53.192.138, port 40658
Reliably Transmitting (NAT) to 67.53.192.138:40658:
BYE sip:4000@67.53.192.138:40658 SIP/2.0
Via: SIP/2.0/UDP 64.75.215.160:5060;branch=z9hG4bK5dc081ee;rport
Max-Forwards: 70
From: "7000"<sip:7000@vm.akamaitel.com>;tag=as7b9966b3
To: "4000"<sip:4000@vm.akamaitel.com>;tag=ba50db02
Call-ID: OTNmZmNmZGUxMjc3ZWI4ZjE5MTBhNWRkZWFkNTJlNDA.
CSeq: 103 BYE
User-Agent: Asterisk PBX 1.6.0
Content-Length: 0

---
Scheduling destruction of SIP dialog 'OTNmZmNmZGUxMjc3ZWI4ZjE5MTBhNWRkZWFkNTJlNDA.' in 32000 ms (Method: ACK)
Really destroying SIP dialog '638538aa57bb3ad537820ce46e2e4dce@10.10.20.50' Method: INVITE

<--- SIP read from UDP://67.53.192.138:40658 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 64.75.215.160:5060;branch=z9hG4bK5dc081ee;rport=5060
Contact: <sip:4000@67.53.192.138:40658>
To: "4000"<sip:4000@vm.akamaitel.com>;tag=ba50db02
From: "7000"<sip:7000@vm.akamaitel.com>;tag=as7b9966b3
Call-ID: OTNmZmNmZGUxMjc3ZWI4ZjE5MTBhNWRkZWFkNTJlNDA.
CSeq: 103 BYE
User-Agent: X-Lite release 1100l stamp 47546
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog 'OTNmZmNmZGUxMjc3ZWI4ZjE5MTBhNWRkZWFkNTJlNDA.' Method: ACK
sbc1*CLI>

Comments: By: Leif Madsen (lmadsen) 2008-10-11 09:24:53

Just to make clear, does this happen without the patch, or only when that patch is applied? The main reason I ask is to determine if it is the patch which is causing the unexpected behaviour, and if so, we should make sure the issue is resolved in the other filed issue. If this happens without the patch, then this is the appropriate place to diagnose this.

Thanks!
By: mattdarnell (mattdarnell) 2008-10-12 02:45:17

I can not test it without the patch because Asterisk will not complete the call without the patch.

-Matt
By: Paul Belanger (pabelanger) 2008-10-12 22:02:12

See attached trace (full.tar.gz). I was able to reproduce the issue using 1.6.0.1.

PB
By: Jason Parker (jparker) 2009-03-17 13:04:58

Have you verified that this is using the same peer to go out after the 302? It looks as though the dst port is changing from 5060 to 5065.

I'm not too well versed in SIP, but would setting insecure=port on the peer perhaps fix this? If no peer is matched, I believe it will use the codec options specified in [general] (or whatever the default is).
By: Joshua C. Colp (jcolp) 2009-04-27 12:30:31

Unfortunately when we get a call forward we use the exact SIP URI they provide. In order to be specific with the codecs offered this would have to make against a peer, which is not done automatically by IP address and port. My suggestion is to make a peer named the IP address that it is being forwarded to and specify your codecs there. chan_sip will then find that peer and use the provided information.