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Summary:ASTERISK-11471: Asterisk 1.6-beta3 does not follow sip redirect using sip/tcp
Reporter:Olaf Holthausen (oholthau)Labels:
Date Opened:2008-02-19 05:52:21.000-0600Date Closed:2008-02-19 09:02:33.000-0600
Priority:MajorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/Interoperability
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:
Description:I tried to connect Asterisk 1.6 beta3 to a Microsoft Speech Server 2007.

Microsoft Speech Server is by default using a sip redirect to another sip port to the same machine. Asterisk is always trying to use the default port (5060) instead of the new one.

Microsoft SPeech Server is using the same IVR stack as Exchange 2007 Unified Messaging.

I assume the Microsoft Solutions become quite popular in the future..

Best Regards
Olaf.

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

Here is the sip debug trace:


   -- Executing [00@asteriske1:1] Dial("SIP/10.25.31.142-08630fb0", "sip/20@10.25.31.141") in new stack
 == Using SIP RTP CoS mark 5
Audio is at 10.25.31.130 port 19064
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.25.31.141:5060:
INVITE sip:20@10.25.31.141 SIP/2.0
Via: SIP/2.0/TCP 10.25.31.130:5060;branch=z9hG4bK66e879df;rport
Max-Forwards: 70
From: "44" <sip:44@10.25.31.130>;tag=as0543e777
To: <sip:20@10.25.31.141>
Contact: <sip:44@10.25.31.130:5060;transport=TCP>
Call-ID: 1168b68c6a4ee209112d4fd40b153d05@10.25.31.130
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0-beta3
Date: Tue, 19 Feb 2008 11:43:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 432935516 432935516 IN IP4 10.25.31.130
s=Asterisk PBX 1.6.0-beta3
c=IN IP4 10.25.31.130
t=0 0
m=audio 19064 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
   -- Called 20@10.25.31.141

<--- SIP read from TCP://10.25.31.141:5060 --->
SIP/2.0 100 Trying
FROM: "44"<sip:44@10.25.31.130>;tag=as0543e777
TO: <sip:20@10.25.31.141>
CSEQ: 102 INVITE
CALL-ID: 1168b68c6a4ee209112d4fd40b153d05@10.25.31.130
VIA: SIP/2.0/TCP 10.25.31.130:5060;branch=z9hG4bK66e879df;rport
CONTENT-LENGTH: 0


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

<--- SIP read from TCP://10.25.31.141:5060 --->
SIP/2.0 302 Moved Temporarily
FROM: "44"<sip:44@10.25.31.130>;tag=as0543e777
TO: <sip:20@10.25.31.141>;tag=3ab9584f1
CSEQ: 102 INVITE
CALL-ID: 1168b68c6a4ee209112d4fd40b153d05@10.25.31.130
VIA: SIP/2.0/TCP 10.25.31.130:5060;branch=z9hG4bK66e879df;rport
CONTACT: <sip:20@10.25.31.141:1048;transport=Tcp;maddr=10.25.31.141;x-mss-call-id=1168b68c6a4ee209112d4fd40b153d05%4010.25.31.130>
CONTENT-LENGTH: 0
SERVER: RTCC/3.0.0.0


<------------->
--- (9 headers 0 lines) ---
   -- Got SIP response 302 "Moved Temporarily" back from 10.25.31.141
   -- Now forwarding SIP/10.25.31.142-08630fb0 to 'SIP/20@10.25.31.141:1048' (thanks to SIP/10.25.31.141-08628878)
 == Using SIP RTP CoS mark 5
Audio is at 10.25.31.130 port 13808
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.25.31.141:5060:
INVITE sip:20@10.25.31.141 SIP/2.0
Via: SIP/2.0/TCP 10.25.31.130:5060;branch=z9hG4bK67449e2e;rport
Max-Forwards: 70
From: "44" <sip:44@10.25.31.130>;tag=as7047f18e
To: <sip:20@10.25.31.141>
Contact: <sip:44@10.25.31.130:5060;transport=TCP>
Call-ID: 03b618ed05ebd123567d5a735a375b84@10.25.31.130
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0-beta3
Date: Tue, 19 Feb 2008 11:43:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 87528796 87528796 IN IP4 10.25.31.130
s=Asterisk PBX 1.6.0-beta3
c=IN IP4 10.25.31.130
t=0 0
m=audio 13808 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
Transmitting (no NAT) to 10.25.31.141:5060:
ACK sip:20@10.25.31.141 SIP/2.0
Via: SIP/2.0/TCP 10.25.31.130:5060;branch=z9hG4bK66e879df;rport
Max-Forwards: 70
From: "44" <sip:44@10.25.31.130>;tag=as0543e777
To: <sip:20@10.25.31.141>;tag=3ab9584f1
Contact: <sip:44@10.25.31.130:5060;transport=TCP>
Call-ID: 1168b68c6a4ee209112d4fd40b153d05@10.25.31.130
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.0-beta3
Content-Length: 0


---

<--- SIP read from TCP://10.25.31.141:5060 --->
SIP/2.0 100 Trying
FROM: "44"<sip:44@10.25.31.130>;tag=as7047f18e
TO: <sip:20@10.25.31.141>
CSEQ: 102 INVITE
CALL-ID: 03b618ed05ebd123567d5a735a375b84@10.25.31.130
VIA: SIP/2.0/TCP 10.25.31.130:5060;branch=z9hG4bK67449e2e;rport
CONTENT-LENGTH: 0


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

<--- SIP read from TCP://10.25.31.141:5060 --->
SIP/2.0 302 Moved Temporarily
FROM: "44"<sip:44@10.25.31.130>;tag=as7047f18e
TO: <sip:20@10.25.31.141>;tag=c436f6fff8
CSEQ: 102 INVITE
CALL-ID: 03b618ed05ebd123567d5a735a375b84@10.25.31.130
VIA: SIP/2.0/TCP 10.25.31.130:5060;branch=z9hG4bK67449e2e;rport
CONTACT: <sip:20@10.25.31.141:1048;transport=Tcp;maddr=10.25.31.141;x-mss-call-id=03b618ed05ebd123567d5a735a375b84%4010.25.31.130>
CONTENT-LENGTH: 0
SERVER: RTCC/3.0.0.0


<------------->
--- (9 headers 0 lines) ---
   -- Got SIP response 302 "Moved Temporarily" back from 10.25.31.141
Transmitting (no NAT) to 10.25.31.141:5060:
ACK sip:20@10.25.31.141 SIP/2.0
Via: SIP/2.0/TCP 10.25.31.130:5060;branch=z9hG4bK67449e2e;rport
Max-Forwards: 70
From: "44" <sip:44@10.25.31.130>;tag=as7047f18e
To: <sip:20@10.25.31.141>;tag=c436f6fff8
Contact: <sip:44@10.25.31.130:5060;transport=TCP>
Call-ID: 03b618ed05ebd123567d5a735a375b84@10.25.31.130
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.0-beta3
Content-Length: 0


---
Really destroying SIP dialog '1168b68c6a4ee209112d4fd40b153d05@10.25.31.130' Method: INVITE


The sip.conf file:
[general]
tcpenable=yes                   ; Enable server for incoming TCP connections (default is yes)
tcpbindaddr=0.0.0.0             ; IP adderss for TCP server to bind to (0.0.0.0 binds to all interfaces)
promiscredir = yes              ; If yes, allows 302 or REDIR to non-local SIP address


[10.25.31.141]
type=friend
host=10.25.31.141
transport=tcp
port=5060
disallow=all
allow=alaw
insecure=port,invite
context=speechy

Comments:By: Joshua C. Colp (jcolp) 2008-02-19 09:02:33.000-0600

Closed as this is a duplicate of issue 11843.