This is the INVITE sent by sipX to extension 9630. Take note of the
following.
1. Contact given by sipXbridge is sip:+***@192.168.2.2:5090
2. Record-Route inserted by the proxy is
sip:192.168.2.2:5060;lr; \
sipXecs-CallDest=AL%2CINT; \
sipXecs-rs=%2Aauth%7E.%2Afrom%7EODg0NzU1NjYx.\
900_ntap%2Aid%7EODQxMC01NDg%60%212feeb13001a96c3aa0337ff5b7ad52d5
INVITE sip:***@192.168.2.215:5060;transport=udp;x-sipX-nonat SIP/2.0
Record-Route:
<sip:192.168.2.2:5060;lr;sipXecs-CallDest=AL%2CINT;sipXecs-rs=%2Aauth%7E.%2Afrom%7EODg0NzU1NjYx.900_ntap%2Aid%7EODQxMC01NDg%60%212feeb13001a96c3aa0337ff5b7ad52d5>
Call-Id: ***@2xx.8x.6x.3x-0
Cseq: 102 INVITE
From: "+1512xxxxxxx" <sip:+***@2xx.8x.6x.3x>;tag=884755661
To: <sip:+***@mydomain.com>
Via: SIP/2.0/UDP 192.168.2.2;branch=z9hG4bK-XX-5573lQWr3kM1kxmqYz5eScEwLQ
Via: SIP/2.0/UDP
192.168.2.2;branch=z9hG4bK-XX-5570rFCDUPiSnEcmm2e9jPzGig~LtZj4yrJmD_vUR1BPFVudQ;id=8410-548
Via: SIP/2.0/UDP
192.168.2.2;branch=z9hG4bK-XX-556aS8mAjSeoiXcrkk_WOyyF`A~Gb6iTJqTxGrjsIy6_sOZvw
Via: SIP/2.0/UDP
192.168.2.2:5090;branch=z9hG4bK9e64cce5cb5d8d1e9811532224296858313831;sipxecs-id=2a38ffa1
Max-Forwards: 16
User-Agent: sipXecs/4.4.0 sipXecs/sipxbridge (Linux)
References:
***@2xx.8x.6x.3x;rel=chain;sipxecs-tag=request-invite-z9hg4bk4a823cdc
Contact: <sip:+***@192.168.2.2:5090;x-sipX-nonat>
Content-Type: application/sdp
Allow: INVITE,BYE,ACK,CANCEL,REFER,OPTIONS,PRACK
Supported: replaces,100rel
Content-Length: 254
Date: Wed, 15 Aug 2012 19:15:59 GMT
Alert-Info: <http://external.call>;info=alert-external;x-line-id=0
Expires: 20
X-Sipx-Handled: X192.168.2.2-60.220.247.5
This is the 200 OK sent back by Cisco. So far so good. It is well
constructed.
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.2;branch=z9hG4bK-XX-5573lQWr3kM1kxmqYz5eScEwLQ
Via: SIP/2.0/UDP
192.168.2.2;branch=z9hG4bK-XX-5570rFCDUPiSnEcmm2e9jPzGig~LtZj4yrJmD_vUR1BPFVudQ;id=8410-548
Via: SIP/2.0/UDP
192.168.2.2;branch=z9hG4bK-XX-556aS8mAjSeoiXcrkk_WOyyF`A~Gb6iTJqTxGrjsIy6_sOZvw
Via: SIP/2.0/UDP
192.168.2.2:5090;branch=z9hG4bK9e64cce5cb5d8d1e9811532224296858313831;sipxecs-id=2a38ffa1
From: "+1512xxxxxxx" <sip:+***@2xx.8x.6x.3x>;tag=884755661
To: <sip:+***@mydomain.com>;tag=000f34d70512009962af5853-0d820f78
Call-ID: ***@2xx.8x.6x.3x-0
Date: Wed, 15 Aug 2012 19:11:55 GMT
CSeq: 102 INVITE
Server: Cisco-CP7940G/8.0
Contact: <sip:***@192.168.2.215:5060;transport=udp>
Record-Route:
<sip:192.168.2.2:5060;lr;sipXecs-CallDest=AL%2CINT;sipXecs-rs=%2Aauth%7E.%2Afrom%7EODg0NzU1NjYx.900_ntap%2Aid%7EODQxMC01NDg%60%212feeb13001a96c3aa0337ff5b7ad52d5>
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Supported: replaces,join,norefersub
Content-Length: 207
Content-Type: application/sdp
Content-Disposition: session;handling=optional
This is the REFER and this is where the trouble is.
1. The REFER is being sent to 60.220.247.5:5080. This is the external
address of sipXbridge.
The contact sent by sipXbridge is sip:+***@192.168.2.2:5090.
This must be the
destination address for mid-dialog request such as REFER.
2. The route header inserted in the REFER is
sip:192.168.2.2:5060;lr;sipXecs-CallDest=AL%2CINT.
This is wrong because it is truncated. Route header must be
sip:192.168.2.2:5060;lr; \
sipXecs-CallDest=AL%2CINT; \
sipXecs-rs=%2Aauth%7E.%2Afrom%7EODg0NzU1NjYx.\
900_ntap%2Aid%7EODQxMC01NDg%60%212feeb13001a96c3aa0337ff5b7ad52d5
REFER sip:60.220.247.5:5080;sipXecs-CallDest=AL%2CINT SIP/2.0
Via: SIP/2.0/UDP 192.168.2.215:5060;branch=z9hG4bK22ea1de8
From: <sip:+***@mydomain.com>;tag=000f34d70512009962af5853-0d820f78
To: "+1512xxxxxxx" <sip:+***@2xx.8x.6x.3x>;tag=884755661
Call-ID: ***@2xx.8x.6x.3x-0
Max-Forwards: 70
Date: Wed, 15 Aug 2012 19:12:09 GMT
CSeq: 102 REFER
User-Agent: Cisco-CP7940G/8.0
Contact: <sip:***@192.168.2.215:5060;transport=udp>
Route: <sip:192.168.2.2:5060;lr;sipXecs-CallDest=AL%2CINT>
Refer-To:
<sip:***@mydomain.com?Replaces=000f34d7-05120006-1e13d7f6-68b8447a%40192.168.2.215%3Bto-tag%3D001da21a556300f5674f3198-2a615a19%3Bfrom-tag%3D000f34d70512009a5d3e90f4-22cea73f>
Referred-By: <sip:***@mydomain.com>
Content-Length: 0
This REFER will be rejected by sipXbridge because it's as funky as it
could get.
Hi Joegen,
Here's a trace of two external calls to the Cisco phone I made. One was put on hold and the next one a transfer.
Ly Tran
-----Original Message-----
Sent: Tuesday, August 14, 2012 8:04 PM
To: Discussion list for users of sipXecs software
Cc: Ly Tran
Subject: Re: [sipx-users] Cisco Hold/Resume
You will usually get better results reporting your issues when there is something the developers could look at. If it's an easy fix, there is no reason why it can't be fixed. Send a trace/tcp dump.
Post by Ly TranThe firmware running on the Cisco phones is 8-12. Works only internally. Cisco to Cisco, LG-Nortel to Cisco and vice versa. Only fails when calls are coming in externally. Fails on Hold/Resume and in Transfer mode, presumably because the call is placed on hold and MOH while the transfer is being initiated.
No have not gotten a capture yet, but can get one.
-----Original Message-----
Micciche
Sent: Tuesday, August 14, 2012 1:06 PM
Subject: Re: [sipx-users] Cisco Hold/Resume
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Post by Ly TranI do have a backup, but do not want to go back to 4.2.1.. We did not
have this problem with the last version which was
4.4.0-382.g16f5.x86_64, its with this latest update to 4.4.0-418.
Which firmware are you running on these phones?
Do you have any traces, logs, tcpdump of the problem?
Does the problem occur only with internal calls? external?
- --
==================================================================
Red Hat, Inc. http://www.redhat.com
Senior Communications Engineer X (81) 44554
+1.919.754.4554 Key: 65F90FE1
==================================================================
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Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/
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