[Serusers] SER/PSTN Gateway conf
Kostas Marneris
K.Marneris at otenet.gr
Tue May 15 14:09:46 CEST 2007
Hello,
The call signalling exchange is a good start but it's not enough.
What some people here asked you is a capture of the SIP BYE mesg
which GW sends to SER.
something like that, for example :
BYE sip:user1 at 1.2.3.4:5060 SIP/2.0
Via: SIP/2.0/UDP 5.6.7.8:5060;branch=z9hG4bK3E01596
From: <sip:1234567890 at 5.6.7.8>;tag=12FB3150-15EC
To: <sip:12341234567890 at mydomain.com>;tag=sei-6416
Date: Tue, 15 May 2007 11:42:43 GMT
Call-ID: 3A6DE4DC-21011DC-998FA089-7CE33D49 at 5.6.7.8
User-Agent: Cisco-SIPGateway/IOS-12.x
Max-Forwards: 70
Route: <sip:3.3.3.3;ftag=12FB3150-15EC;lr=on>
Timestamp: 1179229379
CSeq: 102 BYE
Reason: Q.850;cause=16
Content-Length: 0
1.2.3.4 : SIP Phone
5.6.7.8 : GW
3.3.3.3 : SER
1234567890 : PSTN number of the caller (@ PSTN side)
12341234567890 : PSTN number of the callee (@ IP side)
Try to find a way (ngrep/tcpdump) to capture this mesg
or all mesgs for this session .
Kostas
flavio wrote:
> 2007/5/12, Greger Viken Teigre <greger at teigre.com>:
>> Copy the list!
>>
> Greg, all,
> first of all thank you for support and patience.
> I follow the getting started cfg example to edit my ser.cfg.
> I would use my gateway in order to handle calls from and to pstn.
> Here is VoIP Calls graph grep from Ethereal, to explain clearly my problem:
>
> Time | 10.28.52.107 | 10.28.19.202 | 10.28.19.124 |
> |21,624 | INVITE SDP ( g729 g711A g711U) |
> |SIP From: sip:0672028405 at 10.28.52.107
> To:sip:0660522014 at 10.28.19.202
> | |(5060) ------------------> (5060) | |
> |21,624 | 100 trying -- your call is important to us
> | |SIP Status
> | |(5060) <------------------ (5060) | |
> |21,624 | | INVITE SDP ( g729 g711A g711U)
> |SIP Request
> | | |(5060) ------------------> (5060) |
> |21,631 | | 100 Trying|
> |SIP Status
> | | |(5060) <------------------ (5060) |
> |21,633 | | 180 Ringing
> |SIP Status
> | | |(5060) <------------------ (5060) |
> |21,634 | 180 Ringing |
> |SIP Status
> | |(5060) <------------------ (5060) | |
> |22,712 | | 200 OK SDP ( g729)
> |SIP Status
> | | |(5060) <------------------ (5060) |
> |22,712 | 200 OK SDP ( g729) |
> |SIP Status
> | |(5060) <------------------ (5060) | |
> |22,717 | ACK | |
> |SIP Request
> | |(5060) ------------------> (5060) | |
> |22,717 | | ACK |
> |SIP Request
> | | |(5060) ------------------> (5060) |
> |22,717 | | ACK |
> |SIP Request
> | | |(5060) ------------------> (5060) |
> |26,037 | BYE | |
> |SIP Request
> | |(5060) ------------------> (5060) | |
> |26,037 | 404 User Not Found |
> |SIP Status
> | |(5060) <------------------ (5060) | |
> |26,040 | ACK | |
> |SIP Request
> | |(5060) ------------------> (5060) | |
> |26,041 | | ACK |
> |SIP Request
> | | |(5060) ------------------> (5060) |
> |26,041 | | ACK |
> |SIP Request
> | | |(5060) ------------------> (5060) |
> |29,570 | | BYE |
> |SIP Request
> | | |(5060) <------------------ (5060) |
> |29,570 | BYE | |
> |SIP Request
> | |(5060) <------------------ (5060) | |
> |29,573 | 481 Transaction Does Not Exist |
> |SIP Status
> | |(5060) ------------------> (5060) | |
> |29,574 | | 481 Transaction Does Not Exist
> |SIP Status
> | | |(5060) ------------------> (5060) |
>
> 10.28.52.107 is IP for my gateway
> 10.28.19.202 is IP for SER
> 10.28.19.124 is IP for my SIP Phone.
>
> As you can see when Gateway send BYE message, SER does not relay it to
> IP Phone, but reply with 404 Not Found SIP Message. Have you any
> suggestions about this?
>
>
>> I don't understand your problem. The BYE will be caught by the loose
>> route section of the config and relayed. The standard getting started
>> should work fine.
>> g-)
>>
>> ------- Original message -------
>> From: flavio <flavio.patria at gmail.com>
>> Sent: 12.5.'07, 15:09
>>
>> > 2007/5/12, Greger V. Teigre <greger at teigre.com>:
>> > > Flavio,
>> > > SER doesn't reply 200 OK to BYE message, the SIP UA does that. So,
>> you
>> > > are fine, the BYE reaches the UA.
>> > > g-)
>> > >
>> >
>>
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