[OpenSIPS-Users] Transfer issue
Anca Vamanu
anca at opensips.org
Mon Nov 9 10:33:46 CET 2009
Hi Jeff,
It seems that the b2b module just does simple forward of REFER request
in your setup.
Have you loaded the refer scenario? You can find it here:
http://www.opensips.org/Resources/B2buaTutorial#toc15. You have to put
in in a file and then set the 'script_scenario' parameter to the path of
the file:
modparam("b2b_logic", "script_scenario", "path_to_scenario_refer.xml")
The you have to call the b2b_init_request function with the "refer"
parameter:
b2b_init_request("refer");
Regards,
Anca
Jeff Kronlage wrote:
> Anca,
>
> Thanks again for your work on this. I've gotten the b2b modules working and I'm attempting to use the REFER scenario, but I'm having some confusion regarding how a REFER with a B2BUA should be handled.
>
> My test environment looks like this:
>
> UA1 (softphone) ---INVITE--> [Opensips] ---PSTN_GATEWAY(UA2)--> POTS Phone
> ..... (Session progress/OK/etc) ....
> UA1 (softphone) ---ACK--> [Opensips] ---> PSTN_GATEWAY(UA2)
>
> At this point, the first call is setup.
>
> UA1 (softphone) ---REFER UA2 to UA3--> [Opensips] **b2b module called**
> B2B ---REFER B2B to UA3--> PSTN_GATEWAY (UA2)
> B2B <--404 NOT FOUND-- PSTN_GATEWAY (UA2)
> B2B ---404 NOT FOUND-> UA1 (softphone)
>
> Obviously this fails. Note this same problem occurs on two completely separate gateways with different hardware. My questions are:
>
> 1) Shouldn't the b2b scenario send an INVITE off to UA3, wait for the OK/ACK, then simply REFER UA2 to UA3? It seems to me that a B2B -> UA3 refer (which is what I appear to be getting) is out-of-dialog, and many gateways can't start a dialog with a REFER anyway.
> 2) If not, any ideas on what I'm doing wrong?
>
> The pertinent parts of my sip dump are below, beginning with the first REFER, are below:
>
> REFER sip:18002441111 at 208.94.157.10:5060;transport=udp SIP/2.0
> Via: SIP/2.0/UDP 70.57.173.114:63390;branch=z9hG4bK-d8754z-501a700165366947-1---d8754z-;rport
> Max-Forwards: 70
> Route: <sip:64.111.16.50;lr;ftag=a837e85e;did=847.0253b4b4>
> Contact: <sip:7194760273 at 70.57.173.114:63390>
> To: <sip:18002441111 at myproxyabc:5060>;tag=b9d5ed0-13c4-4af6e7b5-46585e8b-24ef0427
> From: "3CXPhone"<sip:7194760273 at myproxyabc.com:5060>;tag=a837e85e
> Call-ID: NDY4YWZhMzRjMDQzZmM2MTU0YTg5YzRlZmFlMzU5NDc.
> CSeq: 4 REFER
> Proxy-Authorization: <omitted>
> User-Agent: 3CXVoipPhone 4.0.9530.0
> Refer-To: <sip:18887779569 at prxdev.sip.data102.com:5060>
> Referred-By: "3CXPhone"<sip:7194760273 at prxdev.sip.data102.com:5060>
> Content-Length: 0
>
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 70.57.173.114:63390;branch=z9hG4bK-d8754z-501a700165366947-1---d8754z-;rport=63390
> To: <sip:18002441111 at myproxyabc.com:5060>;tag=b9d5ed0-13c4-4af6e7b5-46585e8b-24ef0427
> From: "3CXPhone"<sip:7194760273 at myproxyabc.com:5060>;tag=a837e85e
> Call-ID: NDY4YWZhMzRjMDQzZmM2MTU0YTg5YzRlZmFlMzU5NDc.
> CSeq: 4 REFER
> Server: OpenSIPS (1.6.1-notls (i386/linux))
> Content-Length: 0
>
> REFER sip:18002441111 at 208.94.157.10:5060 SIP/2.0
> Via: SIP/2.0/UDP 64.111.16.50;branch=z9hG4bK7ea5.271c79b2.0
> To: sip:18002441111 at 208.94.157.10:5060
> From: <sip:7194760273 at myproxyabc.com:5060>;tag=b9952cfdcb0f3026fcffe5bf74779956-dca7
> CSeq: 2 REFER
> Call-ID: B2B.462.0.1257695261
> Content-Length: 0
> User-Agent: OpenSIPS (1.6.1-notls (i386/linux))
> Contact: <sip:opensips at myproxyabc.com:5060>
>
> SIP/2.0 404 Not Found
> From: <sip:7194760273 at myproxyabc.com:5060>;tag=b9952cfdcb0f3026fcffe5bf74779956-dca7
> To: <sip:18002441111 at 208.94.157.10:5060>;tag=b9d5ed0-13c4-4af6e7c2-465890d5-3ca2cb05
> Call-ID: B2B.462.0.1257695261
> CSeq: 2 REFER
> Via: SIP/2.0/UDP 64.111.16.50;branch=z9hG4bK7ea5.271c79b2.0
> Content-Length: 0
>
> As always, help is much appreciated!
>
> Jeff K
>
> -----Original Message-----
> From: users-bounces at lists.opensips.org [mailto:users-bounces at lists.opensips.org] On Behalf Of Anca Vamanu
> Sent: Friday, November 06, 2009 9:49 AM
> To: OpenSIPS users mailling list
> Subject: Re: [OpenSIPS-Users] Transfer issue
>
> Hi,
>
> The REFER handing support has been added in B2BUA. Please update from
> svn to use this feature.
> To enable it you have to load a simple scenario document that describes
> the behavior of the B2BUA when a REFER message is received and then call
> b2b_init_request("refer") for the initial Invite message.
> I have also updated the documentation page and you can find there also
> the scenario document for this feature:
> http://www.opensips.org/Resources/B2buaTutorial#toc15.
>
> Regards,
> Anca
>
>
> Jeff Kronlage wrote:
>
>> Hi Bogdan,
>>
>> Thanks for the fantastic news.
>>
>> I don't suppose you have any samples of how to interpret a REFER and perform a transfer?
>>
>> I've started pouring through the documentation for the B2BUA, but I'm still grinding through it :)
>>
>> Thanks,
>>
>> Jeff
>>
>> -----Original Message-----
>> From: users-bounces at lists.opensips.org [mailto:users-bounces at lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu
>> Sent: Saturday, October 24, 2009 11:18 AM
>> To: OpenSIPS users mailling list
>> Subject: Re: [OpenSIPS-Users] Transfer issue
>>
>> Hi Iñaki,
>>
>> Actually this is why the B2BUA module was designed in opensips - in most
>> of the cases you need to control/change the dialog(s) but without any
>> media dependencies/penalties (like you have now in most of the IP-PBXs).
>>
>> So you actually can have a highly scalable signalling B2BUA - the
>> opensips module could be used to locally (on opensips) interpret the
>> REFER and do the call transfer, totally transparent to the other party.
>>
>> Regards,
>> Bogdan
>>
>> Iñaki Baz Castillo wrote:
>>
>>
>>> El Sábado, 24 de Octubre de 2009, Jeff Kronlage escribió:
>>>
>>>
>>>
>>>> Our setup has been initially
>>>> engineered for thousands of concurrent calls, and we're hoping to avoid
>>>> having a dozen Asterisk machines :)
>>>>
>>>>
>>>>
>>> What you are looking for is the dream all want: a scalable SIP B2BUA (no media
>>> handling), so a cluster of these B2BUA's would be located behind a proxy
>>> (which does load balancing and failover). And it would be greater if the B2BUA
>>> share information (about current dialogs and so) in some way (memcache? common
>>> database?...).
>>>
>>> You could implement it with SipServlets (see Sailin SIP application server or
>>> others), or FreeSwitch which allows calls without handling the media...
>>> Of course, Asterisk is not the most suitable solution: it involves media
>>> handling ("canreinvite" is a hack), it has a very poor SIP stack... and
>>> basically it's designed to be a single PBX box.
>>>
>>>
>>>
>>>
>>>
>>>
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>
>
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