@ All, <br><br>This has very exciting possibilities for me. I'm definitely going to look at the <br>b2b module and test this out. I will share my findings. <br><br>regards<br><br><div class="gmail_quote">On Thu, Mar 18, 2010 at 10:32 AM, Bogdan-Andrei Iancu <span dir="ltr"><<a href="mailto:bogdan@voice-system.ro">bogdan@voice-system.ro</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">Hi Jeff,<br>
<br>
as opensips will act as b2b, your call will be actually split in 2 calls<br>
(from SIP point of view) - a call C1 from GW to opensips and another one<br>
C2 from opensips to UAC. So at re-INVITE time, opensips b2b will hung up<br>
C2 and replace it with a C3 to a new destination, bridging it with C1<br>
<br>
Regards,<br>
<font color="#888888">Bogdan<br>
</font><div><div></div><div class="h5"><br>
Jeff Kronlage wrote:<br>
> I'm confused on this as well - wouldn't you be effectively placing two<br>
> calls (one via a non-T38 gateway, one via a T38 gateway) to the same<br>
> destination? Figuring that most T38 is going to terminate to a single<br>
> analog device, I would think that were this possible at a SIP level, the<br>
> device would already be "busy" before the second call came in as fax<br>
> machines don't typically drop the line very rapidly?<br>
><br>
> Jeff<br>
><br>
> -----Original Message-----<br>
> From: <a href="mailto:users-bounces@lists.opensips.org">users-bounces@lists.opensips.org</a><br>
> [mailto:<a href="mailto:users-bounces@lists.opensips.org">users-bounces@lists.opensips.org</a>] On Behalf Of Bogdan-Andrei<br>
> Iancu<br>
> Sent: Wednesday, March 17, 2010 11:23 AM<br>
> To: OpenSIPS users mailling list<br>
> Subject: Re: [OpenSIPS-Users] T.38 detection/redirect in OpenSIPS<br>
><br>
> right, that is exactly what the b2b is up to do - to be able (at<br>
> signalling level) to manipulate the call legs<br>
><br>
> Regards,<br>
> Bogdan<br>
><br>
> Brett Nemeroff wrote:<br>
><br>
>> Bogdan,<br>
>> But at this point, you are now playing with a dialg that is already<br>
>> connected to an endpoint. You'd need to drop the first call to<br>
>> establish a new call with the reinvite. Right?<br>
>> -Brett<br>
>><br>
>> On Mar 17, 2010, at 11:50 AM, Bogdan-Andrei Iancu<br>
>><br>
> <<a href="mailto:bogdan@voice-system.ro">bogdan@voice-system.ro</a><br>
><br>
>> > wrote:<br>
>><br>
>><br>
>><br>
>>> Hi Brett,<br>
>>><br>
>>> Brett Nemeroff wrote:<br>
>>><br>
>>><br>
>>>> I don't think there is any way to do this without an RTP capable<br>
>>>> device in the mix.<br>
>>>><br>
>>>><br>
>>> you do not need to look into RTP as the FAX is advertised in the<br>
>>> re-INVITE (in SDP) - so you can detect it from opensips script by<br>
>>> inspecting the SDP of reINVITES<br>
>>><br>
>>><br>
>>>> What you may be able to do is have asterisk detect that it's a fax,<br>
>>>> then reject it if it is.. I don't know if you can do all that<br>
>>>><br>
> without<br>
><br>
>>>> answering the call.<br>
>>>><br>
>>>><br>
>>> no, you cannnot, as first the call is established (from sip point of<br>
>>> view) as a simple audio call and after that re-negotiated (via<br>
>>> re-INVITE) for FAX<br>
>>><br>
>>><br>
>>>> Then you can forward it back to the proxy if it is a fax with maybe<br>
>>>><br>
> a<br>
><br>
>>>> prefix.<br>
>>>><br>
>>>> A lot of assumptions in there. Would like to hear if you find<br>
>>>> something that works. Not sure if you can SIP Spiral yet in asterisk<br>
>>>> anyway. ;)<br>
>>>><br>
>>>><br>
>>> I do not see the need of Asterisk - maybe with some changes, the b2b<br>
>>> module will be able to handle this - see my prev email.<br>
>>><br>
>>> Regards,<br>
>>> Bogdan<br>
>>><br>
>>><br>
>>><br>
>>>> -Brett<br>
>>>><br>
>>>><br>
>>>> On Wed, Mar 17, 2010 at 10:51 AM, David J. <<a href="mailto:david@styleflare.com">david@styleflare.com</a><br>
>>>> <mailto:<a href="mailto:david@styleflare.com">david@styleflare.com</a>>> wrote:<br>
>>>><br>
>>>> Matt,<br>
>>>><br>
>>>> I am for sure probably wrong, but I think you would need<br>
>>>> Asterisk or<br>
>>>> Variant to Determine that it is a Fax Call,<br>
>>>> I dont think UAC's send T38 information without negotiating with<br>
>>>> the<br>
>>>> other side who request that it is capable, then it brings you to<br>
>>>> Jeff's<br>
>>>> answer.<br>
>>>><br>
>>>> See above.<br>
>>>><br>
>>>><br>
>>>> Matthew S. Crocker wrote:<br>
>>>><br>
>>>><br>
>>>>> Can OpenSIPS make routing decisions based on the SDP information<br>
>>>>><br>
>>>>><br>
>>>> in an INVITE?<br>
>>>><br>
>>>><br>
>>>>> Lets say I have the following config<br>
>>>>><br>
>>>>> PSTN -> t.38 Gateway -> OpenSIPS -> UserAgent<br>
>>>>><br>
>>>>> I have a TN from the PSTN routed to the UserAgent, I'd like to<br>
>>>>><br>
>>>>><br>
>>>> provide a service so the user can use the TN for both voice &<br>
>>>> faxing.<br>
>>>><br>
>>>><br>
>>>>> Voice call goes through normally (g.711 g.729 codec)<br>
>>>>><br>
>>>>> Fax call starts off as a normal voice call (INVITE, 180, 183,<br>
>>>>><br>
>>>>><br>
>>>> 200). Once the call is answered the originating end (PSTN)<br>
>>>><br>
> starts<br>
><br>
>>>> sending fax tones. The Gateway hears the fax tones and attempts<br>
>>>><br>
> to<br>
><br>
>>>> RE-INVITE with T.38 in the SDP. I'd like OpenSIPS to see the<br>
>>>><br>
> T.38<br>
><br>
>>>> capability in the SDP and redirect the call to a fax->e-mail<br>
>>>> gateway. So, the 2nd INVITE comes in, OpenSIPS sends the INVITE<br>
>>>> to the fax gateway and a BYE to the user. The fax gateway does a<br>
>>>> 200 and negotiates T.38 with the PSTN gateway.<br>
>>>><br>
>>>><br>
>>>>> I know I can route the call through Asterisk and have it do a<br>
>>>>><br>
>>>>><br>
>>>> quiet answer and listen for the modem sounds. I'd like to avoid<br>
>>>> using Asterisk for all RTP traffic and only use it for the fax<br>
>>>> gateway traffic (i.e. once it has been determined to be a fax<br>
>>>> Asterisk steps in and handled the T38 -> E-mail)<br>
>>>><br>
>>>><br>
>>>>> -Matt<br>
>>>>><br>
>>>>><br>
>>>>><br>
<br>
<br>
</div></div>--<br>
<div><div></div><div class="h5">Bogdan-Andrei Iancu<br>
<a href="http://www.voice-system.ro" target="_blank">www.voice-system.ro</a><br>
<br>
<br>
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</div></div></blockquote></div><br><br clear="all"><br>-- <br>TC<br>