T.38 detection/redirect in OpenSIPS

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T.38 detection/redirect in OpenSIPS

Matthew S. Crocker

Can OpenSIPS make routing decisions based on the SDP information in an INVITE?

Lets say I have the following config

PSTN -> t.38 Gateway -> OpenSIPS ->  UserAgent

I have a TN from the PSTN routed to the UserAgent,  I'd like to provide a service so the user can use the TN for both voice & faxing.

Voice call goes through normally (g.711 g.729 codec)

Fax call starts off as a normal voice call (INVITE, 180, 183, 200).  Once the call is answered the originating end (PSTN) starts sending fax tones. The Gateway hears the fax tones and attempts to RE-INVITE with T.38 in the SDP.  I'd like OpenSIPS to see the T.38 capability in the SDP and redirect the call to a fax->e-mail gateway.  So,  the 2nd INVITE comes in, OpenSIPS sends the INVITE to the fax gateway and a BYE to the user.  The fax gateway does a 200 and negotiates T.38 with the PSTN gateway.

I know I can route the call through Asterisk and have it do a quiet answer and listen for the modem sounds.  I'd like to avoid using Asterisk for all RTP traffic and only use it for the fax gateway traffic (i.e. once it has been determined to be a fax Asterisk steps in and handled the T38 -> E-mail)

-Matt

--
Matthew S. Crocker
President
Crocker Communications, Inc.
PO BOX 710
Greenfield, MA 01302-0710
http://www.crocker.com
P: 413-746-2760


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Re: T.38 detection/redirect in OpenSIPS

Jeff Pyle
If this is possible at all, I would suspect it might require a rather creative home-grown scenario in the B2BUA module.  Since the call is already established, a simple proxy isn't going to be able to tear down the one side and reestablish it with another endpoint.


- Jeff


On Mar 17, 2010, at 11:43 AM, Matthew S. Crocker wrote:

>
> Can OpenSIPS make routing decisions based on the SDP information in an INVITE?
>
> Lets say I have the following config
>
> PSTN -> t.38 Gateway -> OpenSIPS ->  UserAgent
>
> I have a TN from the PSTN routed to the UserAgent,  I'd like to provide a service so the user can use the TN for both voice & faxing.
>
> Voice call goes through normally (g.711 g.729 codec)
>
> Fax call starts off as a normal voice call (INVITE, 180, 183, 200).  Once the call is answered the originating end (PSTN) starts sending fax tones. The Gateway hears the fax tones and attempts to RE-INVITE with T.38 in the SDP.  I'd like OpenSIPS to see the T.38 capability in the SDP and redirect the call to a fax->e-mail gateway.  So,  the 2nd INVITE comes in, OpenSIPS sends the INVITE to the fax gateway and a BYE to the user.  The fax gateway does a 200 and negotiates T.38 with the PSTN gateway.
>
> I know I can route the call through Asterisk and have it do a quiet answer and listen for the modem sounds.  I'd like to avoid using Asterisk for all RTP traffic and only use it for the fax gateway traffic (i.e. once it has been determined to be a fax Asterisk steps in and handled the T38 -> E-mail)
>
> -Matt
>
> --
> Matthew S. Crocker
> President
> Crocker Communications, Inc.
> PO BOX 710
> Greenfield, MA 01302-0710
> http://www.crocker.com
> P: 413-746-2760
>
>
> _______________________________________________
> Users mailing list
> [hidden email]
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users


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Re: T.38 detection/redirect in OpenSIPS

David J.-2
In reply to this post by Matthew S. Crocker
Matt,

I am for sure probably wrong, but I think you would need Asterisk or
Variant to Determine that it is a Fax Call,
I dont think UAC's send T38 information without negotiating with the
other side who request that it is capable, then it brings you to Jeff's
answer.

See above.


Matthew S. Crocker wrote:

> Can OpenSIPS make routing decisions based on the SDP information in an INVITE?
>
> Lets say I have the following config
>
> PSTN -> t.38 Gateway -> OpenSIPS ->  UserAgent
>
> I have a TN from the PSTN routed to the UserAgent,  I'd like to provide a service so the user can use the TN for both voice & faxing.
>
> Voice call goes through normally (g.711 g.729 codec)
>
> Fax call starts off as a normal voice call (INVITE, 180, 183, 200).  Once the call is answered the originating end (PSTN) starts sending fax tones. The Gateway hears the fax tones and attempts to RE-INVITE with T.38 in the SDP.  I'd like OpenSIPS to see the T.38 capability in the SDP and redirect the call to a fax->e-mail gateway.  So,  the 2nd INVITE comes in, OpenSIPS sends the INVITE to the fax gateway and a BYE to the user.  The fax gateway does a 200 and negotiates T.38 with the PSTN gateway.
>
> I know I can route the call through Asterisk and have it do a quiet answer and listen for the modem sounds.  I'd like to avoid using Asterisk for all RTP traffic and only use it for the fax gateway traffic (i.e. once it has been determined to be a fax Asterisk steps in and handled the T38 -> E-mail)
>
> -Matt
>
>  


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Re: T.38 detection/redirect in OpenSIPS

Brett Nemeroff
I don't think there is any way to do this without an RTP capable device in the mix.

What you may be able to do is have asterisk detect that it's a fax, then reject it if it is.. I don't know if you can do all that without answering the call.

Then you can forward it back to the proxy if it is a fax with maybe a prefix. 

A lot of assumptions in there. Would like to hear if you find something that works. Not sure if you can SIP Spiral yet in asterisk anyway. ;)
-Brett


On Wed, Mar 17, 2010 at 10:51 AM, David J. <[hidden email]> wrote:
Matt,

I am for sure probably wrong, but I think you would need Asterisk or
Variant to Determine that it is a Fax Call,
I dont think UAC's send T38 information without negotiating with the
other side who request that it is capable, then it brings you to Jeff's
answer.

See above.


Matthew S. Crocker wrote:
> Can OpenSIPS make routing decisions based on the SDP information in an INVITE?
>
> Lets say I have the following config
>
> PSTN -> t.38 Gateway -> OpenSIPS ->  UserAgent
>
> I have a TN from the PSTN routed to the UserAgent,  I'd like to provide a service so the user can use the TN for both voice & faxing.
>
> Voice call goes through normally (g.711 g.729 codec)
>
> Fax call starts off as a normal voice call (INVITE, 180, 183, 200).  Once the call is answered the originating end (PSTN) starts sending fax tones. The Gateway hears the fax tones and attempts to RE-INVITE with T.38 in the SDP.  I'd like OpenSIPS to see the T.38 capability in the SDP and redirect the call to a fax->e-mail gateway.  So,  the 2nd INVITE comes in, OpenSIPS sends the INVITE to the fax gateway and a BYE to the user.  The fax gateway does a 200 and negotiates T.38 with the PSTN gateway.
>
> I know I can route the call through Asterisk and have it do a quiet answer and listen for the modem sounds.  I'd like to avoid using Asterisk for all RTP traffic and only use it for the fax gateway traffic (i.e. once it has been determined to be a fax Asterisk steps in and handled the T38 -> E-mail)
>
> -Matt
>
>


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Re: T.38 detection/redirect in OpenSIPS

Bogdan-Andrei Iancu
In reply to this post by Matthew S. Crocker
Hi Matthew,

you do not need to look into the media part, as you can "spot" the FAX
presence via the re-INVITE with T38 codec in SDP (you can detect it from
opensips cfg).

So, maybe using the b2b module for something like:
    - allow the voice call to be setup via the b2b in a transparent way
    - if the re-INVITE wth T38 is received from GW, b2b will close the
leg to the users UA and create a new leg to something able to handle the
fax - of course, the b2b will bridge the existing leg (towards PSTN) and
the new leg.

Regards,
Bogdan


Matthew S. Crocker wrote:

> Can OpenSIPS make routing decisions based on the SDP information in an INVITE?
>
> Lets say I have the following config
>
> PSTN -> t.38 Gateway -> OpenSIPS ->  UserAgent
>
> I have a TN from the PSTN routed to the UserAgent,  I'd like to provide a service so the user can use the TN for both voice & faxing.
>
> Voice call goes through normally (g.711 g.729 codec)
>
> Fax call starts off as a normal voice call (INVITE, 180, 183, 200).  Once the call is answered the originating end (PSTN) starts sending fax tones. The Gateway hears the fax tones and attempts to RE-INVITE with T.38 in the SDP.  I'd like OpenSIPS to see the T.38 capability in the SDP and redirect the call to a fax->e-mail gateway.  So,  the 2nd INVITE comes in, OpenSIPS sends the INVITE to the fax gateway and a BYE to the user.  The fax gateway does a 200 and negotiates T.38 with the PSTN gateway.
>
> I know I can route the call through Asterisk and have it do a quiet answer and listen for the modem sounds.  I'd like to avoid using Asterisk for all RTP traffic and only use it for the fax gateway traffic (i.e. once it has been determined to be a fax Asterisk steps in and handled the T38 -> E-mail)
>
> -Matt
>
>  


--
Bogdan-Andrei Iancu
www.voice-system.ro


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Re: T.38 detection/redirect in OpenSIPS

Bogdan-Andrei Iancu
In reply to this post by Brett Nemeroff
Hi Brett,

Brett Nemeroff wrote:
> I don't think there is any way to do this without an RTP capable
> device in the mix.
you do not need to look into RTP as the FAX is advertised in the
re-INVITE (in SDP) - so you can detect it from opensips script by
inspecting the SDP of reINVITES
>
> What you may be able to do is have asterisk detect that it's a fax,
> then reject it if it is.. I don't know if you can do all that without
> answering the call.
no, you cannnot, as first the call is established (from sip point of
view) as a simple audio call and after that re-negotiated (via
re-INVITE) for FAX
>
> Then you can forward it back to the proxy if it is a fax with maybe a
> prefix.
>
> A lot of assumptions in there. Would like to hear if you find
> something that works. Not sure if you can SIP Spiral yet in asterisk
> anyway. ;)
I do not see the need of Asterisk - maybe with some changes, the b2b
module will be able to handle this - see my prev email.

Regards,
Bogdan

> -Brett
>
>
> On Wed, Mar 17, 2010 at 10:51 AM, David J. <[hidden email]
> <mailto:[hidden email]>> wrote:
>
>     Matt,
>
>     I am for sure probably wrong, but I think you would need Asterisk or
>     Variant to Determine that it is a Fax Call,
>     I dont think UAC's send T38 information without negotiating with the
>     other side who request that it is capable, then it brings you to
>     Jeff's
>     answer.
>
>     See above.
>
>
>     Matthew S. Crocker wrote:
>     > Can OpenSIPS make routing decisions based on the SDP information
>     in an INVITE?
>     >
>     > Lets say I have the following config
>     >
>     > PSTN -> t.38 Gateway -> OpenSIPS ->  UserAgent
>     >
>     > I have a TN from the PSTN routed to the UserAgent,  I'd like to
>     provide a service so the user can use the TN for both voice & faxing.
>     >
>     > Voice call goes through normally (g.711 g.729 codec)
>     >
>     > Fax call starts off as a normal voice call (INVITE, 180, 183,
>     200).  Once the call is answered the originating end (PSTN) starts
>     sending fax tones. The Gateway hears the fax tones and attempts to
>     RE-INVITE with T.38 in the SDP.  I'd like OpenSIPS to see the T.38
>     capability in the SDP and redirect the call to a fax->e-mail
>     gateway.  So,  the 2nd INVITE comes in, OpenSIPS sends the INVITE
>     to the fax gateway and a BYE to the user.  The fax gateway does a
>     200 and negotiates T.38 with the PSTN gateway.
>     >
>     > I know I can route the call through Asterisk and have it do a
>     quiet answer and listen for the modem sounds.  I'd like to avoid
>     using Asterisk for all RTP traffic and only use it for the fax
>     gateway traffic (i.e. once it has been determined to be a fax
>     Asterisk steps in and handled the T38 -> E-mail)
>     >
>     > -Matt
>     >
>     >
>
>
>     _______________________________________________
>     Users mailing list
>     [hidden email] <mailto:[hidden email]>
>     http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> Users mailing list
> [hidden email]
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>  


--
Bogdan-Andrei Iancu
www.voice-system.ro


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Re: T.38 detection/redirect in OpenSIPS

Brett Nemeroff
Bogdan,
But at this point, you are now playing with a dialg that is already
connected to an endpoint. You'd need to drop the first call to
establish a new call with the reinvite. Right?
-Brett

On Mar 17, 2010, at 11:50 AM, Bogdan-Andrei Iancu <[hidden email]
 > wrote:

> Hi Brett,
>
> Brett Nemeroff wrote:
>> I don't think there is any way to do this without an RTP capable
>> device in the mix.
> you do not need to look into RTP as the FAX is advertised in the
> re-INVITE (in SDP) - so you can detect it from opensips script by
> inspecting the SDP of reINVITES
>>
>> What you may be able to do is have asterisk detect that it's a fax,
>> then reject it if it is.. I don't know if you can do all that without
>> answering the call.
> no, you cannnot, as first the call is established (from sip point of
> view) as a simple audio call and after that re-negotiated (via
> re-INVITE) for FAX
>>
>> Then you can forward it back to the proxy if it is a fax with maybe a
>> prefix.
>>
>> A lot of assumptions in there. Would like to hear if you find
>> something that works. Not sure if you can SIP Spiral yet in asterisk
>> anyway. ;)
> I do not see the need of Asterisk - maybe with some changes, the b2b
> module will be able to handle this - see my prev email.
>
> Regards,
> Bogdan
>
>> -Brett
>>
>>
>> On Wed, Mar 17, 2010 at 10:51 AM, David J. <[hidden email]
>> <mailto:[hidden email]>> wrote:
>>
>>    Matt,
>>
>>    I am for sure probably wrong, but I think you would need
>> Asterisk or
>>    Variant to Determine that it is a Fax Call,
>>    I dont think UAC's send T38 information without negotiating with
>> the
>>    other side who request that it is capable, then it brings you to
>>    Jeff's
>>    answer.
>>
>>    See above.
>>
>>
>>    Matthew S. Crocker wrote:
>>> Can OpenSIPS make routing decisions based on the SDP information
>>    in an INVITE?
>>>
>>> Lets say I have the following config
>>>
>>> PSTN -> t.38 Gateway -> OpenSIPS ->  UserAgent
>>>
>>> I have a TN from the PSTN routed to the UserAgent,  I'd like to
>>    provide a service so the user can use the TN for both voice &
>> faxing.
>>>
>>> Voice call goes through normally (g.711 g.729 codec)
>>>
>>> Fax call starts off as a normal voice call (INVITE, 180, 183,
>>    200).  Once the call is answered the originating end (PSTN) starts
>>    sending fax tones. The Gateway hears the fax tones and attempts to
>>    RE-INVITE with T.38 in the SDP.  I'd like OpenSIPS to see the T.38
>>    capability in the SDP and redirect the call to a fax->e-mail
>>    gateway.  So,  the 2nd INVITE comes in, OpenSIPS sends the INVITE
>>    to the fax gateway and a BYE to the user.  The fax gateway does a
>>    200 and negotiates T.38 with the PSTN gateway.
>>>
>>> I know I can route the call through Asterisk and have it do a
>>    quiet answer and listen for the modem sounds.  I'd like to avoid
>>    using Asterisk for all RTP traffic and only use it for the fax
>>    gateway traffic (i.e. once it has been determined to be a fax
>>    Asterisk steps in and handled the T38 -> E-mail)
>>>
>>> -Matt
>>>
>>>
>>
>>
>>    _______________________________________________
>>    Users mailing list
>>    [hidden email] <mailto:[hidden email]>
>>    http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>> ---
>> ---------------------------------------------------------------------
>>
>> _______________________________________________
>> Users mailing list
>> [hidden email]
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>
>
> --
> Bogdan-Andrei Iancu
> www.voice-system.ro
>
>
> _______________________________________________
> Users mailing list
> [hidden email]
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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Re: T.38 detection/redirect in OpenSIPS

Matthew S. Crocker
In reply to this post by Matthew S. Crocker

What if we don't worry about the 2nd INVITE (switch to T.38) coming from the gateway.  Pass the INVITE on to the UA phone and then catch the (415, 488 ??) error coming back.   We could handle Faxing like we handle Call Forward Busy (486 error handler). Then we could have user_preferences for CFBZ (486 Busy), CFNA (200 timeout), CFOOS (180 timeout), CFT38 (488 Not Acceptable Here) and re-write the URI to go to the Asterisk Fax gateway.


The SIP flow would be something like:

PSTN -> PROXY  INVITE  (SDP/G711)
PROXY -> PSTN  180 Trying
PROXY -> UA INVITE (SDP/G711)
UA -> PROXY 180 Trying
UA -> PROXY 183 Ringing
PROXY -> PSTN  183 Ringing
UA -> PROXY 200 Ok (SDP/G711)
PROXY -> PSTN  200 Ok (SDP/G711)
** RTP Established between UA & PSTN  (mediaproxy/rtpproxy ??) **
** Gateway detects fax tone and attempts to REINVITE to T.38 **
PSTN -> PROXY INVITE (SDP/T38)
PROXY -> PSTN  180 Trying
PROXY -> UA  INVITE (SDP/T38)
UA -> PROXY  488 Not Acceptable Here
** PROXY Error route rewrite URI **
PROXY -> ASTERISK   INVITE (SDP/T38)
ASTERISK -> PROXY   180 Trying
ASTERISK -> PROXY   200 Ok (SDP/T38)
PROXY -> PSTN 200 Ok (SDP/T38)
** RTP established between PSTN & ASTERISK **
PROXY -> UA   BYE/CANCEL

Question:  Can you change RTP source/dest ip/port during a REINVITE?

-Matt

----- Original Message -----

> From: "Bogdan-Andrei Iancu" <[hidden email]>
> To: "OpenSIPS users mailling list" <[hidden email]>
> Sent: Wednesday, March 17, 2010 12:39:56 PM
> Subject: Re: [OpenSIPS-Users] T.38 detection/redirect in OpenSIPS
>
> Hi Matthew,
>
> you do not need to look into the media part, as you can "spot" the FAX
>
> presence via the re-INVITE with T38 codec in SDP (you can detect it
> from
> opensips cfg).
>
> So, maybe using the b2b module for something like:
>     - allow the voice call to be setup via the b2b in a transparent
> way
>     - if the re-INVITE wth T38 is received from GW, b2b will close the
>
> leg to the users UA and create a new leg to something able to handle
> the
> fax - of course, the b2b will bridge the existing leg (towards PSTN)
> and
> the new leg.
>
> Regards,
> Bogdan
>
>
> Matthew S. Crocker wrote:
> > Can OpenSIPS make routing decisions based on the SDP information in
> an INVITE?
> >
> > Lets say I have the following config
> >
> > PSTN -> t.38 Gateway -> OpenSIPS ->  UserAgent
> >
> > I have a TN from the PSTN routed to the UserAgent,  I'd like to
> provide a service so the user can use the TN for both voice & faxing.
> >
> > Voice call goes through normally (g.711 g.729 codec)
> >
> > Fax call starts off as a normal voice call (INVITE, 180, 183, 200).
> Once the call is answered the originating end (PSTN) starts sending
> fax tones. The Gateway hears the fax tones and attempts to RE-INVITE
> with T.38 in the SDP.  I'd like OpenSIPS to see the T.38 capability in
> the SDP and redirect the call to a fax->e-mail gateway.  So,  the 2nd
> INVITE comes in, OpenSIPS sends the INVITE to the fax gateway and a
> BYE to the user.  The fax gateway does a 200 and negotiates T.38 with
> the PSTN gateway.
> >
> > I know I can route the call through Asterisk and have it do a quiet
> answer and listen for the modem sounds.  I'd like to avoid using
> Asterisk for all RTP traffic and only use it for the fax gateway
> traffic (i.e. once it has been determined to be a fax Asterisk steps
> in and handled the T38 -> E-mail)
> >
> > -Matt
> >
> >  
>
>
> --
> Bogdan-Andrei Iancu
> www.voice-system.ro
>
>
> _______________________________________________
> Users mailing list
> [hidden email]
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

--
Matthew S. Crocker
President
Crocker Communications, Inc.
PO BOX 710
Greenfield, MA 01302-0710
http://www.crocker.com
P: 413-746-2760


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Re: T.38 detection/redirect in OpenSIPS

Bogdan-Andrei Iancu
In reply to this post by Brett Nemeroff
right, that is exactly what the b2b is up to do - to be able (at
signalling level) to manipulate the call legs

Regards,
Bogdan

Brett Nemeroff wrote:

> Bogdan,
> But at this point, you are now playing with a dialg that is already
> connected to an endpoint. You'd need to drop the first call to
> establish a new call with the reinvite. Right?
> -Brett
>
> On Mar 17, 2010, at 11:50 AM, Bogdan-Andrei Iancu <[hidden email]
>  > wrote:
>
>  
>> Hi Brett,
>>
>> Brett Nemeroff wrote:
>>    
>>> I don't think there is any way to do this without an RTP capable
>>> device in the mix.
>>>      
>> you do not need to look into RTP as the FAX is advertised in the
>> re-INVITE (in SDP) - so you can detect it from opensips script by
>> inspecting the SDP of reINVITES
>>    
>>> What you may be able to do is have asterisk detect that it's a fax,
>>> then reject it if it is.. I don't know if you can do all that without
>>> answering the call.
>>>      
>> no, you cannnot, as first the call is established (from sip point of
>> view) as a simple audio call and after that re-negotiated (via
>> re-INVITE) for FAX
>>    
>>> Then you can forward it back to the proxy if it is a fax with maybe a
>>> prefix.
>>>
>>> A lot of assumptions in there. Would like to hear if you find
>>> something that works. Not sure if you can SIP Spiral yet in asterisk
>>> anyway. ;)
>>>      
>> I do not see the need of Asterisk - maybe with some changes, the b2b
>> module will be able to handle this - see my prev email.
>>
>> Regards,
>> Bogdan
>>
>>    
>>> -Brett
>>>
>>>
>>> On Wed, Mar 17, 2010 at 10:51 AM, David J. <[hidden email]
>>> <mailto:[hidden email]>> wrote:
>>>
>>>    Matt,
>>>
>>>    I am for sure probably wrong, but I think you would need
>>> Asterisk or
>>>    Variant to Determine that it is a Fax Call,
>>>    I dont think UAC's send T38 information without negotiating with
>>> the
>>>    other side who request that it is capable, then it brings you to
>>>    Jeff's
>>>    answer.
>>>
>>>    See above.
>>>
>>>
>>>    Matthew S. Crocker wrote:
>>>      
>>>> Can OpenSIPS make routing decisions based on the SDP information
>>>>        
>>>    in an INVITE?
>>>      
>>>> Lets say I have the following config
>>>>
>>>> PSTN -> t.38 Gateway -> OpenSIPS ->  UserAgent
>>>>
>>>> I have a TN from the PSTN routed to the UserAgent,  I'd like to
>>>>        
>>>    provide a service so the user can use the TN for both voice &
>>> faxing.
>>>      
>>>> Voice call goes through normally (g.711 g.729 codec)
>>>>
>>>> Fax call starts off as a normal voice call (INVITE, 180, 183,
>>>>        
>>>    200).  Once the call is answered the originating end (PSTN) starts
>>>    sending fax tones. The Gateway hears the fax tones and attempts to
>>>    RE-INVITE with T.38 in the SDP.  I'd like OpenSIPS to see the T.38
>>>    capability in the SDP and redirect the call to a fax->e-mail
>>>    gateway.  So,  the 2nd INVITE comes in, OpenSIPS sends the INVITE
>>>    to the fax gateway and a BYE to the user.  The fax gateway does a
>>>    200 and negotiates T.38 with the PSTN gateway.
>>>      
>>>> I know I can route the call through Asterisk and have it do a
>>>>        
>>>    quiet answer and listen for the modem sounds.  I'd like to avoid
>>>    using Asterisk for all RTP traffic and only use it for the fax
>>>    gateway traffic (i.e. once it has been determined to be a fax
>>>    Asterisk steps in and handled the T38 -> E-mail)
>>>      
>>>> -Matt
>>>>
>>>>
>>>>        
>>>    _______________________________________________
>>>    Users mailing list
>>>    [hidden email] <mailto:[hidden email]>
>>>    http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>> ---
>>> ---------------------------------------------------------------------
>>>
>>> _______________________________________________
>>> Users mailing list
>>> [hidden email]
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>      
>> --
>> Bogdan-Andrei Iancu
>> www.voice-system.ro
>>
>>
>> _______________________________________________
>> Users mailing list
>> [hidden email]
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>    
>
> _______________________________________________
> Users mailing list
> [hidden email]
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>  


--
Bogdan-Andrei Iancu
www.voice-system.ro


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Re: T.38 detection/redirect in OpenSIPS

Jeff Kronlage
I'm confused on this as well - wouldn't you be effectively placing two
calls (one via a non-T38 gateway, one via a T38 gateway) to the same
destination?  Figuring that most T38 is going to terminate to a single
analog device, I would think that were this possible at a SIP level, the
device would already be "busy" before the second call came in as fax
machines don't typically drop the line very rapidly?

Jeff

-----Original Message-----
From: [hidden email]
[mailto:[hidden email]] On Behalf Of Bogdan-Andrei
Iancu
Sent: Wednesday, March 17, 2010 11:23 AM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] T.38 detection/redirect in OpenSIPS

right, that is exactly what the b2b is up to do - to be able (at
signalling level) to manipulate the call legs

Regards,
Bogdan

Brett Nemeroff wrote:
> Bogdan,
> But at this point, you are now playing with a dialg that is already
> connected to an endpoint. You'd need to drop the first call to
> establish a new call with the reinvite. Right?
> -Brett
>
> On Mar 17, 2010, at 11:50 AM, Bogdan-Andrei Iancu
<[hidden email]

>  > wrote:
>
>  
>> Hi Brett,
>>
>> Brett Nemeroff wrote:
>>    
>>> I don't think there is any way to do this without an RTP capable
>>> device in the mix.
>>>      
>> you do not need to look into RTP as the FAX is advertised in the
>> re-INVITE (in SDP) - so you can detect it from opensips script by
>> inspecting the SDP of reINVITES
>>    
>>> What you may be able to do is have asterisk detect that it's a fax,
>>> then reject it if it is.. I don't know if you can do all that
without
>>> answering the call.
>>>      
>> no, you cannnot, as first the call is established (from sip point of
>> view) as a simple audio call and after that re-negotiated (via
>> re-INVITE) for FAX
>>    
>>> Then you can forward it back to the proxy if it is a fax with maybe
a

>>> prefix.
>>>
>>> A lot of assumptions in there. Would like to hear if you find
>>> something that works. Not sure if you can SIP Spiral yet in asterisk
>>> anyway. ;)
>>>      
>> I do not see the need of Asterisk - maybe with some changes, the b2b
>> module will be able to handle this - see my prev email.
>>
>> Regards,
>> Bogdan
>>
>>    
>>> -Brett
>>>
>>>
>>> On Wed, Mar 17, 2010 at 10:51 AM, David J. <[hidden email]
>>> <mailto:[hidden email]>> wrote:
>>>
>>>    Matt,
>>>
>>>    I am for sure probably wrong, but I think you would need
>>> Asterisk or
>>>    Variant to Determine that it is a Fax Call,
>>>    I dont think UAC's send T38 information without negotiating with
>>> the
>>>    other side who request that it is capable, then it brings you to
>>>    Jeff's
>>>    answer.
>>>
>>>    See above.
>>>
>>>
>>>    Matthew S. Crocker wrote:
>>>      
>>>> Can OpenSIPS make routing decisions based on the SDP information
>>>>        
>>>    in an INVITE?
>>>      
>>>> Lets say I have the following config
>>>>
>>>> PSTN -> t.38 Gateway -> OpenSIPS ->  UserAgent
>>>>
>>>> I have a TN from the PSTN routed to the UserAgent,  I'd like to
>>>>        
>>>    provide a service so the user can use the TN for both voice &
>>> faxing.
>>>      
>>>> Voice call goes through normally (g.711 g.729 codec)
>>>>
>>>> Fax call starts off as a normal voice call (INVITE, 180, 183,
>>>>        
>>>    200).  Once the call is answered the originating end (PSTN)
starts
>>>    sending fax tones. The Gateway hears the fax tones and attempts
to
>>>    RE-INVITE with T.38 in the SDP.  I'd like OpenSIPS to see the
T.38

>>>    capability in the SDP and redirect the call to a fax->e-mail
>>>    gateway.  So,  the 2nd INVITE comes in, OpenSIPS sends the INVITE
>>>    to the fax gateway and a BYE to the user.  The fax gateway does a
>>>    200 and negotiates T.38 with the PSTN gateway.
>>>      
>>>> I know I can route the call through Asterisk and have it do a
>>>>        
>>>    quiet answer and listen for the modem sounds.  I'd like to avoid
>>>    using Asterisk for all RTP traffic and only use it for the fax
>>>    gateway traffic (i.e. once it has been determined to be a fax
>>>    Asterisk steps in and handled the T38 -> E-mail)
>>>      
>>>> -Matt
>>>>
>>>>
>>>>        
>>>    _______________________________________________
>>>    Users mailing list
>>>    [hidden email] <mailto:[hidden email]>
>>>    http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>> ---
>>>
---------------------------------------------------------------------

>>>
>>> _______________________________________________
>>> Users mailing list
>>> [hidden email]
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>      
>> --
>> Bogdan-Andrei Iancu
>> www.voice-system.ro
>>
>>
>> _______________________________________________
>> Users mailing list
>> [hidden email]
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>    
>
> _______________________________________________
> Users mailing list
> [hidden email]
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>  


--
Bogdan-Andrei Iancu
www.voice-system.ro


_______________________________________________
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[hidden email]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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Re: T.38 detection/redirect in OpenSIPS

Stanisław Pitucha
In reply to this post by Matthew S. Crocker
On 17/03/10 17:15, Matthew S. Crocker wrote:

> UA -> PROXY 200 Ok (SDP/G711)
> PROXY -> PSTN  200 Ok (SDP/G711)
> ** RTP Established between UA & PSTN  (mediaproxy/rtpproxy ??) **
> ** Gateway detects fax tone and attempts to REINVITE to T.38 **
> PSTN -> PROXY INVITE (SDP/T38)
> PROXY -> PSTN  180 Trying
> PROXY -> UA  INVITE (SDP/T38)
> UA -> PROXY  488 Not Acceptable Here
> ** PROXY Error route rewrite URI **
> PROXY -> ASTERISK   INVITE (SDP/T38)

Your plan will fail around here. The dialog was already established
between the UA and PSTN. The invite which you have not has both "to" and
"from" tags set and anyone seeing it will respond - "I don't know
anything about this dialog, go away". Not much you can do about it
without B2BUA functionality. To solve the problem "properly", you will
need a separate call in this moment.

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Re: T.38 detection/redirect in OpenSIPS

Matthew S. Crocker
In reply to this post by Matthew S. Crocker

One call from a T38 capable gateway.

Voice calls go through like normal calls
T38 calls start off as voice calls and get reinvited by the gateway to switch to t38 (normal 38 behaviour)

Under a normal scenario the reinvite for T38 fails because the UA doesn't support T38 and sends a 488 Not Acceptable here.  The gateway then sends a 3rd invite with G711 for failback.  The call proceeds as g711 and the user on the UA gets blasted with modem squealing.

I'm thinking OpenSIPS could catch the 488 Not Acceptable Here and redirect the call to a T38 capable fax server.  The user would get ringing, answer, hear the initial fax squeal and then the call would disconnect.  The gateways 2nd Invite (REINVITE for T38) would get a 200 Ok with SDP pointing the RTP to the T38 capable fax server (Asterisk/Hylafax).  Hopefully mediaproxy can keep up with the changes in the RTP stream legs so NAT Traversal can be supported.

-Matt



----- Original Message -----

> From: "Jeff Kronlage" <[hidden email]>
> To: "OpenSIPS users mailling list" <[hidden email]>
> Sent: Wednesday, March 17, 2010 1:38:22 PM
> Subject: Re: [OpenSIPS-Users] T.38 detection/redirect in OpenSIPS
>
> I'm confused on this as well - wouldn't you be effectively placing
> two
> calls (one via a non-T38 gateway, one via a T38 gateway) to the same
> destination?  Figuring that most T38 is going to terminate to a
> single
> analog device, I would think that were this possible at a SIP level,
> the
> device would already be "busy" before the second call came in as fax
> machines don't typically drop the line very rapidly?
>
> Jeff
>
> -----Original Message-----
> From: [hidden email]
> [mailto:[hidden email]] On Behalf Of Bogdan-Andrei
> Iancu
> Sent: Wednesday, March 17, 2010 11:23 AM
> To: OpenSIPS users mailling list
> Subject: Re: [OpenSIPS-Users] T.38 detection/redirect in OpenSIPS
>
> right, that is exactly what the b2b is up to do - to be able (at
> signalling level) to manipulate the call legs
>
> Regards,
> Bogdan
>
> Brett Nemeroff wrote:
> > Bogdan,
> > But at this point, you are now playing with a dialg that is already
> > connected to an endpoint. You'd need to drop the first call to
> > establish a new call with the reinvite. Right?
> > -Brett
> >
> > On Mar 17, 2010, at 11:50 AM, Bogdan-Andrei Iancu
> <[hidden email]
> >  > wrote:
> >
> >  
> >> Hi Brett,
> >>
> >> Brett Nemeroff wrote:
> >>    
> >>> I don't think there is any way to do this without an RTP capable
> >>> device in the mix.
> >>>      
> >> you do not need to look into RTP as the FAX is advertised in the
> >> re-INVITE (in SDP) - so you can detect it from opensips script by
> >> inspecting the SDP of reINVITES
> >>    
> >>> What you may be able to do is have asterisk detect that it's a
> fax,
> >>> then reject it if it is.. I don't know if you can do all that
> without
> >>> answering the call.
> >>>      
> >> no, you cannnot, as first the call is established (from sip point
> of
> >> view) as a simple audio call and after that re-negotiated (via
> >> re-INVITE) for FAX
> >>    
> >>> Then you can forward it back to the proxy if it is a fax with
> maybe
> a
> >>> prefix.
> >>>
> >>> A lot of assumptions in there. Would like to hear if you find
> >>> something that works. Not sure if you can SIP Spiral yet in
> asterisk
> >>> anyway. ;)
> >>>      
> >> I do not see the need of Asterisk - maybe with some changes, the
> b2b
> >> module will be able to handle this - see my prev email.
> >>
> >> Regards,
> >> Bogdan
> >>
> >>    
> >>> -Brett
> >>>
> >>>
> >>> On Wed, Mar 17, 2010 at 10:51 AM, David J. <[hidden email]
> >>> <mailto:[hidden email]>> wrote:
> >>>
> >>>    Matt,
> >>>
> >>>    I am for sure probably wrong, but I think you would need
> >>> Asterisk or
> >>>    Variant to Determine that it is a Fax Call,
> >>>    I dont think UAC's send T38 information without negotiating
> with
> >>> the
> >>>    other side who request that it is capable, then it brings you
> to
> >>>    Jeff's
> >>>    answer.
> >>>
> >>>    See above.
> >>>
> >>>
> >>>    Matthew S. Crocker wrote:
> >>>      
> >>>> Can OpenSIPS make routing decisions based on the SDP information
> >>>>        
> >>>    in an INVITE?
> >>>      
> >>>> Lets say I have the following config
> >>>>
> >>>> PSTN -> t.38 Gateway -> OpenSIPS ->  UserAgent
> >>>>
> >>>> I have a TN from the PSTN routed to the UserAgent,  I'd like to
> >>>>        
> >>>    provide a service so the user can use the TN for both voice &
> >>> faxing.
> >>>      
> >>>> Voice call goes through normally (g.711 g.729 codec)
> >>>>
> >>>> Fax call starts off as a normal voice call (INVITE, 180, 183,
> >>>>        
> >>>    200).  Once the call is answered the originating end (PSTN)
> starts
> >>>    sending fax tones. The Gateway hears the fax tones and
> attempts
> to
> >>>    RE-INVITE with T.38 in the SDP.  I'd like OpenSIPS to see the
> T.38
> >>>    capability in the SDP and redirect the call to a fax->e-mail
> >>>    gateway.  So,  the 2nd INVITE comes in, OpenSIPS sends the
> INVITE
> >>>    to the fax gateway and a BYE to the user.  The fax gateway does
> a
> >>>    200 and negotiates T.38 with the PSTN gateway.
> >>>      
> >>>> I know I can route the call through Asterisk and have it do a
> >>>>        
> >>>    quiet answer and listen for the modem sounds.  I'd like to
> avoid
> >>>    using Asterisk for all RTP traffic and only use it for the fax
> >>>    gateway traffic (i.e. once it has been determined to be a fax
> >>>    Asterisk steps in and handled the T38 -> E-mail)
> >>>      
> >>>> -Matt
> >>>>
> >>>>
> >>>>        
> >>>    _______________________________________________
> >>>    Users mailing list
> >>>    [hidden email] <mailto:[hidden email]>
> >>>    http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >>>
> >>>
> >>> ---
> >>>
> ---------------------------------------------------------------------
> >>>
> >>> _______________________________________________
> >>> Users mailing list
> >>> [hidden email]
> >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >>>
> >>>      
> >> --
> >> Bogdan-Andrei Iancu
> >> www.voice-system.ro
> >>
> >>
> >> _______________________________________________
> >> Users mailing list
> >> [hidden email]
> >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >>    
> >
> > _______________________________________________
> > Users mailing list
> > [hidden email]
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
> >  
>
>
> --
> Bogdan-Andrei Iancu
> www.voice-system.ro
>
>
> _______________________________________________
> Users mailing list
> [hidden email]
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
> _______________________________________________
> Users mailing list
> [hidden email]
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

--
Matthew S. Crocker
President
Crocker Communications, Inc.
PO BOX 710
Greenfield, MA 01302-0710
http://www.crocker.com
P: 413-746-2760


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Re: T.38 detection/redirect in OpenSIPS

Bogdan-Andrei Iancu
In reply to this post by Jeff Kronlage
Hi Jeff,

as opensips will act as b2b, your call will be actually split in 2 calls
(from SIP point of view) - a call C1 from GW to opensips and another one
C2 from opensips to UAC. So at re-INVITE time, opensips b2b will hung up
C2 and replace it with a C3 to a new destination, bridging it with C1

Regards,
Bogdan

Jeff Kronlage wrote:

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


--
Bogdan-Andrei Iancu
www.voice-system.ro


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Re: T.38 detection/redirect in OpenSIPS

tcb-2-3
@ All,

This has very exciting possibilities for me. I'm definitely going to look at the
b2b module and test this out. I will share my findings.

regards

On Thu, Mar 18, 2010 at 10:32 AM, Bogdan-Andrei Iancu <[hidden email]> wrote:
Hi Jeff,

as opensips will act as b2b, your call will be actually split in 2 calls
(from SIP point of view) - a call C1 from GW to opensips and another one
C2 from opensips to UAC. So at re-INVITE time, opensips b2b will hung up
C2 and replace it with a C3 to a new destination, bridging it with C1

Regards,
Bogdan

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


--
Bogdan-Andrei Iancu
www.voice-system.ro


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--
TC

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