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tnelson at rockbochs.com
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 Posted: Tue Oct 21, 2014 10:24 pm Post subject: [asterisk-users] res_fax T.38 Gateway with SpanDSP - Force R |
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Greetings-
Working with the T.38 gateway functionality that is sparsely documented [1], I'm attempting to get the following functional:
Asterisk calling system -> Asterisk system in T.38 Gateway Mode (box in question) -> SIP Provider
The problem is:
-The provider is not initiating a reinvite to T.38, even though it is 100% supported
-Asterisk is not detecting the CNG tones from the far side of the call and initiating a T.38 session on that call leg (with the SIP provider), but *DOES* attempt to initiate a T.38 session with the calling Asterisk system (which rejects with SIP/488 as expected)
So, how does one force a reinvite to T.38 on the outbound call leg in this scenario? I did find the same problem from another user on the list in the archives, but didn't find a solution contained within the responses [2].
Thank you,
--Tim
[1] https://wiki.asterisk.org/wiki/display/AST/T.38+Fax+Gateway
[2] http://lists.digium.com/pipermail/asterisk-users/2012-July/273535.html |
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tnelson at rockbochs.com
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| Posted: Wed Oct 22, 2014 2:55 pm Post subject: [asterisk-users] res_fax T.38 Gateway with SpanDSP - Force R |
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----- Original Message -----
| Quote: | Working with the T.38 gateway functionality that is sparsely
documented [1], I'm attempting to get the following functional:
|
| Quote: | Asterisk calling system -> Asterisk system in T.38 Gateway Mode (box
in question) -> SIP Provider
|
| Quote: | -The provider is not initiating a reinvite to T.38, even though it is
100% supported
-Asterisk is not detecting the CNG tones from the far side of the
call and initiating a T.38 session on that call leg (with the SIP
provider), but *DOES* attempt to initiate a T.38 session with the
calling Asterisk system (which rejects with SIP/488 as expected)
|
| Quote: | So, how does one force a reinvite to T.38 on the outbound call leg in
this scenario? I did find the same problem from another user on the
list in the archives, but didn't find a solution contained within
the responses [2].
|
*bump*
Any thoughts? I'm quite familiar with the T.38 functionality within Callweaver, and a function is provided there to do exactly what I need ( SipT38SwitchOver() ). However, given Callweaver is ancient at this point, and better T.38 features such as "gateway" do not function, I am pressed to use Asterisk (11.13.1) with SpanDSP (0.0.5, latest from Github since spandsp.org is down) for this job. 
Thanks!
--Tim
--
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dfullertasterisk at sh...
Guest
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| Posted: Thu Oct 23, 2014 8:35 am Post subject: [asterisk-users] res_fax T.38 Gateway with SpanDSP - Force R |
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On 10/22/2014 03:55 PM, Tim Nelson wrote:
| Quote: | ----- Original Message -----
| Quote: | Working with the T.38 gateway functionality that is sparsely
documented [1], I'm attempting to get the following functional:
|
| Quote: | Asterisk calling system -> Asterisk system in T.38 Gateway Mode (box
in question) -> SIP Provider
|
| Quote: | -The provider is not initiating a reinvite to T.38, even though it is
100% supported
-Asterisk is not detecting the CNG tones from the far side of the
call and initiating a T.38 session on that call leg (with the SIP
provider), but *DOES* attempt to initiate a T.38 session with the
calling Asterisk system (which rejects with SIP/488 as expected)
|
| Quote: | So, how does one force a reinvite to T.38 on the outbound call leg in
this scenario? I did find the same problem from another user on the
list in the archives, but didn't find a solution contained within
the responses [2].
|
*bump*
Any thoughts? I'm quite familiar with the T.38 functionality within Callweaver, and a function is provided there to do exactly what I need ( SipT38SwitchOver() ). However, given Callweaver is ancient at this point, and better T.38 features such as "gateway" do not function, I am pressed to use Asterisk (11.13.1) with SpanDSP (0.0.5, latest from Github since spandsp.org is down) for this job.
Thanks!
--Tim
|
I can't help with your root problem (maybe check "core show function
FAXOPT"?), but the spandsp site is up. Try using www.spandsp.org.
Downloads are available here: http://www.spandsp.org/downloads/spandsp/
-Dave
--
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New to Asterisk? Join us for a live introductory webinar every Thurs:
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lmoore at omninet.net.au
Guest
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| Posted: Thu Oct 23, 2014 8:49 am Post subject: [asterisk-users] res_fax T.38 Gateway with SpanDSP - Force R |
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On 23/10/2014 3:55 AM, Tim Nelson wrote:
| Quote: | ----- Original Message -----
| Quote: | Working with the T.38 gateway functionality that is sparsely
documented [1], I'm attempting to get the following functional:
|
| Quote: | Asterisk calling system -> Asterisk system in T.38 Gateway Mode (box
in question) -> SIP Provider
|
| Quote: | -The provider is not initiating a reinvite to T.38, even though it is
100% supported
-Asterisk is not detecting the CNG tones from the far side of the
call and initiating a T.38 session on that call leg (with the SIP
provider), but *DOES* attempt to initiate a T.38 session with the
calling Asterisk system (which rejects with SIP/488 as expected)
|
| Quote: | So, how does one force a reinvite to T.38 on the outbound call leg in
this scenario? I did find the same problem from another user on the
list in the archives, but didn't find a solution contained within
the responses [2].
|
*bump*
Any thoughts? I'm quite familiar with the T.38 functionality within Callweaver, and a function is provided there to do exactly what I need ( SipT38SwitchOver() ). However, given Callweaver is ancient at this point, and better T.38 features such as "gateway" do not function, I am pressed to use Asterisk (11.13.1) with SpanDSP (0.0.5, latest from Github since spandsp.org is down) for this job.
|
No thoughts on your problem, I do think you will need a newer version of
spandsp through - the site seems to be up now.
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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lmoore at omninet.net.au
Guest
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| Posted: Thu Oct 23, 2014 9:08 am Post subject: [asterisk-users] res_fax T.38 Gateway with SpanDSP - Force R |
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On 22/10/2014 11:23 AM, Tim Nelson wrote:
| Quote: | Greetings-
Working with the T.38 gateway functionality that is sparsely documented
[1], I'm attempting to get the following functional:
|
What type of endpoint are you using which is originating the call and is
it T.38 capable?
Larry.
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
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lmoore at omninet.net.au
Guest
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| Posted: Thu Oct 23, 2014 11:08 am Post subject: [asterisk-users] res_fax T.38 Gateway with SpanDSP - Force R |
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On 23/10/2014 10:07 PM, Larry Moore wrote:
| Quote: |
On 22/10/2014 11:23 AM, Tim Nelson wrote:
| Quote: | Greetings-
Working with the T.38 gateway functionality that is sparsely documented
[1], I'm attempting to get the following functional:
|
What type of endpoint are you using which is originating the call and is
it T.38 capable?
Larry.
|
Have you had a look at
https://wiki.asterisk.org/wiki/display/AST/SIP+Direct+Media+Reinvite+Glare+Avoidance
As an exercise you could disable T.38 on 'Asterisk calling system', if
you have an ATA which is originating the call to 'Asterisk calling
system' disable T.38 on that device too and disable in your sip.conf
using t38pt_udptl=no.
If you are using SendFax() on 'Asterisk calling system' ensure T.38 is
not able to be used.
If using an ATA connecting to 'Asterisk calling system' ensure you have
set in your peer's configuration canreinvite=no or directmedia=no,
depending on the version of Asterisk you are running on this system.
On Asterisk system in '(box in question)' set directmedia=no for the
peer which is connecting to 'SIP Provider' and also to 'Asterisk calling
system', you may want to set setvar=FAXOPT(gateway)=yes in your peer
config to 'SIP Provider' otherwise it will need to be set in your dialplan.
Set your verbose & debug to at least 3 on '(box in question)', possibly
a little higher and send a fax - you may now see the Fax Gateway detect
CED. Not sure if this is suppressed in
You may want enable udptl debugging on '(box in question)'.
Larry.
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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