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[joakimsen at gmail.com]

Is there any trick to getting T.38 fax to work with SIP? I had it
working and one day with no changes *poof* it stopped working and
hasn't worked for months. The only common factor is Asterisk 1.4.x
(always try to use the latest version) and NAT.

I've tried:

-Linksys ATA
-Grandstream ATA
-Audicodes ATA

All do the same thing. Call connects, hear the first 2sec of fax tone
and then just silence, but the call usually stays open.

I've tried two T.38-capable providers.

I've tried two different routers:
-Linksys WRT54GS running DD-WRT (Linux)
-Dell Optiplex 170L running PFSense (BSD)

Different Linux distros on the servers:
-SuSE 64bit
-RHEL 32bit
-SuSE 32bit

Is there any magic to get this to work? As far as I can tell the only
possible config option is "t38pt_udptl = yes" which I have set under
[general] & the peer.

[mkezys at gmail.com]

Hello,

This can help: http://80.86.84.71/kolmiwiki/index.php/Send_Receive_Fax-T38

Regards,
Mindaugas Kezys
http://www.kolmisoft.com
MOR PRO - Advanced Billing for Asterisk PBX
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Andreas van
dem Helge
Sent: Thursday, March 13, 2008 5:16 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] T.38 SIP Issues

Is there any trick to getting T.38 fax to work with SIP? I had it
working and one day with no changes *poof* it stopped working and
hasn't worked for months. The only common factor is Asterisk 1.4.x
(always try to use the latest version) and NAT.

I've tried:

-Linksys ATA
-Grandstream ATA
-Audicodes ATA

All do the same thing. Call connects, hear the first 2sec of fax tone
and then just silence, but the call usually stays open.

I've tried two T.38-capable providers.

I've tried two different routers:
-Linksys WRT54GS running DD-WRT (Linux)
-Dell Optiplex 170L running PFSense (BSD)

Different Linux distros on the servers:
-SuSE 64bit
-RHEL 32bit
-SuSE 32bit

Is there any magic to get this to work? As far as I can tell the only
possible config option is "t38pt_udptl = yes" which I have set under
[general] & the peer.

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[joakimsen at gmail.com]

Has someone submitted a bugreport regarding enabling > 9600kbps fax? I
always wonder why it would never negociate 14400kbps... when it did
work a single page on fine resolution would take 4 minutes.

Thank you very much for that link. I knew there had to be more
possible configurations for T.38. I will give it a try... but I think
I can get away without patching chan_sip.c, no? that just seems to
enable higher bitrates.

And Linksys SPA2102 is one of the exact devices I have in my lab.

On Thu, Mar 13, 2008 at 1:52 PM, Mindaugas Kezys <mkezys at gmail.com> wrote:

[mkezys at gmail.com]

Hello,

Higher speeds then 9600kbps are not permited by patents.

Regards,
Mindaugas Kezys
http://www.kolmisoft.com
MOR PRO - Advanced Billing Solution for Asterisk PBX
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Andreas van
dem Helge
Sent: Friday, March 14, 2008 3:28 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] T.38 SIP Issues

Has someone submitted a bugreport regarding enabling > 9600kbps fax? I
always wonder why it would never negociate 14400kbps... when it did
work a single page on fine resolution would take 4 minutes.

Thank you very much for that link. I knew there had to be more
possible configurations for T.38. I will give it a try... but I think
I can get away without patching chan_sip.c, no? that just seems to
enable higher bitrates.

And Linksys SPA2102 is one of the exact devices I have in my lab.

On Thu, Mar 13, 2008 at 1:52 PM, Mindaugas Kezys <mkezys at gmail.com> wrote:

[steveu at coppice.org]

Mindaugas Kezys wrote:

[rjcarvalho at gmail.com]

I made some tests with FAX in Asterisk 1.4 using T.38 between two ATAs
connected to legacy FAX machines, and realized that only SIP can make
passthrough in the server while RTP go direct between endpoints. Is it
possible for RTP data stream also to make passthrough in Asterisk?

Thanks,
Ricardo Carvalho.


On Fri, Mar 14, 2008 at 1:13 PM, Steve Underwood <steveu at coppice.org> wrote:

[steveu at coppice.org]

Ricardo Carvalho wrote:

[joakimsen at gmail.com]

Asterisk receives T.38 RTP packet from one SIP peer and sends it to
the other SIP peer, it does not process the packets.

By your argument I can't do T.38 @ 1440bps unless the manufactures of
the Ethernet cable, switch, router, keystone jacks, network cards,
CPU, RAM, etc all paid for the royalties for the T.38 patent.

It's like G729 pass-thru.... Just the endpoints need to have the codec.

On Fri, Mar 14, 2008 at 7:58 AM, Mindaugas Kezys <mkezys at gmail.com> wrote: