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 <[EMAIL PROTECTED]> wrote: > 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: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] 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 <[EMAIL PROTECTED]> wrote: > > 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: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] 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. > > > > _______________________________________________ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > _______________________________________________ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
