similar to: Asterisk 11.5.0 T38 Faxing

Hello List, I have been trying to setup T38 gatewaying with the following setup SIP ->Asterisk -> DAHDI TE410P with Libss7 -> TELCO I'm using asterisk Asterisk 1.8.3.2 and DAHDI Version: SVN-trunk-r9697M Echo Canceller: HWEC I'm aware there's no support for T38 gateway but I have been trying to get the patches https://issues.asterisk.org/view.php?id=13405 to work. It seems

Hi all, I have a question regarding using T38 for fax sending and here is my scenario: fax -> SIP ATA (T38 enabled) -> Asterisk #1 -> IAX TRUNK -> Asterisk #2 -> SIP ATA (T38 enabled) -> fax My question is, how can I know if I'm really using T38? is T38 information coming to the other side (because of SIP to IAX conversion) or just plain g711a data? I'm using Linksys

Hi all I installed DAHDI Version - 2.6.1 DAHDI Tools Version - 2.6.1 libss7-trunk Asterisk 11.0.1 from source on Fedora 12 x86_64. Now i`m unable to load chan_dahdi and libss7: myserver*CLI> module load chan_dahdi.so ?ERROR[10124]: chan_dahdi.c:17842 process_dahdi: Unknown signalling method 'ss7' at line 37. myserver*CLI> module load libss7.so Unable to load module libss7.so

The Asterisk Development Team has announced the release of libss7 2.0.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/libss7 The release of libss7 2.0.0 resolves several issues reported by the community and would not have been possible without your participation. Please note that this version of libss7 has been released in anticipation of what

The Asterisk Development Team has announced the release of libss7 2.0.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/libss7 The release of libss7 2.0.0 resolves several issues reported by the community and would not have been possible without your participation. Please note that this version of libss7 has been released in anticipation of what

Hi I'm trying to get the spa 8000 used with a fax machine using t38 passthru i have tried with 1.6.2 and 1.8.3 and is still a no go the provider i use is 012 in israel wich supports t38 (i use it with ffa) could anybody give me a clue how to get this working if it should t38pt is set to yes in sip.conf Thanks, Israel -------------- next part -------------- An HTML attachment was

The Asterisk development team has released new versions of three libraries used with Asterisk. They are: libpri-1.2.8: This release contains a number of bugfixes that had been unreleased for months, along with clarification of the licensing of the source code. The change log is here: http://downloads.digium.com/pub/telephony/libpri/ChangeLog-1.2.8 libpri-1.4.7: This release contains primarily

Hei! I'm trying to send special characters out to ss7 link, but libss7 seems to convert them to zeroes. The challenge is that our service provider demands some of the regional numbers to be sent in format D0+number. When I use D in front of the number in dialplan, libss7 replaces it with 00, So I have a dial string: exten => _[A-Z].,1,Dial(DAHDI/g1/DD0501,,g) But in SS7 trace I

Hi. Dont know if this is an Asterisk or Dahdi or LibSS7 Error. So Im writing to Asterisk List. If somebody knows where to search (dahdi lists or libSS7 lists) will be appreciated. Im getting this error after a certain time, My config is: Hardware: 3 Digium Quad E1 TE4XXP libss7 version: SVN-branch-1.0-r286 DAHDI Version: 2.4.0 Echo Canceller: Asterisk 1.6.2.14 CentOS release 5.5 (Final) Kernel

Hi all, I've an asterisk 1.2.5 running very well for about a 9 months, and suddenly i cannot dial extensions 4XXX from SIP Phones. Now comes the wired stuff... I can dial this extensions from IAX phones as well as from Analogue extensions connected to our legacy pbx, that is installed on front of asterisk. So : Zapata Calls to SIP extensions 4XXX - OK IAX to SIP 4XXX-OK SIP to SIP 4XXX -

Hi All, I've been knee deep in T38 faxing for a couple of weeks now, trying to find a version of Asterisk that would pass through T38 with an Audiocodes Mediant 1000 and MP203 ATA. I had problems with 1.6.0.x through 1.6.1.10. Tested 6 different versions. Either it just would not work or fail back to G.711, or re-invite with wrong T38FaxMaxDatagram sizes, faxes would work one-way and not

Hello, I need help to solve a problem that I am having using Asterisk 13, PJSIP and T38. My setup is as follows: SIP Provider --> Asterisk 13 --> Patton --> Physical Fax I need to get the fax directly in T38 to Patton. The provider sends me the fax in T38. If I receive the T38 fax on Asterisk (using an hylafax device), I can properly receive the fax. If I send a T38 fax with Asterisk

Hi, I'm working with asterisk 1.4.35 and found an issue regarding codecs negotiation when T38 is enabled (t38pt_udptl=yes). In particular if the INVITE sdp contains no allowed codec the call is not rejected with "488 - Not acceptable here" but it goes through and the 200 OK SDP is as follows: v=0 o=root 27285 27285 IN IP4 xxx.xxx.xxx.xxx s=session c=IN IP4 xxx.xxx.xxx.xxx t=0 0

Hi My call flow is T38 static IP gateway --> Asterisk --> Sip Provider--> PSTN Call is placed in reverse direction - from PSTN to T38 Gateway. T38 GW reinvites with T38, and asterisk passes it along to the SIP provider. The SIP provider challenges it and asterisk reponds to the Challenge with INVITE with Auth credentials...however, the Asterisk changes the SDP and replaces the T38

I have problem with fax machine Panasonic DX600. It's connected to Grandstream Handy Tone 386 which is connected to Asterisk. Asterisk is connected to my SIP provider. To some numbers I can't send FAX, and I get following error on CLI. WARNING[2237] chan_sip.c: Unknown SDP media type in offer: image 31358 udptl t38 I believe that Panasonic DX600 machine supports T38. And when I have

Hi, I?m trying to use SS/ in Asterisk. I'm thinking in chan_ss7 and libss7, and I want to know some other experience with this. Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100121/f8c4937e/attachment.htm

Hi I have finished the installation of my VoIP basic configuration ... Actually: - All calls from my E1 are received by a Cisco AS5300 and sent to my Asterisk (in G711 by SIP). - All user are connected by SIP to the Asterisk - All calls from User are sent by asterisk to the Cisco AS5300 Now, i want see if i can supply T38 Fax Gateway .... I am search to: - Cisco Receive all

Dear all, I'm trying to enable Asterisk to work with FAX using T38. I've tried Asterisk 1.2.4 with the available patch found at URL http://bugs.digium.com/view.php?id=5090 and also with the new 1.4 Beta3 that is announced to support it too. With both Asterisk versions, I've sent with success FAXes between two FAX machines each one attached to an ATA interface, both registered in

Hello List, To whom it might concern: I have been working in some SlackBuilds (script for making Slackware Packages) for my personal use, but thought they might be useful for someone else here. Beside of the exceptional distributions used so far (CentOS, Debian, Ubuntu, etc.), you might want to test Asterisk on a Slackware Linux box, as it offers outstanding stability and flexibility as

Hello List, I think I've discovered a little bug in t.38 bug in 1.6.0.22 regarding the speed (T38MaxBitRate) used to send the faxes. Asterisk always responds with a=T38MaxBitRate:2400. I've tried with Patton and Grandstream devices and the result is always the same. Patton ignores the parameter and sends the fax at 9600.