similar to: T38 and codecs negotiation

Hi, I'm tryng to place a call through Asterisk Manager Originate Action. Since I want separate CDR for each of the two legs of the call, I'm forking CDR with ForkCDR as the first Channel has picked up. The problem is that, while the first CDR is fine, in the second one the "answer" field is always empty, "billsec" field is 0 and "disposition" field is

Asterisk 1.4.32 (Also 1.4.26, 1.4.33) Broadvox ITSP (xxx.xxx.xxx.xxx) Linksys 2102 (yyy.yyy.yyy.yyy) Both peers : canreinvite=yes t38pt_udptl = yes I'm having some trouble getting a T.38 fax call established with Broadvox. During negotiation, Asterisk sends a SIP re-invite (T38 switchover) to Broadvox with the Asterisk server's IP address in the Connection Information (c) instead of

Hello! unchanged asterisk crashes during udptl / t.38 negotiation with telekom - they do not support t.38 / udptl. In detail: fax client -> asterisk -> telekom -> easybell -> asterisk -> fax server Fax server sends t.38 reinvite via asterisk to easybell. Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): - 2447581897 4 IN IP4 46.17.15.23

Here is the diff from chan_sip.c 15 days ago and 16 days ago. 15 days ago is the point outgoing calls made via grandstream budgetone stopped working. Any help on why it breaks? Any possible fix? /tmp# diff asterisk/channels/chan_sip.c asterisk.works/channels/chan_sip.c 289d288 < int capability; 3921,3922d3919 < p->capability = user->capability;

Hi, I've managed to get HYLAFAX---->T38MODEM----->ASTERISK---->CISCOAS5400 working, but when they are negotiating asterisk drops a message telling "Unknown RTP codec 96 received from gateway" Do somebody know how to fix it ? Thank you ! << [ TYPE: Control (4) SUBCLASS: Ringing (3) ] [SIP/GWCISCO5400O-600bfcc8] << [ TYPE: Control (4) SUBCLASS: Answer (4) ]

non-video mode, it is never able to add video to the channel? Is this correct, or am I missing something? It looks as if the codec 'jointcapability' is calculated at the start of the call, and can never be added to (with exceptions for T.38 fax) as any SDP update is masked using the existing 'jointcapability' and knocks out the newly requested codec. Is that right? Thanks, Steve

The latest chan_sip.c works for my budgetones with the following lines removed. YMMV. I haven't bothered to dig in and see what those lines actually do. Did soneone just get wacky with cut and paste from the peer while loop? Or am I breaking something else. Jon --- chan_sip.c.broken Fri Nov 7 02:17:47 2003 +++ chan_sip.c Fri Nov 7 02:16:23 2003 @@ -3928,8 +3928,8 @@ static int

Hello *, is the rtpip patch still valid for asterisk 1.6 (with some code changes, obviously)? https://issues.asterisk.org/view.php?id=8161 Or, in asterisk 1.6 there is an alternative to using it? This is the difffile I produced for chan_sip.c in asterisk 1.6.2.11 --- chan_sip.c 2010-10-12 13:47:49.000000000 +0200 +++ chan_sip.c.orig 2010-10-12 13:47:27.000000000 +0200 @@ -987,9 +987,6 @@

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 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

Hello We are experiencing some difficulties with T38 faxing. I have a Asterisk 11.5.0 with libss7 and Sangoma A104DE digital interface card . The operating system is Centos 6 We are using this server to terminate calls to Telco. So calls are coming to asterisk from sip and we are sending calls to Telco with Dahdi. (It is a one way interconnection only from asterisk to telco ,not from telco to

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 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

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

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

Disclaimer: Yes, I know faxing over G711 is unreliable. :-) We're running Asterisk 1.0.9 which talks to a Audiocodes SIP Gateway. We're running Sipura SPA-2002's as ATA's and faxing within our own voice network is working. If we try and fax out to the world however, we're running into a problem. When the call connects and the modem tones begin to negotiate, our SIP/PSTN

Dear all, Did anyone successfully test T38 fax pass thru to Cisco as53xx? We've tried 1.2.4 with latest patch and latest svn trunk and T38 patch but still not work. Reinvites from Cisco are correctly passed back to the originating gateway, but fax never able to connect. Cisco IOS 12.3.x configuration voice service voip fax protocol t38 ls-redundancy 5 hs-redundancy 2 fallback

I found a bug in the latest T38 passthrough patches, the effect is that a non-SIP call after being put on hold is then lost, no resume is possible. The fix is to be applied in the chan_sip.c file: } else { /* No bridged peer with T38 enabled*/ transmit_response_with_sdp(p, "200 OK", req, 1); } - } + } else transmit_response_with_sdp(p, "200