similar to: does asterisk support FAX t38 protocol?

I have a TDM card in a asterisk machine. I found that once I used it to call out, the call status changed to "connected" even the callee is still ring. How could asterisk distinguish the "ringing" and "connected" in zap channel? thanks.

Need to replace our older soho fax machine with something more current. Would like to run the fax line through *, but haven't been able to make spandsp work correctly with digium TDM04b card. Our fax volume is very low (maybe a few per week), but we have multiple offices in three geographic locations and would like to be able to email the images to the correct location. For planning purposes,

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

Hi, I remember seeing a T38 Gateway application for Asterisk 1.6 floating around, but I can't seem to find it again. Does anyone have any pointers to it? I really want to be able to send an incoming T38 connection directly to the PSTN. Thanks. -- James

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

Hi, I am looking for an Mail-2-Fax and in a second step Fax-2-Mail-solution that works via T38 with Asterisk, currently still version 1.4 but it also should work with 1.6. For Mail-2-Fax I am thinking that you either have to install a special printer-driver on your Windows-PC (Mac and Linux would be good too), where you can print your fax too and where you have to enter the destination number.

Gentlemen, I have 3 faxes attached to an Asterisk. Fax - SPA2102 - Asterisk. 0851711201 and 0851711290 is on our WAN, no NAT. 0197673581 is outside our WAN and needs to be NAT'ed. Sending a fax from 0851711201 to 0851711290, no problem, switches to T38 and fax goes through. Sending a from 0197673581 to 0851711201, no problem as long as i dont enable T38 on 0197673581. But, if i enable T38

Is it possible to force T38 for all invocations ReceiveFAX() ? Receiving fax always worked OK on Callweaver though I could put SipT38Switchover() into the dial plan. I can't with Digium fax, and it always fails at the point it decides to switch to T38.

How can I get asterisk to work with faxes in my configuration? I have a WAN with Asterisk at the centre and Mediatrix 1104 gateways at the end nodes providing tone to legacy PBX's and fax machines. The Asterisk is connected to the PSTN via a Digium single port t1. The end nodes are connected via frame-relay 128kbps links. I want to use g.729 between the end nodes and the Asterisk box at

Hi everyone, I have been trying to get T.38 Faxing to work with Iristel sip trunks for last few days but havn't been sccussful. I am using Asterisk 1.6.2.8 and SpanDSP 0.6. Here is what I see in the tcpdump capture: 1. Call come in from the trunk as regular voice call with g.711 codec 2. Asterisk answers the call and recognizes the CNG and sends the call to fax extension 3. Eventually

Hello to all, Is there an "how-to" for asterisk and setting up a t38 fax gateway (SIP) ? I look at http://bugs.digium.com/view.php?id=5090 to patch asterisk chan_sip.c file. What are the next steps to get a t38 fax gateway with asterisk ? Regards Harry PS: I use hylafax server. ___________________________________________________________________________ Faites de

Hi all.. Im using ABE C3.2.1 version and here Im having the following issue with respect to T38 fax calls. Somehow asterisk when it receives Session modification request with the T38 Codec, is not forwarding the request to the other end. Also asterisk is responding with 200 Ok to this RE INVITE with an audio line rather an media line with image. The session id here strangely matches to that of

Using spandsp-0.0.6-pre17, SendFax on 1.6.1.18 and ReceiveFax on 1.6.2.8. Sip.conf on both sides has t38pt_udptl = yes. -- Executing [s at fax-tx-test:3] SendFAX("SIP/side-sip-00000009", "/var/spool/asterisk/fax/20091113_1455.tif") in new stack [Mar 20 17:05:34] WARNING[6433]: app_fax.c:178 phase_e_handler: Error transmitting fax. result=49: The call dropped

Hi, I'm trying to send faxes using Asterisk 1.4 and T38 with sip but Asterisk rejects the t38. Anybody know if is possible to transmit t38 fax with Asterisk 1.4? following settings: --- sip.conf --- [general] bindport=5060 bindaddr=0.0.0.0 disallow=all allow=ulaw allow=alaw context=from-outside t38pt_udptl=yes [operator] qualify=no nat=yes host=189.160.126.201 dtmfmode=rfc2833

Anyone have any testing experience with T38 and HT-502 Grandstream? I just want to confirm that t.38 is working on this device. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110801/fea18d8c/attachment.htm>

Hi, I am looking for a very low cost way of receiving and sending T38 fax reliably. Is there any possible solution using Asterisk as the PSTN SIP gateay and Digium E1/T1 card? Is there other open source package that can help to accomplish this purpose? Regards, Mark -------------- next part -------------- An HTML attachment was scrubbed... URL:

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 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

Fax over T38 is failing, on the same system it worked with Callweaver. What do I need to post to be get further assistance please?

Hi, I continue the discussion from https://issues.asterisk.org/view.php?id=19103 If T.38 reinvite detection should still work, why it doesn't? If I use faxdetect = t38 it does never detect the fax, even using alaw. Cheers, Darkbasic