similar to: ReceiveFAX G.711 + Realtime

Hello, We are trying to send faxes by T.38 protocol to a remote SIP proxy from a local extension. The local extension sends the INVITE, Asterisk sends the call to the Proxy the call is connected with a regular audio codec. After a few seconds the remote proxy sends an INVITE with UDPTL and the Asterisk sends it to the local extension and it's accepted, but (here the problem starts) just

Hello users, Recently i have installed the free version of FaxForAsterisk and trying to work with it by sending a fax on T38. My version information is as follows i)Asterisk 1.6.0.20 ii)res_fax-1.6.0.14_1.1.6-x86_32 iii)res_fax_digium-1.6.0.14_1.1.6-i686_32 sip.conf [general] t38pt_udptl=yes extensions.conf [default] exten => _XXXXXXXXXX,1,NoOp(Fax Incoming Call) exten =>

I am testing out inbound faxing using res_fax and res_fax_spandsp.so My system answers the call but then sets there on the ReseiveFax line then comes back with an error that it exceeded the maximum retries. How would I go about debugging this? Below is my very simple dialplan code I am using, and the fax show version gives the following as well. FAX For Asterisk Components:

Hi folks, I am having a small problem with asterisk-1.6.2.7 + app_fax on OpenBSD 4.7 -release. Everything seems to work fine. I have a macro which answers, receives the fax to a tiff, and then runs a script (mailfax) to convert that to pdf and email it. It all works perfectly except for some errors I am seeing in the console. After it hangs up I get a dozen or so messages in the cli

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

I am testing Fax for Asterisk. But, I meet a problem. I try to Send a Fax (.tiff) from the first asterisk (Asterisk1) to the second asterisk (Asterisk2). Asterisk1 initiates an INVITE with audio G.711. Asterisk2 accepts this INVITE. Immediately, Asterisk2 sends an re-INVITE with T.38 to Asterisk1. But, Asterisk1 responds with "488 not acceptable here". I double check t38pt_udptl = yes in

Hi all, I've got a perl agi script that exec()'s the FFA version of receivefax to... receive a fax. However, after the fax is received, the script seems to die. This is what I have: $main::agi->exec("receivefax","/tmp/${$}.tiff|fs"); $main::agi->verbose("FAX COMPLETE",1); I never see the "FAX COMPLETE" message on the console, I've set

What I have is: * Asterisk 1.8.10.1~dfsg-1ubuntu1, * SPA112 ATA with analog fax in 1-st FXS port connected, * SIP trunk with provider supporting T.38. My network looks like this: * spa112 (192.168.33.200/24) and Asterisk (192.168.5.253/24) in neighbouring LANs, * Asterisk connects to the provider (80.75.130.136) via router (82.200.7.184). Router has full DNAT to Asterisk server. What happens?

Hello! I?m trying to add fax functionality to my asterisk installation. Right now I?m focusing on receiving faxes. This is not explained in a book, but I assume that I can use same context, add ?fax? extension and if someone calls to send fax - it will autodetect. Right? Per book, I made following setup additions: 1. In sip.conf [general] I added: ;FAX stuff faxdetect=yes t38pt_udptl=yes 2.

Hello, We have been working with the ReceiveFax application for some weeks now in order to receive faxes in T.38 and it works fairly well, but there are some faxes that for some reason we are not able to receive correctly. The asterisk version we are using is 1.6.0.6 with spandsp-0.0.5pre4 and the asterisk machine is behind a CISCO mediaGW to be able to communicate with the PSTN. The SIP call

Dear All, I have created an inbound context in sip.conf that forward incoming call to opensips server...The problem appears as soon as I enable t38pt_udptl = yes under General context...The Asterisk negotiate the SIP session with OpenSIPS without adding voice codec to INVITE packet...It just contains T.38 protocol...When t38pt_udptl is disabled everything looks OK and Ulaw is negotiated with

FYI, asterisk DOES now support g723, but you have to pay for it: http://store.yahoo.com/asteriskpbx/asteriskg729.html -----Original Message----- From: Dan Fernandez <danfernandez00@hotmail.com> Date: Mon, 5 May 2003 17:33:05 -0300 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Has anyone gotten SIP_CODEC= to work? Basically, since I?d like to use g723 for sip

Hi all, I realize that the application Receivefax can't handle with more than one fax at the same time. In a environment with a lot of fax, some caller get the signal but the operation can't be completed. Is there a way to send busy tone to the second caller? Att, Flavio Roberto Miranda MSN:flaviormiranda at hotmail.com Skype: flaviormiranda -------------- next part

Hi all, I am running to the following problem, when using the below dialplan to receive fax, everything works perfect till this line exten => receive,n,ReceiveFAX(${FAXFILE}): and then the following line cannot be executed, it's like asterisk can't go back to dialplan and continue, the good news is when i check what is received in my fax folder i find that the file is a valid one (not

For our internal fax machines, I'm checking if the faxes are going to branch offices. If they are, I want to capture and email them to the branches. I've set up extension 8447 to test this. A fax machines is connected via an SPA 2102 on 173. Any calls from 173 are sent to: [outbound-fax] exten => 8447,1,Answer() exten => 8447,n,GoSub(Capture-Fax,s,1) exten

Hi, On a new Raspberry2 with Asterisk 11.13.1 binary-installed, ReceiveFAX exits with FAILED status. Each attempt is made with something like: channel originate Local/12345 at faxgateway application SendFAX myfile.tif I'm not sure if failure "mostly" comes from SendFAX or from ReceiveFAX but it fails with 100% rate. uname -a output is: Linux myhost 3.18.7-v7+ #755 SMP PREEMPT

dialplan [FaxIncoming] exten=s,1,NoOp(Incoming fax on 46-va) same=n,Set(FAXFILE=/var/spool/asterisk/fax/${STRFTIME(${EPOCH},,%Y%m%d)}_${STRFTIME(${EPOCH},,%H%M)}) same=n,Answer() same=n,ReceiveFAX(${FAXFILE}.tif,df) same=n,Hangup() exten=>h,1,NoOp(FAXSTATUS: ${FAXSTATUS} FAXERROR: ${FAXERROR} FAXPAGES: ${FAXPAGES} @ bitrate ${FAXBITRATE}) same=n,System(scp ${FAXFILE}.tif

Hi, I am using Aserisk 1.4.23.1 and trying to use SIP_CODEC to define the codec being used. I have exclusively Polycom phones for this test, and basically I want all communications to use g729 (preferred codec), except for pagine 20 phones (which busts my g729 license count). In that case I want to use gsm. I have therefore specified Set(SIP_CODEC=gsm) I my dialplan before the appropriate

We want some of our users to use G729, and some others to use ULAW. Our PSTN gateway provider supports both, so that's not a problem, and if I force him (the PSTN gateway) to allow G729 only, the outgoing call will take place with G729. The problem is that I want to have my PSTN provider configured to allow ULAW as a first priority, then G729. I did it like that: [mypstngate] type=friend

Hi, I'm sending twice the same file using Hylafax's sendfax app. The first time I'm "dialing" a DID attached to the ReceiveFAX application. The second time I'm dialing an internal extension attached to the same ReceiveFAX application : 1. sendfax/hylafax/iaxmodem ----> asterisk ----> dadhi ----> dahdi ----> asterisk ----> spandsp 2.