similar to: Free Fax for Asterisk

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

Hello, For those customers with only analog lines, who ask for fax2email and email2fax, whats the most reliable solution available and tested with Asterisk? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120529/3e28b56e/attachment.htm>

Hi All; Again, the Cisco IP Phones 7942G and using Skinny: I upgraded the firmware to version 8.5 (skinny) and I am using skinny channel (chan_skinny) and the skinny.conf file. The phones are registering, but when we use them to place a call, we only hear tooooooooooo in the handset and we do not hear voice (even when we dial the digits, we only hear toooooooo .. but it dials and destination

Is it possible to do SIP<->Asterisk<->TDM in a single step with FFA? Or does FFA always use TIFF files? I'm using Free FFA on 1.6.2.15 and I want to be able to use SPA 2102 ATA's at the fax machines and send faxes directly over a PRI. Mark -- Mark Willis Star One Telecom Office: 1-800-889-7001 Cell: 210 880 5097 http://staronetel.com

[0004f2xxxxxx](poly650) defaultuser=0004f2xxxxxx callerid="Front Desk" <1600> mailbox=1600 *setvar=callidnum=1234561600* and from extensions.conf: [outgoing] ; Outbound unrestricted domestic calls exten => _1NXXXXXXXXX,1,Verbose(Outbound call from ${callidnum} to ${EXTEN} on ${STRFTIME(${EPOCH},,%D)} at ${STRFTIME(${EPOCH},,%T)}.) *exten =>

In what appears to be the most current documentation of FFA I'm directed to http://www.digium.com/en/docs/FAX/faa-download.php However the download selector utility found there "ain't working". Specifically, none of the drop lists appear to be populated (in any browser, on any platform), thereby preventing any actual 'selection'. Has anyone else noticed this? Is there

I've been having problems segfaulting when receiving faxes. So I tried FFA. I used the register utility and my license is in /var/lib/asterisk : ls /var/lib/asterisk/licenses/ -l total 4 -rw-r--r--. 1 root root 325 Jun 5 22:41 FFA-DNCXXXXXXX.lic but: Executing [17772822954 at FaxIncoming:4] ReceiveFAX("SIP/ec2faxcall18-00000002",

Hello, We succeed to send faxes using FFA, when the files are converted to tif from PDF using gs, but it doesn't work with tif files we copy/upload directly from our PCs. We saw in the manual that the size is important, since we got the error "FAX handle 0: failed to queue document 'filename.tif'", so we set it to 1680x2285, but it's still rejected. Is there a way

Hi, I have a 1.6.2.6 fax installation with a FFA license which seems to be installed correctly (in fax show stats, I see that I have 1 Digium G.711 licensed channel, and 1 Digium T.38 licensed channel). When trying to call my business line with a fax machine, it looks like it's ringing to my asterisk box, then transfer the call to my extension. In the logs, I see (after the line where

Just finished with the instructions from digium website/ net on how to compile FFA: After restart, modules did not get loaded so tried to load manually: [Dec 18 14:31:26] WARNING[11002]: loader.c:359 load_dynamic_module: Error loadin ile: No such file or directory [Dec 18 14:31:26] WARNING[11002]: loader.c:653 load_resource: Module 'res_fax.so Verified the files exist: astbh00*CLI>

Dear, I have asterisk 1.6.2.13 in a small hardware PBX (512 GB RAM and Celeron), and last days when I call from one extension to another of the same PBX after I dial the number the rings sound after 20 seconds. In the CLI log, when I debug the AGI, I see always goes good until dialparties.agi, and after that there are 20 seconds without any log, and so the ring sound. I've read

Dears , I am running asterisk version 1.6.0.6 with asterisk Fax Version 1.6.0_1.0.10, I installed on it two channels (one Fax for Asterisk channel and one Free Fax for Asterisk) My license files located at /var/lib/asterisk/licenses/ The two channels was working fine until I restart my PC,I did not change anyof my network cards or mac Address , How can I reload the channel license and

Hello: We have a TDM2433E Digium Card (12 FXS, 12 FXO) and Asterisk 1.8.7.0 running. Everything seems to be ok but call transfers. This is the issue: *A, B, C and D are in FXS ports*. 1) A calls B. B anwers. 2) B tries to transfer the call to C dialing *2 (code for attended transfer). 3) A hears MOH. B dials number C. 4) Asterisk says the dialed number is incorrect or non existing. We tried

When I receive a call, I want to automatically park it from the dialplan so that I can retrieve it later. However, I don't want callers to be aware that they are being parked, so I want to play a ringing tone to the caller. Park() is supposed to be able to do this: Park([timeout][,return_context[,return_exten[,return_priority[,options[,parking_lot_name]]]]]) options r: Send ringing

Hey Guys! I am getting this wired error when i am calling IAX trunk. Everything works! but i want to get rid on these RED WARNING messages.. what is wrong here ? I have func_aes.so module loaded. also i remove and test but still same error. -Satish == Using SIP RTP CoS mark 5 -- Executing [7623 at from-sip:1] Macro("SIP/7527-0000000d", "orasebcamdial,7623") in

Matt, At some point you need to consider how much is too much... I run a call center with more then 125 commissioned phone sales reps and more than 60 customer service reps. We run dual servers, fiber from one provider and 6 bonded T1's from another provider. We purchase our so trunks from a wholesale company who is a major provider to resellers. Being so, their network is extremely

All; I'm running Asterisk 1.8.15-cert3 with the newest version of spandsp. I've even tried unloading that and using Digium's FFA module but I receive the same error on an outbound transmission: [2014-02-06 14:35:14] ERROR[19066]: udptl.c:294 encode_open_type: UDPTL (SIP/XXXXXXXXXXX_outbound-00000000): Buffer overflow detected (59 + 127 > 175) I only get this with one

Hi All; If I set a context other than the default context, then I do not see a generation for a configuration in the extensions_additional.conf for this context, but always the generation for the configuration is for the default context (from-internal). Normally, I have to put some Phones in a context and another Phones in a context, and give each context a privilages, but if I do this, then I

I'm trying to decide if I need to open an issue for this or if it's just a misconfiguration issue of some sort. Here's the situation - yesterday morning, I downloaded asterisk 1.8.19.1 and installed it on a fresh CentOS 5.8 installation and got a shell of a basic asterisk install setup (minimum required configuration files, etc, with no dialplan or sip peers setup yet). In the

I'm trying to declare a few date-related global variables to ease my dialplan. When I declare the following in the [globals] context of extensions.conf, I get unexpected results: YEAR = ${STRFTIME(${EPOCH},,%Y)} MONTH = ${STRFTIME(${EPOCH},,%m)} DAY = ${STRFTIME(${EPOCH},,%d)} TIMESTAMP = ${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)} If I evaluate these variables in the dialplan later, using exten