similar to: 1.6.0.5: I need a really simple analog SendFax dialplan

In my quest to actually send a fax, I'm now stuck trying to send the confirm. First I send the fax: -- Executing [send at outbound-fax:2] System("Console/dsp", "env echo -e "Channel:DAHDI/g0/12036378447\\nContext:fax-tx\\nExtension: s\\nPriority: 1\\n" >/var/spool/asterisk/outgoing/call-1254012878.0") in new stack -- Auto fallthrough, channel

Hi, folks. I'm having a heck of a time trying to get outgoing T38 faxing (I don't need inbound right now) working with FFA and Gafachi. G711 faxing works (as well as can be expected over the internet), but I want the higher reliability of T38. I'm running Asterisk 10-beta1. When I drop my callfile in to make the call, I get this: -- Attempting call on SIP/18884732963 at

Hi everybody, I've installed Free Fax For Asterisk in my Asterisk box but I don't understand how it works as when using SendFax application from dialplan, I can't find how to introduce destination fax number. How this application works? Do I have to call destination fax using Dial application, detect somehow that it's a fax and then use SendFAX application specifying FAXOPTs and

On 1.6.0.18-rc3 using app_fax.so, spandsp-0.0.5, anytime I use SendFAX asterisk restarts: [Nov 15 19:00:36] VERBOSE[17013] logger.c: -- Executing [s at fax-tx-test:1] ESC[1;36;40mNoOpESC[0;37;40m("ESC[1;35;40mSIP/nhi-rive rside-sip-00000000ESC[0;37;40m", "ESC[1;35;40mContext fax-tx-testESC[0;37;40m") in new stack [Nov 15 19:00:36] VERBOSE[17013] logger.c: --

Hi, For the project I've used the Eicon DIVA card. It has 8 BRI ports, and for about 25% of the time there are 7 or 8 in use. So we want to replace it with an E1 card. Only issue is, replace it with what? The idea we have been playing with was to get a Digium E1 card (we already have bought lot of Quad E1 cards :-) and then just put it back to back against Asterisk server. And instead of

Hi, Do you have any example showing how to use SendFAX ? I can see several examples of ReceiveFAX but not a single one showing SendFAX. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081127/b41ca08b/attachment.htm

Hello, after I solved my problem with the fax processing after receiving, I got another problem while sending a fax: the header is not set properly. I use a PHP_Script to upload a PDF file and to generate a call file. A bash script is looking for existent call files in the web directory and moves them the asterisk's outgoing directory. Ok, my call file looks like this: ====== $cf_commands

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

I've been working on a way to send faxes from a bash script. It requires changing what parameters are used for each command. I keep having problems with quoted strings passed to the command. I've created a really short test script which fails on my system. commandline="sendfax -f noreply at prudentrx.com -n -x \"Test fax\" -d 13105551212 /etc/hosts" echo

On Nov 30, 2007 6:35 PM, Kingsly John <member+centos at kingsly.net> wrote: > > What you have isn't a "real" modem .. it's just a winmodem that is usable > under linux. "real" modems don't need drivers to work. > > 11/30 00:31:52 dem mdm_command: string 'AT+FCLASS=2.0' > 11/30 00:31:52 dem mdm_command: string 'ERROR' ->

On Dec 1, 2007 5:23 PM, Robert <kerplop at sbcglobal.net> wrote: > > All the more reason to quit wasting your time with that junk and go shopping for a USRobotics Courier v.everything -- external if you have a serial port available. I notice that there are several on ebay right now. If it has no manual included, download one from the mfr. > Lesson learned - I just hate to give

I followed the Faxing.txt that comes with samba and my MS Word docs print just fine but are not spooled automatically by faxspool. Does anyone have this working with samba? Thanks! Alex. -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/ms-tnef Size: 1272 bytes Desc: not available Url :

Hello, I noticed Asterisk 1.8.4.1 execute number dial twice Log == Using SIP RTP CoS mark 5 -- Executing [912066604 at sipphones:1] Set("SIP/4773-0003e920", "CALLERID(num)=2066604") in new stack == Extension Changed 4773[sipphones] new state InUse for Notify User 4701 -- Executing [912066604 at sipphones:2] Dial("SIP/4773-0003e920",

I have a dial plan that includes a company phone directory as a main menu option. If they just sit at the main menu, after 20 seconds, they are transferred to the operator. If the user picks an extension from the directory, they are transferred to the proper extension. If the called number is not available, they are transferred into VoiceMailMain. They leave a message, and hang up. The hang

Hello All, I was wondering where does Asterisk stores the blacklist numbers? I looked into the dialplan and it shows that it *"Set(DB(blacklist/${blacknr})=1)"* the number... Does it save in MySQL DB? hyperion*CLI> show dialplan app-blacklist-add [ Context 'app-blacklist-add' created by 'pbx_config' ] '1' => 1.

Morning, Have some questions regarding receiving and sending faxes... 1:st example: exten => 101,1,Answer() exten => 101,2,Wait(3) exten => 101,3,ReceiveFAX(/var/spool/asterisk/tmp/fax.tiff) exten => 101,4,System(tiff2pdf -p A4 /var/spool/asterisk/tmp/fax.tiff > /var/spool/asterisk/tmp/fax.pdf) exten => 101,5,System(mutt -s 'New FAX for you sir' -a

Has anyone implemented "hot seating" in any neat way? This where people can log in to any phone in the company and have their calls/voicemail come to that particular handset.....

Hello, I'm trying to send a tif file, using Fax for Asterisk and the call is executed, but when I get the reINVITE with T.38 data, the local server doesn't recognize that we have this capability and sends a 488 message. These are the logs: <--- SIP read from UDP:xxx.xxx.xxx.xx8:5060 ---> INVITE sip:1234567 at 10.0.0.3:5060 SIP/2.0 Via: SIP/2.0/UDP

Hello everyone... I'm trying to get up a testing pbx installation. Following instructions of what've read from the handbook and from asterisk's wiki, I wrote the dial plan as follows: [general] ; ; static = yes ;[globals] ; [default] ; exten => 0,1,Answer() exten => 0,2,Playback(fcopba1) exten => 0,3,Hangup() exten => *0,1,Answer() exten => *0,2,Record(fcopba1:gsm)

When I first to the "add extension" the "show dialplan" has the lines that say "SIP/" but after I do a "save dialplan" and a "stop gracfully" and restart the lines with "SIP/" are gone. ************************ "Show dialplan" before: ************************ asterisk01*CLI> [ Context 'default' created by