similar to: fax - conversion problem

*@home has this for it's incoming fax macro --- start snip --- [ext-fax] exten => in_fax,1,GotoIf($[${FAX_RX} = system]?2:analog_fax,1) exten => in_fax,2,Macro(faxreceive) exten => in_fax,3,system(tiff2ps -2eaz -w 8.5 -h 11 ${FAXFILE} | ps2pdf ${FAXFILE}.pdf) exten => in_fax,4,system(mime-construct --to ${EMAILADDR} --subject "Fax from ${CALLERIDNUM}

Does anyone know of any obscur reference for detecting an incoming fax. I currently have AMP running and everything else is working great. Installed the spandsp patches and software... using the default AMP extensions.conf, I start sending a fax, I hear it pick up and transfer to voicemail after 20s. Fax is set for system... Here is the detail from the extensions.conf [global] FAX_RX = system

I've upgraded Asterisk from CVS, spandsp and app_txfax and app_rxfax but i'm still unable to send/receive faxes :-(. I'm using amp_fax to send and this is what i get from logs: Sep 6 11:02:52 VERBOSE[10750]: -- Attempting call on Zap/g1/666 for application txfax(/var/tmp/ast_fax-1125997371.10240.1804289383.0|caller) (Retry 1) Sep 6 11:02:52 DEBUG[10750]: Dialing

Hello, Is there or can anyone provide a comprehensive guide (designed for Linux/Asterisk novices) to installing/setting up Asterisk in order to support Fax2Mail service? In my case, I would like Asterisk to receive fax calls to predefined numbers (ranges) and to associate each of these numbers to email addresses. Thank you in advance. David --------------------------------- Yahoo!

Looking for someone with a successful experience similar to this; I have a need to cross connect a 3COM NBX PBX PRI interface to asterisk, but over a long distance. We do not need any IP connectivity and the solution requires G.711u audio so there is no benefit to using IP. Has anyone here successfully cross connected any PBX PRI interface expecting NI2 PRI signaling B8ZS/ESF with an

Hello, I have spandsp setup to accept incoming faxes and receiving tif files via Email. Using tiff2pdf, or tiff2ps -a2 or even tiffsplit, the last page of the fax is cut off and the quality of the text looks "squished". I "figure" it's a tiff parsing thing, as opposed to a problem with my spandsp installation (heh). Has anyone experienced the same thing, or can

Hello Guys, I'm somewhat of a newbie and am desperately seeking for some help... I've managed to get asterisk up and running on my server, and signed up with a broadvoice account... I'm having no problem dialing and communicating between extensions, but whenever anyone tries to call my broadvoice account, they are greeted by no ring or anything, but rather simply a direct to

I would like to know which kind of solutions are available, both software and hardware, for human operator in an asterisk environment. The operator should be able to provide the basic standard operation, like to transfer calls and to see if the extensions are busy or not and so on. Thanks in advance, Andrea Chi ricevesse questa mail per errore e' gentilmente pregato di cancellarla.

How many times do you ask for help here before getting a respone? Every single thing I do No matter what I get busy extensions. I am willing to pay someone to help here. Anybody got a clue? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050802/d0d1326c/attachment.htm

I all. I have configured a pair of * servers, sip connected each other Mi problem is the following If on the first * i configure a queue containing phone number of the second * (i.e with a round robin strategy) I have non problem as far as all phones are online. If one of the remote phone number is unavailable, when the round-robin strategy touch that phone the call is answered by the voicemail

Is there a way to 1) disable asterisk from writing in the "full" log ? ( /var/log/asterisk/full ) or 2) implement a log rotation or similar of the full log ? I see the full log grows a lot (about 100 MB per Month) thanks in advance, Andrea Chi ricevesse questa mail per errore e' gentilmente pregato di cancellarla. Visitate il sito http://www.frameweb.it

Is there a way to run a macro in a GotoIfTime statement ?? from the wiki documentation it seems not, but...... I would like to do something like this: ......... 554,3,GotoIfTime(08:30-14:30|mon-wed|*|*?Macro(exten-vm,novm,567)) it does not work, as expected from documentation any workaround to call an extension WITHOUT vm (also if vm for that extension is present...) as a consequence of a Time

Hi all, I would like to eliminate about 150 lines in log /var/log/messages) every time a call is placed/received If I type, on the asterisk console, set verbose 0 the lines in the log disappear, but it appears to me too drastic as a method.... The lines shown in the log don't appear (at least to me) very critical: no problems at all are shown. Isn't any way to turn off this debug ? I

Hi all, I just arrived in Italy from Cebit, qhere I spoke with digium and Beronet people. They told me to try to use the mISDN stack to drive beronet and the new upcoming digium ISDN Cards. SO I searched, find http://www.beronet.com/download/card_installation_guide.pdf, and I immediately got the error: asterisk01:~ # cd /usr/src/install-misdn/ asterisk01:/usr/src/install-misdn # make install

....of chan_misdn ? I found a bug, and I don't know where to report it. Andrea Chi ricevesse questa mail per errore e' gentilmente pregato di cancellarla. Visitate il sito http://www.frameweb.it

Hi all, I would like to know if it exists the possibility to send to different context according to the caller IP Addres I receive H323 calls, and I have to route this to different devices according to the caller ip. I tried to use the context=first-context alias=999999 context=second-context alias=888888 but I was not able to succed in this; Moreover, I think the keyword alias is related to

I realize I cannot use inband audio for phones (voicemail and internal ivr, password for external trunks and other thing not working) So I put everywhere rfc2833. Doing this, anyway, make any EXTERNAL IVR NOT working. I see a lot of posts about this, but no solution, becouse using inband audio (which works for outside...) breaks inside IVR Is it possible to define to use inband audio ONLY on

Hi, I am trying to define DID Routes via AMP (last version 1.10.008) I succeded in defining single DID route, one per extension, let's say i.e. DID number 0101234567 set destination to extension 567 DID number 0101234555 set destination to extension 555 and so on Now I was trying to define only one route to a custom application DID number 0101234XXX routes to Custom-App

Hi all, I would like to know if there is a solution to this question. Scenario: Two asterisk servers connected across the Intenet ( in SIP or IAX mode, no matter) with both of them having static ip addresses Then I add a second link (with another provider), with another NIC at both side, and again both of them having static ip addresses. Is there a way to tell asterisk to use both of these

Hi all. We have Snom phones which do have a defined key in order to drop incoming call WITHOUT answering. Pressing that key, a "SIP/2.0 486 Busy Here" message is sent back. We have other phones (I.E. DECT Siemens C450IP, or ATCOM 320 or other) which DO NOT have any key to do that (or the key does not work, as is with Siemens C450 IP ): you have to answer and immediatly after hangup the