similar to: Receiving faxes with spandsp question

Hello. I've been trying to setup Free Fax for Asterisk on a Debian machine with Asterisk 1.8. I have managed to register and installed the Digium modules. Sending and receiving through it have resulted in failure. The output of fax show capabilities is: Registered FAX Technology Modules: Type : DIGIUM Description : Digium FAX Driver Capabilities : SEND

Dear all, When receiving a fax, the extension is "spawned", despite nothing but positive messages (see below) The sending fax considers it a success & the verbose output of asterisk gives a "FAX_SUCCESS" and a "NO_ERROR" error in the ReceiveFax command. The problem is that all the next steps (conversion of the fax to pdf & sending it to a mailbox) are never

Hello everyone, I'm trying to send a received fax with mutt, when I try it from the Linux shel it works, but when trying with Asterisk's System command it doesn't. Successful Linux command: echo | mutt -s "New fax" earohuanca at gmail.com -a /tmp/faxes/201306191111.tif Unsuccessful Asterisk Command: same => n,System(mutt -s "New fax" elder.arohuanca at

Hello, I would like to verify if a specific SIP header exists, and if yes, extract the partial content from another header. 1. Is there a way to verify if a specific header exists? 2. How do I extract data that is between the first : and the following @? Specifically, The data looks like <sip:1234567890 at 10.0.0.1:5060> and I would like to get only the 1234567890 I tried to use the CUT()

Is anyone else using the AMI with res_fax_spandsp.so for real-time status? I am working on migrating a FAX application from res_fax_digium.so to res_fax_spandsp.so. I have noticed that the spandsp module generates far fewer AMI status events than the Digium module and the generated events contain less information. For example when sending a fax there is no longer an event for every page. There

Hi Everyone, I am having some issues getting my 7961 working with Trixbox. I have loaded the SIP code (8-3-3SR2) fine but when the phone boots up it goes into an unprovisioned state. A status message shows up and says ?Error Verifying Config Info?. I have read quite a bit on this topic (getting 7961?s to work with Asterisk and TB) and only came across a few postings where other people

Hi all, I have problem for receiving fax from multiple page fax that sent from fax machine (analog). The error is : WARNING T.30 Page did not end cleanly This is my dialplan [inboundfax] exten => s,1,NoOp(**** FAX RECEIVED from ${CALLERID(num)} ${STRFTIME(${EPOCH},,%c)} ****) exten => s,n,Set(FAXOPT(ecm)=yes) exten =>

On a test fax: -- Executing [s at incoming-fax:1] Set("DAHDI/4-1", "FAXFILE=/var/spool/asterisk/fax/20110415_1825") in new stack -- Executing [s at incoming-fax:2] Answer("DAHDI/4-1", "") in new stack -- Executing [s at incoming-fax:3] ReceiveFAX("DAHDI/4-1", "/var/spool/asterisk/fax/20110415_1825.tif") in new stack

Hi, ? We are new to IP phone firmware upgradation (Sorry if it is a re-post of previous question(s)). ? Recently we have bought a cisco 7942G IP phone. It currently has SIP 42.9-0-2SR1S firmware loaded on it. We do not see any option to configure a SIP Proxy where we can provide SIP Server (Asterisk PC/Device)? IP address (with current firmware on it) to register it with Asterisk. ? Do we need to

Anyone have the 7970 xml config for sip yet? Aaron

We have two asterisk back to back connected over PRI T1 and i am testing FAX over T1 line. Its working fine and i am getting fax file as well but i got "WARNING T.30 ECM carrier not found" is it safe to ignore this WARNING ? spandsp-0.0.6pre18 Asterisk-1.8.3.3 [from-pstn] ;Fax testing exten => 8000,1,Set(FAXFILE=/var/spool/asterisk/fax/${CALLERID(num).tif) exten =>

Trying to make the fax detection work. My current setup (with no fax) is done without Answer(), so the call is answered only when someone actually picks-up the phone. But when the incoming call is fax, I can her the tone and call is never forwarded to "Fax" extension. But... Strange thing happens when I (mistakenly) put a call on hold: -- Executing [youngandson-test at

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

Question: Naturally there are times when need to I reformat an extension in a context as such: ;Reformat add CC1 exten => _NXXNXXXXXX,1,Goto(1${EXTEN},1) -or- ;Reformat 011 with with +CC exten => _011X. ,1,Goto(+${EXTEN:3},1) It's a helpful trick, BUT there are times when I want to send the call to another context in its original un-reformatted state. Naturally the ${EXTEN}

Kevin Larsen <kevin.larsen at pioneerballoon.com> schrieb: > The phone you gave your wife is really old. Are you sure it supports SIP > OPTIONS? Can you make a call in or out to it? If you can, it is more > likely that it just doesn't support that and you can't use a qualify > statement. No, I'm not sure. And no, I can't make any call, right now... At least,

Darryl Moore <darryl at moores.ca> schrieb: > Ahh. Seen that before! That suggests to me that you don't have your > sip.conf records setup right. > > What's your sip.conf look like? Well, here what I wrote in my sip.conf: register => 00493511111111:MYSECRET at pbxluca/00493511111111 register => 00493512222222:MYSECRET at pbxfax/00493512222222 register =>

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Guenther Boelter <gboelter at gmail.com> schrieb: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA256 > > On 05/31/2015 02:31 PM, Luca Bertoncello wrote: > > Hi list! > > > > Now all works as expected, at least in the simulation I did with > > AsteriskNOW. Hopefully it will work later, when Deutsche Telekom

Skipped content of type multipart/alternative-------------- next part -------------- Index: pbx.c =================================================================== RCS file: /usr/cvsroot/asterisk/pbx.c,v retrieving revision 1.14 diff -u -r1.14 pbx.c --- pbx.c 19 Apr 2003 02:41:22 -0000 1.14 +++ pbx.c 21 Apr 2003 02:27:43 -0000 @@ -713,6 +713,8 @@ { char *first,*second; char tmpvar[80] =

Hi list! So, new day, new problem... I tried right now to call from my cellphone a peer in my Asterisk. The cellphone has correct credentials, but it's NOT registered on my Asterisk, now. I just tried to call a peer in my network, from a peer not yet registered. And it works... :( The very curious thing is, that I can't find how the call will be accepted... Every section in my dialplan

Hi everyone. We have an Asterisk server running Debian Squeeze, with Asterisk v1.8.13.1 (basically, the Debian Stable version for Squeeze, but with some minor source code changes specific to our site). We're trying to upgrade to 11.13.1 (The Debian Stable version for Jessie), but I've run into a snag when compiling res_fax_spandsp (and yes, we really need that module). The old