similar to: Speeding up the dial of DTMF's in SIP channel

I'm developing an application that monitors the state of the incoming calls using Manager events. So, as a part of it, I need to "override" the control of the extensions by the dialplan itself. The problem is that, if I don't declare the incoming extension, Asterisk hangs up the call by default. So I want to know if there's some kind of "ManagerControl() application

I've set voicemail almost successfully, only a minor detail remains :-) I can't get the dates in my local language (spanish). In sip.conf, zapata.conf and voicemail.conf, I've set: language=es and my locale is "es" also. However, the days and months names still appear in english in the emails!!! Thursday 11 de May de 2006, 18:49:34. instead of Martes 11 de mayo de

Is there a way to access only certain positions in the $EXTEN variable? I'd like to filter my international calls based on the destination country: My dialplan looks like this (1XX0. is the international calling convention for Chile) exten => _1XX0.,1,Dial(SIP/${EXTEN:4}@external_provider) But, I'd like to, depending on the destination country (digits 5 and eventually 6 of EXTEN),

I've defined my dialplan as showed below. My internal lines are numbered as 12345XX, and internal users can call another by the entire 7-digits extension, or by just last 2 digits. [invalid] exten => _X.,1,Playback(pbx-invalid) exten => _X.,2,Hangup() [internal] include => invalid exten => _XX,1,Dial(SIP/12345${EXTEN}) ; Short alias for internal lines exten =>

I'm trying to set the Pickup feature. I'm setting my extensions.conf as: exten => _*.,1,Pickup(SIP/${EXTEN:1}) but if, for example, extension 03 is ringing by a call made from extension 01, and I try to pick it up from extension 02 (by dialing *03 from extension 02), I can see in the Asterisk console (Verbosity set to 10): -- Executing Dial("SIP/01-512c",

I'm communicating a softphone (SJPhone) to a Grandstream phone GXP-2000. The codec order on each one is the next: SJPhone: GSM - iLBC - PCMA - PCMU GXP2000: G729 - GSM - PCMA - PCMU (I have a G729 license, so there's no problem with transcoding G729) In my sip.conf, I've defined the following codec order: disallow=all allow=g729 allow=gsm allow=g726 allow=alaw allow=ulaw And my

I've the following dialplan. exten => _XX,hint,SIP/${EXTEN} exten => _XX,1,Dial(SIP/${EXTEN},10,j) exten => _XX,2,VoiceMail(${EXTEN}@default,u|j) exten => _XX,3,Hangup() exten => _XX,102,Goto(110) exten => _XX,103,Playback(pbx-invalid) exten => _XX,104,Hangup() exten => _XX,110,VoiceMail(${EXTEN}@default,b|j) exten => _XX,111,Hangup() exten =>

Is there a way in Asterisk (I guess there's, it's only I can't figure out how :-)) to: 1.- Generate a call to channel 1 (example, to PSTN v?a an E1 card, using Zap/g1) 2.- Generate a call to channel 2 (example, an internal SIP extension). 3.- Once both channel have answered, connect the call between them. This way, I can, for example, play audios in both channels before they are

I've noticed the following situation: In two softphones, I've configured the next codec order for each one softphone 1: 1 - PCMA 2 - GSM softphone 2: 1 - GSM 2 - PCMA and in Asterisk, the order is: disallow=all allow=gsm allow=alaw If I call from softphone 1 to softphone 2, I presume that Asterisk should do transcoding (canreinvite is set to no):

I've an Asterisk 1.2.4 installation, where I've also installed the G729 codec license. I'd like to upgrade that installation to 1.2.5, but I'm not sure if I'll lost the license in the process (and if I'll be able to recover it later!!!). Is there any special consideration I've to keep in mind in this case, or should I just run the typical "make + make

I'm trying to set a dialing rule in my dialplan. As a part of it, from my point of view, this works wrong priorityjumping=no [test_context] exten => 1234,1,Dial(SIP/test,15,G(text_context,1234,2),j) ; With "j" flag exten => 1234,2,Playback(digits/2) exten => 1234,3,Playback(digits/3) exten => 1234,102,Playback(digits/4) In this case, if I dial the extension, and

*************************************************************************************************** CISTI'2013 DOCTORAL SYMPOSIUM 8th Iberian Conference on Information Systems and Technologies Lisbn, Portugal, June 19 - 23, 2013 http://www.aisti.eu/cisti2013/index.php?option=com_content&view=article&id=64&Itemid=68&lang=en

*************************************************************************************************** CISTI'2013 DOCTORAL SYMPOSIUM 8th Iberian Conference on Information Systems and Technologies Lisbn, Portugal, June 19 - 23, 2013 http://www.aisti.eu/cisti2013/index.php?option=com_content&view=article&id=64&Itemid=68&lang=en

Hello all, I'm having a quite simple configuration like: SIP provider <=> asterisk SIP <=> lan Everythings works fine but sometime I can't get incoming call. here are some of the logs from set debug 25 set verbosity 25 sip show debug and sip.conf and a part of extension.conf thanks in advance Reliably Transmitting (NAT) to 212.27.52.5:5060: OPTIONS sip:freephonie.net

Is there a way to send DTMF's to a channel before the call is answered? For example, send DTMF's to a SIP channel after the 180 Ringing or 183 Session Progress have been received from it, but before the 200 OK, or in the E1 side, after the Q931_ALERTING is received, but before the Q931_CONNECT. If I use Dial(SIP/XXXX,D(my_dtmfs)), it will wait until SIP/XXXX have answered to send the

After many hours today thinking that I had placed a bug into my dialplan, I realized that for some reason DTMF tones are simply not making it into asterisk! Calling into my pbx transmits crystal-clear audio in both directions. But dialing DTMF's from pstn->pbx is unsuccessful, while pbx->pstn works fine. The tones simply don't make it through. Tiny brief fragments are all. Please

Hello everybody, I need help, I have a problem with conferences in asterisk, when many people are in a conference sometimes there're users pressing phone keys and this action emits a sound (DTMF of the phone keys), so, I need to find the way of not listening this sound.. I'm using MeetMe(variable,pFX).. I tried whithout "F" but it doesn't work because users continue

Hi! anyone knows where to get some info for kernel (maybe via sysctl) and or samba tuning for high performance ? I have read all the samba docs available, so aim looking for others tips besides the tcp tunings usually applied in smb.conf ? i am setting a server on a client site, with many clients (about 100), and i am using a real server hardware (an HP netserver with xeon procesor@2.8Ghz, 1Gig of

Hi All, I'm not found nothing about how to solve this problem. I have installed Ms Project 98 without problem, but when I attempt to run it fault showing the error message "Not enough memory ....". Can anybody help me? I'm using wine version 20041019 in debian gnu/linux with a 2.6.7 kernel. Thanks. ?lvaro Pe?a.

Hi, Is there a way to enable the usage of UNIQUEID CDR field using a MySQL database backend for storing CDRs without having to recompile asterisk-addons as stated here http://www.voip-info.org/wiki-Asterisk+cdr+mysql ? After version 1.4 it is said in release that it can be done (not sure if it applies to mysql backend) How would it be the syntax in cdr.conf? I tried this without success in