similar to: [Bug 15462] New: Need some way to detect end of swf playback

I'm dialing DTMF's in a SIP channel using the options: [sip.conf] dmtfmode=info [extensions.conf] exten => _XXXXXXX,1,Dial(SIP/gateway,,D(${EXTEN})) (this is a custom SIP gateway, which receives the DTMF's sent from softphones through Asterisk, and based on them, build the destination PSTN number). My problem is that Dial send the DTMF's to the SIP/gateway user at a rate

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

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.

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 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 =>

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*************************************************************************************************** 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

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 =>

Please, I can`t see my messages. Can anyone confirm if it is reaching to the list? Thnaks! ?lvaro ----- Original Message ----- From: Alvaro Lassance <lassance@sidercom.com.br> To: <samba@lists.samba.org> Sent: Thursday, December 27, 2001 1:39 PM Subject: SETEUID > > > Hello! > > > > Anyone knows how I install the "seteuid method" in a RH 7.0? >

Hi, I'm sorry if this is a silly question, but i know that is not possible in Courier, so, I need to check if I can do it with Dovecot. Can I use PAM authentication, witch get the users data from a external database (like mysql)? I've found many ways to do this stuff disconnectedly (like pam authentication with passwd ), but i can put all together? I can't use the passwd...

Hi folks, We are working with Dovecot 2.0.9 with Solr support and there is a thing, a little strange for us. Let me explain. We have this conf for Solr: plugin { ... fts = solr fts_solr = url=http:// solr.domain:8983/solr/ break-imap-search quota = maildir ... } With 'break-imap-search', Dovecot connects with solr, solr indexes all, searchs are

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 have a huge dataset(15k X 18) and tried to use the heatmap in R to examine the patterns. However, I found that heatmap and heatmap.2 gave me completely different outputs. Here are the codes: ------------ > dim(as.matrix(data.dcpm)) [1] 15462??? 18 > > heatmap(as.matrix(data.dcpm), col=topo.colors(100)) > heatmap.2(as.matrix(data.dcpm), col=topo.colors(100), key=TRUE,

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 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 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

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

http://bugs.freedesktop.org/show_bug.cgi?id=10470 --- Comment #4 from Riccardo Magliocchetti <riccardo at datahost.it> 2008-04-09 07:10:59 PST --- With latest git the game is playable, Klaus do you have any chance to try it? -- Configure bugmail: http://bugs.freedesktop.org/userprefs.cgi?tab=email ------- You are receiving this mail because: ------- You are the QA contact for the