similar to: VoiceMail application: "j" option not working as I supposed

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

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 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'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

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'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

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 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

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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

Is there a way to distinguish, in the answer of the Dial command, when a channel is not available (for example, an unregistered but valid SIP user) v/s when the dialed channel is inexistent, even when it matches an extension? For example, I've the following simple dial plan: exten => _XX,1,Dial(SIP/${EXTEN},10,) exten => _XX,2,GotoIf(DIALSTATUS = CHANUNAVAIL?4:3) exten =>

Hi Rsync Support, Recently we encountered issue on our prod environment because the rsync seems hanging, it took time building the list.Previously the rsync process was working before 10:24am not until 10:25am. See sample log below. We have one source server and the data files will be rsync to 2 webservers. Please advise what could be the cause of the issue. Please let me know if you need

Hi, I currently I have a Samba share configured as follows: [pub_fileshare] comment = Public fileshare path = /u02/pub guest ok = Yes writeable = Yes There is a subfolder under /u02/pub called /u02/pub/expenses/hardware that I need to make read only. How do I do this? I am new to using Samba. I configured the share /u02/pub/expenses/hardware using the configuration below. This works as it is

Ok guys & gals here is the scenario: 1.) Host RHEL 4 U3 2.6.9-34.0.2.EL 2.) OCFS2 latest version 3.) Successfully formatted & mounted OCFS2 filesystems on 2 nodes /dev/sdb1 /u02/oradata/usdev/voting /dev/sdc1 /u02/oradata/usdev/data01 /dev/sdd1 /u02/oradata/usdev/data02 /dev/sde1 /u02/oradata/usdev/data03 4.) Downloaded & installed ASMLib 2.0 on both nodes 5.) Ran

Running ocfs v 1.0.9-PROD12 on RHAS 2.1, kernel 2.4.9-e.25smp. I have (what look to ls and rm) as empty directories. But when I try to rmdir, returns "Directory not empty". This has gone through a reboot (several actually). Any ideas what I can do? [root@delphi1 tmp]# ls -l /oradata/u02 total 514 drwxr-xr-x 1 oracle dba 131072 Feb 4 2003 bogus drwxr-xr-x 1 oracle

hi, mount -t ocfs2 -o datavolume,nointr -L "oracrsfile" /u02 when i mount linux partation using above command i recieve the following error mount.ocfs2: Invalid argument while mounting /dev/sdd1 on /u02. Check 'dmesg' for more information on this error. thanks in advance zeeshan

Dear all, I've configured Asterisk Voicemail, but after some tests I realised that when some call is sent to the voicemail of someone which username begins with "j" letter, Asterisk gives me the error: WARNING[5865]: app_voicemail.c:2412 leave_voicemail: No entry in voicemail config file for 'ohn' (for a called user named john, for example) Is this some kind of