similar to: How to make the sum of a ${VARIABLE} + 1 ??

Hi list, I need some help to improve my cdr, now in my company are asking me how to know which of my phone numbers are most used when receiving calls from the PSTN and incoming the IVR was thinking about using userfield field, and I'm trying to do, I have at the moment 4 channel DAHDI ; DAHDI CHANNEL 3=23XXXXX6 context=in callerid=asreceived group=1 signalling=fxs_ks channel => 3

pbx.c:3143 pbx_extension_helper: No such label 'outgoing|PHONE NUMBER|1' in extension 'PHONE NUMBER' in context 'phones' [Apr 20 15:43:15] WARNING[11793]: pbx.c:8650 pbx_parseable_goto: Priority 'outgoing|PHONE NUMBER' must be a number > 0, or valid label PHONE NUMBER = the number I called. This dialplan worked fine in version 1.4. Michael

hi, All one thing confused me a long time. when i change the extensions.conf file. why not take effects after restart the asterisk? details as follow: my dailplan is : [95040] exten => _95040XXXXX,1,Set(CALLINNUM=${CALLERID(dnid)}) exten => _95040XXXXX,n(start),Answer exten => _95040XXXXX,n(welcome),Background(${welcomefile},,123) ... exten => i,1,Playback(invalid) exten =>

I run a hotdesking system based on the example from Asterisk: The Definitive Guide. Calls come into the [hotdesk] context, which verifies the phone has a logged in user and sends the call to users,${EXTEN},1 if there is a user logged in. The [users] context then includes several other contexts for internal/external call handling, as follows: [users] include => internal include =>

Hello list, how come on my Asterisk 1.6.2.11, I have no help available ?! asterisk*CLI> core show application Dial -= Info about application 'Dial' =- [Synopsis] Not available [Description] Not available [Syntax] Not available [Arguments] Not available [See Also] Not available Kind regards, Jonas. -------------- next part -------------- An HTML attachment was scrubbed...

Hello. I'm using asterisk 1.4.30. I've found this patch for app_queue.c : https://issues.asterisk.org/view.php?id=11700 Can I easily implement this by issuing : */wget 'https://issues.asterisk.org/file_download.php?file_id=17192&type=bug' -O - | patch -p0/* ?? Does this mean I have a "patched" asterisk ? (I ask this because some applications require a

Hello list, whenever I press the #-key I hear a voice saying 'transfer'. How can I use the #-key without this voice-message or without having it the function of unattended transfer ?! The T or t option is not set in my Dial()-command so I don't know where this transfer is coming from in the first place. Kind regards, Jonas. -------------- next part -------------- An HTML

Hello list, my CLI is spammed with : [Sep 13 08:31:38] > doing dnsmgr_lookup for 'ssw6.itsp.tld' [Sep 13 08:31:38] > doing dnsmgr_lookup for 'ssw6.itsp.tld' [Sep 13 08:31:47] > doing dnsmgr_lookup for 'ssw4.itsp.tld' [Sep 13 08:31:48] > doing dnsmgr_lookup for 'ssw4.itsp.tld' [Sep 13 08:31:49] > doing dnsmgr_lookup for 'ssw4.itsp.tld' [Sep

Hello, what does this message mean ? [Oct 12 14:03:32] DEBUG[9064] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for public_ip:2049 I find this in my debug log file when "core set debug 25". Is something failing, or is this just informative ? Kind regards, Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello list, has anyone else also noticed spontaneous reboots ?! I noticed this today and also yesterday. Can't really see if there is a fixed time between the reboots. Normally al of my SIP peers are registered. When I put up the CLI today I saw that a lot of SIP accounts where UNREACHABLE and needed to register again (what they slowly did). These are realtime SIP peers that reside on

Just done a clean install of rc5 on a totally new machine and found the following:- /etc/init.d/asterisk start errors on line 109 - there is no 0 before $VERBOSITY as in the other lines. More interesting is that after make samples I have no iax2 available. Dave Cotton

I downloaded the latest 1.8 (27922) but got some segmentation faults. The first one was when it loaded cdr_odb, and so I changed menuselect not to compile that one, but the second one was when it tried to load chan_agent and so I stopped there to see if anyone else was seeing this. The agents.conf is all commented out except for [general] . Anyone know what is happening? Thanks. P.S. I deleted

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> <meta http-equiv="content-type" content="text/html; charset=ISO-8859-1"> </head> <body text="#000000" bgcolor="#ffffff"> <font size="+1">Does anyone have links to the most recent audiocodes

How can I let asterisk immediately dials a trunk when off hook?

Hello list, I need to know how to deal with a redundant network with only one asterisk server, which is receiving registrations from the end points on both of its ethernet ports. This means extension 201 is registering both from eth0 and from eth1. Is there a way/software which can act as a middle man between asterisk and the ethernet ports, and by default sends registrations to asterisk only

I am having a one way audio issue with xlite clients behind NAT. They can connect to the server and make calls but no audio is heard on the other end. my sip conf [general] context=default bindport=5060 bindaddr=0.0.0.0 srvlookup=yes canreinvite=no[tomfmason] type=friend secret=secret callerid="Thomas Johnson" <XXXX> host=dynamic nat=yes canreinvite=no disallow=all allow=gsm

Hi, Is it possible to record say 30 seconds of audio and then have LumenVox convert to text ? or any available tool open source for speech to text . Regards Dhaval -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100914/b56c3d9c/attachment.htm

Hi Everyone, I have a provider whose DID used to come into the box just fine but recently stopped working. Nothing has been changed on our end. Here is what I get when doing "sip set debug peer PROVIDER": Sending to 123.123.123.123 : 5060 (no NAT) ^^^^ That is ALL I am getting with sip debug turned on. With Allow Anonymous SIP set to YES, then the call comes in properly and you see

Evening, Has anyone seen a how-to on getting Asterisk to work with Google Talk and Google Voice? Thanks

Hello, I read on the wiki : Asterisk 1.8 will allow to read SIP response codes in the dialplan via *${HASH(SIP_CAUSE,<channel-name>)}*. Additionally make sure you're using the destination channel, not the source channel. But when I use this in my dialplan, this 'variable' is empty. Dialplan : exten => h,n,NoOp(sip cause = ${HASH(SIP_CAUSE,${CHANNEL})}) exten =>