similar to: Pattern Matching

Hi all, This is my first post! I'm newbie, yesterday I installed Asterisk@home, and I got lot of Kernel panics, after trying to reinstall it, this messages dissapeared. My problem now is that the default password for maint user in AMP is not working... I got this error message when I try to connect via another computer in the same network, after trying to log -> maint/password: FORBIDDEN

Hi. I have the next configuration... I dial from my analog phone in the TDM400P to extension 102, and the second agi works about 1 out of 10 times, the other nine it gives me these error on the asterisk console: -- Starting simple switch on 'Zap/2-1' -- Executing Macro("Zap/2-1", "receivecall") in new stack -- Executing AGI("Zap/2-1",

hi i updated asterisk today via svn no i can'T start asterisk i get core dumps. i have to comment some modules then i can start: noload => format_au.so noload => format_mp3.so noload => format_pcm_alaw.so.so noload => format_pcm_alaw.so compiling was fine just some warnings somebody has any idea? -------------- next part -------------- An HTML attachment was scrubbed...

Hello all, I try to use a agi script to get a variable from * und put them into a script which gives me another variablke and put this in *. My problem is now it seems the var ID is empty coz i always jump into the result 0 loop. The $MSN should be in the SetCIDNum. #!/usr/bin/php -q <?php include("/var/lib/asterisk/agi-bin/phpagi.php"); $agi = new AGI(); $ID =

Hi, i can't compile the latest svn update from zaptel: /lib/modules/2.6.14-1.1653_FC4smp/build make -C /lib/modules/2.6.14-1.1653_FC4smp/build SUBDIRS=/usr/src/zaptel modules make[1]: Entering directory `/usr/src/kernels/2.6.14-1.1653_FC4-smp-i686' CC [M] /usr/src/zaptel/zaptel.o /usr/src/zaptel/zaptel.c:6193:5: warning: "CONFIG_ZAPATA_DEBUG" is not defined

My number is not submitted. I updated my asterisk but this error still occurs coz of the "" in the SetCallerID tag thats why it will be a empty SetCallerID is submitted. Is there a fix to correct this error? -- Executing SetCIDNum("SIP/31-752a", "4989427xxxx") in new stack -- Executing SetCIDName("SIP/31-752a", "4989427xxxx") in new stack

I have a problem with the cdr. We terminate through a pstn provider to the pstn network. The problem is now the cdr accounts the connection to the gateway. Coz the gateway is answering our call and then forward to the pstn number. So i have billsecs all the time even it is only ringing or so. Somebody has a solution for that? -- Executing Dial("SIP/1000114-fcf8",

Hello I patche dmy 7970 with the current SIP image i have 2 lines on it via sip and 6 hint speeddials but it seems thats only a speeddial no infos about busy status or so comes to the speddial button. somebody can help me?

Hi, I am moving my call center to 1.4. Previously I was recording calls in agents.conf with the following config recordagentcalls=yes recordformat=wav createlink=yes So I had the filename in all calls which was *connected to agents*. I am looking for a similar functionality for 1.4. I am now recording calls using the following configuration. [general] persistentmembers = no eventwhencalled =

I have put canreinvite=no for all my internal SIP-clients in sip.conf because I want Asterisk to be in the middle of the RTP-stream so he can provide MusiconHold and so... Now, what the Asterisk CLI tells me when I make a call from my one internal SIP-phone to another internal SIP-phone is : Verbosity is at least 25 == Spawn extension (intern, 51, 1) exited non-zero on

Hi! This teamware mailclient (teamware.com) that we use at the office has problems adding files as attachments from our Samba 3.0.2a share. The attachment file browser sees the files but fails to add them as attachments (with an error message: "valid.stf - file not found" regardless of the file name in question). I tracked this thing down to scandinavian letters (if your mail

Hi All, I have a CentOS server that I am trying to configure Asterisk on 1.4 on. Everything seems to go ok, with regards to compiling Zaptel, Libpri, Asterisk (will be using kernel 2.6 timer and ztdummy) Unfortunately I can't insmod / modprobe ztdummy. [root @xyz src]# modprobe ztdummy FATAL: Module ztdummy not found. FATAL: Error running install command for ztdummy

Hi all, Somehow my reply didn't reach this list. I've attached the conclusion. Cheers, Cristian. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/vorbis/attachments/20080831/9f467742/attachment.htm -------------- next part -------------- An embedded message was scrubbed... From: "Alasdair King" <alasdairking at

Hi i have an issue getting agentLogin working /etc/asterisk/queues.conf member => Agent/1001 member => Agent/1002 extension.conf exten => 110,1,Wait,1 exten => 110,2,AgentLogin() now, i call 110 by a firefly client, trying to login in as 1001 agent: Aug 12 16:31:36 DEBUG[1103408048]: chan_sip.c:4423 build_route: build_route: Contact hop: <sip:sip3@192.168.1.151:5060> --

Guys. Ive seen on my asterisk messages log that asterisk has shutdown itself about 12 times in 5 days... The logs show nothing but: [Jun 28 09:40:02] WARNING[3172]: Unicall/4 event Drop call [Jun 28 09:40:02] WARNING[3172]: Unicall/4 event Release call [Jun 28 09:40:02] VERBOSE[3172]: [Jun 28 09:40:02] -- Unicall/4 released [Jun 28 09:40:02] VERBOSE[3084]: [Jun 28 09:40:02] Asterisk cleanly

Hello: I'm having issues with my CentOS 5.1/Xen installation. If I run the xen-bridge, I seem to get flaky ethernet. By flaky I mean everything seems fine from the host machine, but if I attempt to contact the host machine from another remote machine (eg. my laptop which is on the same subnet as the xen machine, separated by 10 ft of cable and a router) I either get extremely high ping

Hi All! I have two [*]s, and both work OK as a simple local PBX. Now, I try to link them using IAX. Let's call those babies a1 and a2. From a1, I want to dial a phone connected to a2. Both boxes have a fixed IP address, and use standard port 5036, say a1.mystrx.com and a2.mystrx.com. Where I (obviously) get confused, is when it comes to inbound, outbound, registration, etc. Taking a hint

Hello all, I'm a developer in Assistive Technology for blind and other disabled people. I produce and distribute some Free software called WebbIE and the Accessibility suite: http://www.webbie.org.uk Working with blind people, I'm interested in low-cost technologies using audio. Ogg Vorbis is therefore of interest to me. I'd like to install the necessary components for Ogg Vorbis

Hello, I am using Queue application with multiple agents in each queue. I want to set the CDR(userfield) for each cdr based on the agent answering the call. Is it possible to do this? Thanks

Hello, I'm using Asterisk with Realtime extensions and ODBC CDR. And I'm have some trouble with the CDR userfield that is not changed when using the SET command in the realtime dialplan. In my dialplan (extensions.conf, the file) I'm setting the userfield like this : exten => s,n,Set(CDR(userfield)="X") Later, my dialplan switches to the realtime part and this is an