similar to: dial plan question

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

Software zur Erkennung von "Spam" auf dem Rechner priamus.teamware-gmbh.de hat die eingegangene E-mail als m?gliche "Spam"-Nachricht identifiziert. Die urspr?ngliche Nachricht wurde an diesen Bericht angeh?ngt, so dass Sie sie anschauen k?nnen (falls es doch eine legitime E-Mail ist) oder ?hnliche unerw?nschte Nachrichten in Zukunft markieren k?nnen. Bei Fragen zu diesem

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 everyone, Im having some troubles getting work Asterisk as SIP Client and a Huawei softswitch as SIP server. I already got my asterisk registered to the Huawei. Im working with a Sipura SPA 2000 registered to Asterisk. When im trying to make an incoming call from the Huawei to asterisk it rings but when i answered the call drp down inmediatly. The sip debug finally show this

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",

Software zur Erkennung von "Spam" auf dem Rechner priamus.teamware-gmbh.de hat die eingegangene E-mail als m?gliche "Spam"-Nachricht identifiziert. Die urspr?ngliche Nachricht wurde an diesen Bericht angeh?ngt, so dass Sie sie anschauen k?nnen (falls es doch eine legitime E-Mail ist) oder ?hnliche unerw?nschte Nachrichten in Zukunft markieren k?nnen. Bei Fragen zu diesem

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...

Software zur Erkennung von "Spam" auf dem Rechner priamus.teamware-gmbh.de hat die eingegangene E-mail als m?gliche "Spam"-Nachricht identifiziert. Die urspr?ngliche Nachricht wurde an diesen Bericht angeh?ngt, so dass Sie sie anschauen k?nnen (falls es doch eine legitime E-Mail ist) oder ?hnliche unerw?nschte Nachrichten in Zukunft markieren k?nnen. Bei Fragen zu diesem

We recently had our PRI installed, we currently have 100 toll-free's pointing to it. I have almost everything working great but.. I have setup the first few numbers we want to use coming in from the PRI and they work great, but.. What I want to do is setup an extension with pattern matching to answer for any numbers called that are pointed to our system and PRI but not yet in

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 All, In our small office calls to the PSTN are currently sent via Asterisk and a Linksys SPA3102 (1 x FXO and 1 x FXS): SIP Phone --> Asterisk --> Linksys SPA3102 --> PSTN If the PSTN is in use on SPA3102 I need a way to get the call to then route out over IAX termination. SIP Phone --> Asterisk--> Linksys SPA3102 --> PSTN (In Use)

>> >> I do not think you can legally make clang a system compiler on >> Windows without licensing headers/libraries from Microsoft. >> So inability to parse all of the SDK headers is a non issue to me, >> can't use them anyway. > > > Care to cite a source? Please don't (either of you) randomly speculate on the legality of clang/LLVM. To be

Hello to all my Asterisk brethren, I have been implementing an Asterisk system for home/office use, and with help from the Wiki and Groups I am doing well. However, with my server and two X-Lite phones I can't figure out how to set a DND or call forward from my extension to another. Probably a real easy thing to do and I am missing something (or a complete section from my

Hi All, Does any one know of a way to make a three way call from Asterisk using X-Lite. I need the ability to be able to call someone on the PSTN using my IAX provider then bring another person from a local extension into the call if needs be? I believe most three way calling is done using a feature of the phone, and X-Lite doesn't look like it supports this. Can this be

I''ve got my first server deployment with ZFS. Consolidating a pair of other file servers that used to have a dozen or so NFS exports in /etc/dfs/dfstab similar to; /export/solaris/images /export/tools /export/ws ..... and so on.... For the new server, I have one large zfs pool; -bash-3.00# df -hl bigpool 16T 1.5T 15T 10% /export that I am starting to

Hi I can't for the life of me work out why this is not working. When in the campon contect if you hit a DTMF key 2 you get moved to the exten => 2 defined in the mainmenu context not the exten => 2 defined in the campon context. What is wrong? The same happens if you hit key 1. [campon] exten => _*1XXX,1,Answer exten => _*1XXX,2,SetCallerID(${CALLERIDNUM}) exten =>