similar to: CDR Accounting Question

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 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...

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

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

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

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! 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

hello all, My problem if i have my extensions and sipusers in a realtime database it is not possible to use BLF or hinting. i see only idle or unavailable status but if the phone is ringing or in use i can't see it. Is there a fix or any workaround? Version is Release 1.4.1 regards rene -------------- next part -------------- An HTML attachment was scrubbed... URL:

Somebody can say me what i can do that the g729 is working? _____ Von: Ren? Enskat [Teamware GmbH] [mailto:ren@teamware-gmbh.de] Gesendet: Montag, 10. April 2006 10:21 An: 'asterisk-users@lists.digium.com' Betreff: G729a error when i load asterisk i got this error and cant start * with the g729 codec: Apr 10 10:21:18 VERBOSE[5873] logger.c: [codec_g729a.so]Apr 10 10:21:18

Hi, I have a basic dial plan question; Here is the scenario. Call comes through IAX and my * authenticate, then collect the digits and dials out, simple :). Here is the dial plan; [did-in] ;for did callers exten => 866219xxxx,1,Ringing exten => 866219xxxx,2,Wait,4 exten => 866219xxxx,3,Answer exten => 866219xxxx,4,Authenticate(/etc/asterisk/authenticate.txt|a) exten =>

Hi to all, I was looking in google and also in this mailing list, but I dont find the solution to my problem, so I subscribe me to the list in order to post this e-mail and find the solution. This is the scenario: GSM Phone ----- GSM Network ---- TDM2406E --- ASterisk 1.4.0 (*) -------- VoIP Provider ------- Sip Phone or H323 Phone The problem is that I am generating calls from SIP and also

Hi, On a Fedora 18, I try to launch a VM with QEMU-GlusterFS native integration. I have enable fedora-virt-preview repo, and gluster-alpha3 repo. Below the list of installed package : glusterfs-3.4.0-0.3.alpha3.fc18.x86_64 glusterfs-devel-3.4.0-0.3.alpha3.fc18.x86_64 glusterfs-fuse-3.4.0-0.3.alpha3.fc18.x86_64 glusterfs-server-3.4.0-0.3.alpha3.fc18.x86_64

Morning all, I updated to 1.4 now but it seems the oracle is not working with it? I get error with 1.2 all is fine: Mar 29 08:10:54 WARNING[3876] config.c: Realtime mapping for 'sippeers' found to engine 'oracle', but the engine is not available Mar 29 08:10:54 NOTICE[3876] chan_sip.c: Registration from 'sip:xxx@xx.xxx-xxx.de' failed for xx.xx.xx.x- Username/auth name

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

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

Hello all, I wondering if you guys could provide me with some advise. We have developed a simple management tool for a small site which is architected in the following way: Server A <==> Server B <==> Server C Server A is an asterisk server with a TE410P which connects to the PSTN as well as to a couple of SIP VoIP termination providers. Server B is another asterisk server

Hello, What i am trying to do is ..... Dail a number and ask if you wana talk to XXX press 1 and if you dont wana talk press any other key. For this purpose i am using this link<http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial> . *I am using this option :- * *M(**x**)*: Executes the macro (x) upon connect of the call (i.e. when the called party answers). IMPORTANT - The CDR

Hi All, I tried with different configurations and referred many articles to configure Asterisk with a Vonage account I have but all my attempts failed. I am a newbie and hope this mailing list will help fixing my problem and configure Asterisk. The error I get after I make a call to outside number like 18007633555 is -- Accepting AUTHENTICATED call from 59.93.69.218, requested format =