search for: aserisk

i am set var Set(CDR(accountcode)=forkcdr-test) into agiphp probe $agi->exec('Set(CDR(accountcode)=55555)'); $agi->exec('SetAccount','123123123'); and no work ... how to solutions. thanks people!!!!! -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am using Aserisk as a SIP server to interconnect differents PBX in differents sites. I am now looking for a tool that can test the performance of this solution: I mean is there a tool that enables me to test the capacity of this SIP server in terms of simultaneous calls that could be treated, the comsuption of band...

New to Aserisk 1.6 and find the 'installation tutorials' seem low to non existent. You go to the main Asterisk page (digium.org) and really just old install instructions for 1.2 are in the examples. Download links only give you asterisk itself and not dahdi or libpri which also are needed to run asteris...

Hi list, i've problems with my * server and the 4 phones which are linked to it. i've 2 grandstream bt100 with the firmware upgraded to 101, a wi-fi phone (i don't know its brand) and another ip phone i don't know its brand. with this sip.conf : [general] port = 5060 bindaddr = 192.168.100.229 context = default ;x changed from default to sip localnet = 192.168.100.0/24

Hi all, I have install G729 license to asterisk 1.4.18 and use distro debian etch 4r2. It'snot complete. when i use ldd commant to show the library it show "/usr/lib/aserisk/modules/codec_g729a.so: /lib/libc.so.6: version 'GLIBC_2.4' not found (required by /usr/lib/asterisk/modules/codec_g729a.so) when i check the GLIBC version on debian it is glibc-2.3.6 please advice me to solve the problem Thanks, ti -------------- next part -------------- An HTML atta...

...ions of the book "AsteriskFutureOf TelephonySecEdit" on page 295 onwards ) Link to the Asterisk book: http://downloads.oreilly.com/books/9780596510480.pdf) and get an error when running service asterisk start. The error is: cat: /var/run/asterisk.pid: No such file or directory . I can run aserisk fine from the non-root user. Please help Code Snippet: 1: 2: 3: 4: 5: 6: 7: 8: 9: 10: 11: 12: [root at messaging run]# /etc/init.d/asterisk restartShutting down asterisk: [FAILED]Starting asterisk: [ OK ][root at messag...

Hi all, Is it possible and easy to make a CTI server with Asterisk? Florent,

Hi, I am a newbie in asterisk. trying to configure firefly third party edition to connect to aserisk 1.0.3 im able to authenticate but cannot dial extensions. I have been reading the documentation cant seem to find the correct configs. Attached the error message and configs. What am I missing? *CLI> Urgent handler Feb 12 15:52:05 NOTICE[16537]: chan_iax2.c:5718 socket_read: Rejected connect a...

Hi.. I am trying to do something but it is giving me some hard time here. I have an IAX2 trunk to FWD which is registered and working just fine. I have => 011|. as my dial pattern to allow that. But if I want to dial a toll free number I would have to dial 011*1800XXXXXXX What trunk dial rule should I use to enable anyone to call a toll free number by simply dialing 1800XXXXXX instead of

Using some scripts that have been posted, we have been able to get paging to phones working quite nicely. However, with a few Asterisk@Home 2.5 installs, (Aserisk 1.2.4) the phones ring but never pick up. Any ideas on why or how to tweak the scripts to get the phone paging working again? Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - <mailto:kerryg@techdatapros.com>...

Hi, I followed the how-to from http://www.alcatelunleashed.com/viewtopic.php?f=44&t=840 All works fine except for Asterisk->Alcatel calls. Actually, calls from Asterisk to analog extensions on the Alcatel work. However, calls from Aserisk to digital extensions on the Alcatel 4400 do NOT work. I get this error in the Asterisk log: -- Executing Dial("SIP/4053-0823dd48", "Zap/g1/2001|30") in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g1/2001 -- Channel 0/1, span 1 got hangup...

Hi, I am using Aserisk 1.4.23.1 and trying to use SIP_CODEC to define the codec being used. I have exclusively Polycom phones for this test, and basically I want all communications to use g729 (preferred codec), except for pagine 20 phones (which busts my g729 license count). In that case I want to use gsm. I have th...

Greetings, I have a problem making a call from Asterisk to Cisco H323 PSTN gateway using H323 channel. I can call but there are no sound in both way. If I call H323 gateway directly from SJPhone I have no problem with sound. Any advice are welcome. Thanks in advance.

I am trying to integrate a Nortel Norstar system with an Asterisk service using a TDM04B card (4 fxo). So far everyting works from Asterisk to Nortel. The problem is when someone dials from the Nortel PBX to the Asterisk server. Asterisk answers the call and provides a dialtone (with DISA) but appartently the DTMF tones are not passed to asterisk and the call cannot proceed. This only

I'm looking for some info on the Asterisk Appliance. I understand it has a gui, but can I still do all the dialplan config that I'm used of doing by hand outside of the gui? If I really wanted to, could I even ignore that the device has a gui and do all my config in the files? I guess I'm just wondering if it will be as flexible as a 'vanilla' asterisk install from source on a

Say I get an ADIT 600 with two FXS8A and one FXO8A from ebay. a. Is it good for Asterisk? b. How do I connect the extensions and lines to it? Do I need a special jack? Can I get that jack in every corner? c. where can I find help for configuring it? d. what kind of backup does it have? Does it need to be reconfigured after a power outage? Shoval Tomer, IT Manager, SofTov Advanced Systems, Ltd.

Hi. I'm proposing a wireless distribution system in infraestructure mode for a valley with no cables and distant houses, respect to Asterisk, I can get a company I work with to provide me with E1 and use LCR to get 4 different telcos based on prefix. I've been reading a very interesting paper about voip on wireless, i recommend it really