similar to: CVS-HEAD and Caller ID -- Pulling my hair out!

I'm having a problem with the callerID that the polycom IP600 phones are displaying. I would like to modify the CIDName and leave CIDNumber as exactly what the phone call came in as(provided they aren't hiding callerID). Most of the calls will be going to the queue, but a few will go directly to the SIP phones. I've done a various combinations of using SetCallerID(),

Hi, I've experiencing this kind of problem. Actually, my asterisk is running perfectly. I've tested it, and I called some computer in my LAN. Then I enter the CLI and entered these commands - show modules - modules status (or so.. I forget) - restart now After I enter the last command, the CLI is exiting and nothing happened. Then I try to run the asterisk with command - asterisk But

I have a pretty basic Mandrake 10.2 w/KDE 3.2 and I installed Asterisk-1.0.1-2mdk. I installed the source of main and contrib from ftp, so at the install time I accepted all the packages needed to be installed too. The installation went smooth, but when I try to execute asterisk (#asterisk -vvv) I encounter few warnings I end with an error. At this point I didn't touch any conf file, I was

Good evening, I am just getting started with Asterisk. I have it installed, and I believe I am on the right track, overall, to get it working, but I can't get the linejack to answer any calls. At this point, all I'm trying to do is have Asterisk answer an inbound call on my linejack, /dev/phone0, and play a greeting or tone. I figure, once I am able to get asterisk to actually answer the

Hi, I've been having a little problem with my asterisk servers, I have 4 identical asterisk servers setup (same hardware, same OS, same config). Once in a while (once or twice a day) one of the server crashes on the cron job reload. But I realized this only happens on 3 of the 4 servers. Tried to spot the difference between that one server that wasn't crashing. The difference I found was

Hey folks, I have a T100P card with a PRI; when doing outbound dialing over the PRI, I can use SetCIDNum(2024561414|a) to force caller ID to display as "The White House" on a land line. This is apparently done as a reverse lookup by Verizon, as I do not hand the PRI the words "The White House" -- 202-456-1414 is an actual White House number. Using SetCIDName("Flying

Admittedly this is probably pretty stupid of me, but there are just some things I can't understand by reading documentation. Any suggestions or recommendations about how to handle my problem are greatly appreciated. I'm trying to achieve the same functionality as my Nortel PBX, without rewriting much 'C' code. In my dialplan I'd like to compare two variables as a means of

Hi all, I tried to set the calleridname of an incoming call to get different incoming labels displayed for different incoming numbers. This does work for hidden number-calls so I can set the displayed CIDName on my cisco7960 from "CID withheld" to "abc CID withheld" If the incoming CID isn't hidden it works to use SetCallerID but not to change only the CIDName with

Hi everyone, I am running asterisk on red hat linux 9 box. The sound card is Intel 82801db AC' 97 audio and the module is i810_audio. It runs well with other applications like xmms and the standard tests deliver a sound . I have also tried to record voice and that works well too. 1-)Now when i run asterisk and i dial out an extension to play any sound there is none. The same thing

when i issue asterisk -vvv command i get this error please help regards Barbra [app_softhangup.so] => (Hangs up the requested channel) == Registered application 'SoftHangup' [codec_lpc10.so] => (LPC10 2.4kbps (signed linear) Voice Coder) == Registered translator 'lpc10tolin' from format 7 to 6, cost 50 == Registered translator 'lintolpc10' from format 6 to 7,

I have an analogue trunk to an AT&T Definity. It has a DISA context defined. From a Definity handset call the analogue port extension 1008 and wait for dial tone from asterisk. It takes between 3&4 rings. Likewise from Asterisk SIP handset <PBX Access No><PBX Extn> takes nearly 10 secs to ring. Is this configurable? Ian Cowley -----Original Message----- From:

hi michael, here are the core dumps. only kphone works when 0.5.5 and * cvs. audiocodes and msn messenger all cause seg faults when calling ccm thru * (or vice-versa) ~kelvin [chan_oh323.so] => (OpenH323 Channel Driver) == Parsing '/etc/asterisk/rtp.conf': Found == Parsing '/etc/asterisk/oh323.conf': Found 0:00.004 OpenH323 Wrapper OpenH323 Wrapper

For those folks who are still using it -- I updated the cid_rewrite script. I noticed that two of the providers were "iffy" and one had changed format a little while ago. It's working again. http://muware.com/asterisk has the latest (1.2.0) Enjoy, -- JM

Sorry if this is a resend, but it didn't appear to go the first time. > Sorry if this is not the correct place to post this..... > > > I have downloaded the cid_rewrite scripts that are located at: > http://www.muware.com/asterisk/ to my AAH v1.1 system. > > I apologize for my ignorance, but it says that I need to modify the > agi_config.php, but doesn't

Hi folks, I think my little agi script is ready for the big one-oh-oh. Available at http://muware.com/asterisk is cid_rewrite-1.0.0. This agi-script does the following: - Standardize incoming caller-id numbers to adhere to US dialing code; NANPA numbers are reformatted to 1+10, international numbers become 011<country-code><number> (this is customizable with a little PHP knowledge).

Dear ALL, Any clues or tips for the following gdb messages. [root@localhost asterisk]# uname -a Linux localhost 2.4.22-1.2115.nptlsmp #1 SMP Wed Oct 29 15:30:09 EST 2003 i686 i686 i386 GNU/Linux localhost*CLI> show version Asterisk CVS-HEAD-09/22/04-11:19:09 built by root@localhost on a i686 running Linux [root@localhost asterisk]# gdb asterisk core.13089 GNU gdb Red Hat Linux

Hello, I'm trying to fill CID data into the astdb using AsteriskWin32's asterisk.exe, to no avail: The batch file stops after the first line, and just waits: ---------------------------------------- rem c:\cygroot\mystuff>import.bat rem rem c:\cygroot\mystuff>C:\cygroot\bin\asterisk.exe -rx 'database put cidname 123 "My cellphone"' rem rem Asterisk module

Hi All, I'm looking for a way to filter the AstDB cidname family to show only those entries with a specified area code in the Asterisk CLI. If this were a SQL database it would be something like: SELECT number, name FROM cidname WHERE number LIKE '1234%' I've tried "database show cidname 1234*" and substituted "%", "$", "-" for the wildcard

I have a new asterisk install (1.0.7) - and in case it's relevant I'm not using autoload option in modules.conf. Generally all is working well. However, when I make a call from my softphone and try to leave a message, the message is cutoff after a few seconds (whenever I pause for 1 second between words). Strangely, when I use an analog phone connected to my ATA, I can record as long as

I have found that I can make the phones display any one word on this second line by adding a fromuser=<word> in sip.conf. This really isn't good enough though. When you look at the received calls or missed calls directory, each item has two lines, the first is the CID name, and the 2nd is supposed to be the CID number. However, if it is asterisk, or some other word, when you hit the