similar to: The parameter of ast_request_and_dial()

I am using Sipura 841 phones and Asterisk CVS-v1-0-06/14/05. Whenever a call comes in with blank CIDName or CIDNum the phone reports the respective variable as "asterisk". I can manually set the variables to whatever I want: CIDName (alpha-numeric) & CIDNum (Numeric). But if I try to make them blank, or null, or maybe throw some alpha characters into CIDNum, they get reported

Hello, Sometimes, when a call comes in from the PSTN through our VoIP gateway, the information that is sent to our web page that logs calls includes the original CID name instead of the one that is we expect to be rewritten on the fly using Asterisk's LookupCIDName: ================= ;extensions.conf [internal] exten => group,1,LookupCIDName exten =>

Hiya, I sent this bugfix to the asterisk-dev mailing list, and modified it as I noticed side effects, but now it appears to be finished. Nobody seemed to notice it there, so I thought I'd post here, as it seems to be something that will be needed as people update to the latest CVS version. So...read on :) Ted programmer_ted@hotmail.com P.S. Read to the very end. The original bugfix

Hello, In case there are other users of the AsteriskWin32 port... I haven't really used the AGI feature of Asterisk to run an application from extensions.conf. *Win32 supports Perl, which I don't know. Apparently, it's also possible to write AGI applications as EXE's (there's a eagi-test.exe file installed by default). => When a call comes in, I'd like an AGI

Hello everybody. Is there a way how to setup asterisk to notify caller's phone? Example: I have some numbers and names in asterisk database ( cidname, cidnum), and I want to display the name of person on my phone ( which has no addressbook, but can display chars ) which I am calling to be sure that I have dialed the right number. Thank you for any answer. Andrej

Hello When a call comes in, I'd like to fork a Python script that broadcasts a message so that users see the CID name + number pop up on their computer screen, and simultaneously ring their phones. The following script doesn't work as planned: It waits until the script ends before moving on to the next step, which is Dial(): =========== exten =>

Greetings, 2 nights ago I upgraded one of my remote servers to the latest CVS Stable, Asterisk CVS-v1-0-02/07/05-16:13:48, and my inbound and outbound caller ID stopped working. At first I thought it was my carrier, so I've been yelling at them for the past day or so. I then upgraded my core network (about 8 asterisk servers) to the latest stable as of 3am last night, Asterisk

Hello all, I recently updated * to 1.0.x from a CVS version downloaded in July, at the request of BroadVoice. Every since the upgrade, the email notifications that a voicemail has been left only contains CIDName and not CIDNum. Here's an example: "Just wanted to let you know you were just left a 0:06 long message (number 3) in mailbox 1 from PECK JASON , on Tuesday, November 16,

Thank you very much. For some reason "emailsubject" was not included in my example config. Well, it's working great now. Last question, I promise :P. Is it possible to change the date format? I want it in Norwegian. -----Opprinnelig melding----- Fra: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] P? vegne av Barry Flanagan Sendt: 22.

Hi! I'm having a ZyXEL P2000W that I'm using together with my Asterisk box (CVS from some week ago). When I get a call directly to the *-box (lars@hostaname.domain.se) I see on the console that Asterisk get the calling users name as CIDName and his SIP-address as CIDNum - but at the P2000W i only get "asterisk" as CLIP? And I can see in the P2000W that "asterisk" is

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

/**************************************************************************** * * * Always Be Conferencing (ABC) * * * * Creator: chris @ Penguin PBX Solutions * *

Hi, Can anybody tell me which version of res_perl I have to install on Asterisk 1.2.4. I tried to compile res_perl version 3.5 on Asterisk 1.2.4 and I got the following error. gcc -Wall -DRES_PERL_BASE="\"/usr/local/res_perl\"" -DMULTIPLICITY - D_REENTRANT -D_GNU_SOURCE -DTHREADS_HAVE_PIDS -fno-strict-aliasing -pipe -Wdeclaration-after-statement

Hi all, Recently, I have upgraded the asterisk as following. asterisk-1.4.13 asterisk-addon-1.4.4 libpri-1.4.1 zaptel-1.4.5.1 Usage of the server: inbound and outbound call, queue, mixmonitor, meetme, moh After upgrade, the server get segfault randomly and asterisk crash and restart itself. I got 2 core dumps of the segfault. Based on the core dump, we can't figure out the root cause to

Hi My hardware PBX run asterisk on vxworks,Because the vxworks not support perl. Now I want to add a callback function to my pbx. now it can store Caller and Called party numbers in queue when Called party is busy Then I malloc a new ast_channel to call.It is should use ast_get_channel_by_exten_locked() or ast_channel_alloc() , my program as follow,But it isn't work, anyone know how to

Hi all, I'm doing some coding, i'll be thankfull if anybody can help me here. does anybody knows if ast_request_and_dial() returns differents "reasons" when dialing to SIP and ZAP devices? For example, if the phone of the callee is BUSY, i think ast_request_and_dial() should return the same reason (int) no matter if the phone is a SIP device or ZAP device but apparently is not

Hi, I am attempting to connect to the blacklist database using PHPAgi and it always seems to hang. The code snippet I am trying is: $r = $agi->get_variable("CALLERID(num)"); $cidnum = $r["data"]; if ($cidnum < 1000000000) # No valid callerid. { exit(0); } $r = $agi->database_get("blacklist", "$cidnum"); if ($r["result"] ==

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

Why Asterisk showing (on SCCP and H323 phones) different CID related to type of Incoming channel: If incoming channel is SIP, on phone is displayed CALLERIDNUM If incoming channel is ZAP, on phone is displayes CALLERIDNAME It vas very frustrating! I lost couple hours of my time to find that my dialplan is not faulty, but asterisk is!

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