similar to: Execute a script upon registration

We have the ability to create random UID's on own system through a custom CGI API. These UID's are written to individual sip configuration files based on the account name, so for instance sip_TEST.conf, sip_TEST2.conf, and sip_TEST3.conf, etc. Many of these UID's are created on the fly and at random times throughout the day. Right now, I have it setup to do a reload every night.

Hi, I want to prevent Asterisk from sending the h extension across to the SIP provider or to prevent it from hitting the script at all. The SIP Provider does not know what to do with the h extensions once it receives it. My SIP Provider takes all digits and forwards them off to a softswitch for processing. Everytime a call hangs up, it complains about running AGI scripts on hungup

It seems that when setting qualify = 200 or qualify = yes in the database for a sip friend/peer, RealTime does not update the registration status like it should. I also have several peers which have been offline and Asterisk still reports them as registered, even though the registration seconds are only 200. Asterisk Ver: CVS HEAD 12/1/2004 Layout of sip_buddies: mysql> describe

Hi, I am experiencing one-way audio from: SIP Device ----> Asterisk -----> Cisco 7200 The Cisco 7200 has a VXE+ card that will allow you to do SIP. I can pass audio from SIP Device to Asterisk through the Cisco 7200 to the other end, but the Cisco 7200 does not return any audio back to the SIP Device or Asterisk, it seems. I have tried upgrading to 12.3T IOS version, but no

Hi i have a similar dialplan: exten => 99,1,Gotoif(....?2:3) exten => 99,2,Meetme(100) exten => 99,3,Meetme(100|options) i'd like to do something like: exten => 99,1,Gotoif(....?2:4) exten => 99,2,Meetme(100) exten => 99,4, ... exit ... exten => 99,3,Meetme(100|options) How can i exit the dialplan? I won't use meetme twice! Thanks nik

Hello, I have a strange situation: I can talk to other SIP phones and via ISDN to the outside, but I don't hear playbacks or the voicemail messages. Asterisk show in the cli, that the corresponding files are played, but I hear nothing at all. Here is as simple example: [monkeys] ??? exten => 99,1,ANSWER() ??? exten => 99,2,PLAYBACK(tt-monkeys) ??? exten => 99,3,HANGUP() The phone

Hi, I having a problem compiling the wrapper for oh323. I am running Debian, kernel version 2.4.18-bf2.4. The pwlib version I have is 1.6.6 and the openh323 version I have is 1.13.5. I execute the following commands first before attempting to compile the wrapper: pwlib_1.6.6: make both openh323 1.13.5 ./configure make opt asterisk-oh323 0.6.2 make

Hi, i am having problem on playing asterisk sound file on my new installed asterisk.. i have the following extension , if i call from any SIP / IAX phone playback or voicemail doesnt play anything .... but when i dial 102, I hear the MP3 music .. exten => 99,1,Answer() exten => 99,2,Playback(prepaid-welcome) exten => 99,3,Hangup() exten => 101,1,VoiceMailMain() exten =>

Why does voicemail prompt me for an extension instead of just asking my password? [voice-mail] exten => 99,1,VoicemailMain(${EXTEN}@inside) exten => 99,2,Hangup Paul Mahler pmahler@signate.com <mailto:pmahler@signate.com> <http://www.signate.com/> Signate, LLC PO Box 60430 Palo Alto, CA 94306 VoIP Systems, Training & Consulting

Hi, When I do a sip show peers in the cli, the status is lagged. This peer its behind a satellite link with 600/900ms of delay. May I change some parameter in the Asterisk? Some times I cant make a phone call from the remote site to my central site. Thanks Este mensaje ha sido analizado por C4I Mail Server en busca de virus y otros contenidos peligrosos, y se considera que esta limpio.

> I would like to have all SIP phones to work on prepaid basis > and without need to dial any access number, instead I would > like to use the phone as normal dialing only the destination > number, for example 00464090510. I use the AccountCode for authentication. This is how, for example: exten => _00XX.,1,DeadAGI(astcc.agi,${ACCOUNTCODE},011${EXTEN:2}) > Once the call is

strace on file access in asterisk shows that * is not even attempting to access the voice files. If I *manually* load app_playback.so, app_macro.so, and then pbx_config.so, I they will load and I get a dialplan. Ok, that's a problem -- autoconf is clearly not working, or there's some other related issue. So I try to use the demo and do "dial 500". This should connect and

I am trying to implement a Centralized Call Waiting System. I have red some document about asterisk group features to manage group and category of a sip channel. I have done a lot of test about it but always it doesn't work correctly if I transfer the call. This is the macro code I use for inbound calls. [macro-test] ; ${ARG1} - technology something like SIP ; ${ARG2} - resource.

Hi everyone, I am new programming with sound codecs. I am making a c++ application that needs to convert a .wav audio file to a .ogg one. If you can send me a demo or something similar I will be very pleased. Thank you for your time, Wanhaven. _________________________________________________________________ Busca desde cualquier p?gina Web con una protecci?n excepcional. Consigue la Barra de

Hi everyone, I am new programming with sound codecs. I am programming in Borland C++ Builder 6 and I would like to convert .wav files to .ogg ones with flac. I have downloaded the FLAC C++ API, and I need some help, is there any function that converts .wav into .ogg??? Please I will be very pleased if you can help me, or showing me an example or something that I can use. Thank you for your time.

Hi, We are having an issue here. It seems that whenever we initialize Asterisk on our network, the router that the Asterisk server is connected to (Cisco 7200) crashes and loses it configuration. This has happended five times and each time we have tested it, it is always when Asterisk starts up. Has anyone else seen this problem? It is very odd because this is a very good router and we

Hi, I am using the asterisk-oh323 wrapper and I am looking to allow registration of h323 endpoints and allow Asterisk to act as a gateway. The idea is simple: H323 endpoints would register with Asterisk. They each would have their own internal extension (like SIP). If a H323 endpoint dials an outbound extension, then the h323 call gets routed to a H323 Gatekeeper which then terminates

Hi, I get this error when trying to dial an outbound extension from a sip phone: -- snip -- -- Executing Dial("SIP/2003-02d1", "OH323/3215435249@h323gk|20") in new stack -- H.323 call to 3215435249@h323gk with codec ALAW -- Called 3215435249@h323gk 0:33.283 H225 Caller:8143908 PWLib Assertion fail: Null pointer reference, file

Full_Name: Zivan Karaman Version: 2.7.2 OS: Windows XP Submission from: (NULL) (195.6.68.214) I'm puzzled by the readRegistry function. Shouldn't the "hive" argument be something like c("HLM", "HCR", "HCU", "HU", "HCC", "HPD") rather than c("HLM", "HCR", "HCU", "HU",

Working under Windows XP, I am compiling a package called 'pgirmess' with the command rcmd build --binary --auto-zip pgirmess I have this message error after having listed: functions text html latex example chm .... zipping help file hcc: not found cp: cannot stat 'c:/TEMP/Rbuild365620874/pgirmess/chm/pgirmess.chm': No such file or directory make[1]: *** [chm-pgirmess] Error 1