similar to: Require only GSM Codec

Has anyone figured out the way of getting the caller id for BSNL on Asterisk 1.4.2 I have tried following link http://bugs.digium.com/view.php?id=6683&nbn=24 but was not able to get it, although did not ge any error too. I always get the caller id as asterisk. Can someone please help. Regards, Sanjay Rajdev

I have a CISCO 7912 phone, the LED on the phone does not glow when there is new voicemail, can we configure Asterisk to have the LED glow on new Voicemail. Regards, Sanjay Rajdev

I have some Analog card on a PCI slot of a remote computer, Is their a way I can figure out remotely the name of the card. I have FC6 installed on the machine. Regards, Sanjay Rajdev

I am having problem while having asterisk work with ODBC (Postgres) The error that I am getting is "config.c: Realtime mapping for 'sippeers' found to engine 'odbc', but the engine is not available" I really donot know what has went wrong. I have set the ODBC connection properly I have verified it using :: [root@asterisk ~]# echo "select 1 " | isql asterisk

Few days back I installed Asterisk 1.4.2 with Zaptel 1.4.0. All SIP accounts were working fine, today I tried to install a fxs Sangoma A200 card and got the following error. [Apr 12 01:15:17] WARNING[31018]: channel.c:3024 ast_request: No channel type registered for 'Zap' [Apr 12 01:15:17] WARNING[31018]: app_dial.c:1090 dial_exec_full: Unable to create channel of type 'Zap'

What is the best alternative for getting the IVR and other prompts recorded for Asterisk. Regards, Sanjay.

Hello, is it possible to rout ALL RTP Data over Asterisk, like SIP1 <---RTP---> Asterisk <---RTP---> SIP2 I know it seems quite useless. But I want to simulate a IAX -> SIP connection and have no Phonecard installed on my computer ;) Thanx, Kalle

Do anyone has an idea about an open source SIP API written in C# that can communicate with Asterisk, to call out? Regards, Sanjay.

Dear Sanjay, Sorry sanjay i miss to explain completely. My PC2PC mean is Dialer2Dialer i want to allow call between Dialer with out any registry and authentication through IAX. so i need to setup Asterisk accept calls from any user and users can call to each other without any password and registration. please help how can i configure Asterisk using IAX in this regards. thanks, Asif Message: 9

I am generating an outbound call through the Manager API and bridging it to an internal Extension, my problem is I am not able to find the logs for the call generated by the Manger API, Since on the same Asterisk server there are many users connected and I am receiving lot of Events back, not able to recognize which was the call generated by me as same time multiple users are dialing out.

What version of SIP do Asterisk 1.4.x uses. Regards, Sanjay.

I am trying to write a customized app using C that would fetch voice file from SQL Server 2000 using ODBC and FREETDS. Currently I am only able to fetch first 63 KB chunk from the DB, and not able to fetch the rest of the file, below is the code that i am using to do so, fd = open(fullpath, O_RDWR | O_CREAT | O_TRUNC, 0770); if (fd < 0) { ast_log(LOG_WARNING, "Failed to write

I have setup Asterisk on 2 Fedora Core 8 machines, and have made it to record all incoming calls. One of the box that have Asterisk 1.4.18 is properly merging calls and the other box that has Asterisk 1.4.15 is recording the calls but not merging them, I have made sure that SOX is installed on the box. Here is the Dialplan of both the machines : exten => 1234,1,Answer() exten =>

I am planning to write a module to find if a Special Information was detected or not. Can anyone please help me to figure out the below fields? 1. The Frequency of a frame 2. Length of frame in milliseconds Thanks in advance. Regards, Sanjay.

Hello Everyone, I have a 2 Asterisk Servers, one in US and another in India. Once someone from US calls, call hit US server and then is forwarded to India which then is answered by someone. i.e. Caller --> US Asterisk Server --> India Asterisk Server --> Employee(India) The Employee in India decides that the call was for Employee in US, so he transfer the call to the employee in US.

Is there anyway to setup a queue with only one agent (device) which is always logged in. So when a call hits that queue the device will ring (if not already on a call) or will be put in the queue if the call is already in place? Thanks Miles -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have queue set up in realtime on Asterisk 1.4.2. Below is the senario that is happenening :: I have created a test queue with only one agent. Once I call the test queue the agents phone rings if the aagent is logged on. everything till here is fine. Now if the agent does not pick up the call, the call automaticaly disconnects after 15 secs as set for the queue, till here also it is fine. But

Hello, We're in the process of setting up an Asterisk server, and are having echo problems. We have a Digium TE110P, and have tried the MG and MARK2 AGGRESSIVE echo cancellers, with a variety of gain levels and training times, and with both trunk and 1.2 branch versions of Zaptel, Libpre, and Asterisk. In all cases, callers from the PSTN hear their own voice echoed back after 1.5-2 seconds;

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello Asterisk Gurus! I have a very simple question. I've just started playing around with Asterisk and BSD box. I also have grandstream ip phone and installed asterisk from ports. Now I'm on my very first steps to configure Asterisk. The question is: " How do I make Asterisk communicate with my Grandstream hard phone?" Thank you in

I am attempting to use an IAX2 channel between two Asterisk systems. This would seem to be a normal thing to do. I actually want to trunk traffic between the two that are in remote locations. However, I have started with what I think is a simple configuration, which should allow for one way calling. Attached are the pertinent parts of my configuration files. I am attempting to place a call on