similar to: LED does not glow on new Voicemail

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 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'

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

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

Hello All, I would like to only use the gsm codec for all the calls, is it possible I want to use minimum possible bandwidth as we have most of calls over Internet. Regards, Sanjay Rajdev

Hallo wine users, I just found a (maybe!?) very unkown project for better wine handling. It is named glow (- some GLasses Of Wine) and enables wine users to start their applications with different versions of wine easily. The motivation for this python-script is of course the problem that some satisfactorily supported windows applications does not any longer run so well in later versions of wine.

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

Hello LLVM community, We have been working hard on a new domain specific optimizing compiler, and we are pleased to announce that we have recently open sourced the project! We would like to introduce you to Glow, an optimizing compiler for neural networks! This new compiler is built on the hard work of this community and we would like to thank all of the contributors to the LLVM project. We

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

Very cool! The first thing that jumps out to me is how tidy and modular the code structure is. The code feels very familiar (stylistically, organizationally, etc.) to me as an LLVM developer. One thing that wasn't at all clear to me is how this is different/similar to TensorFlow XLA (previously mentioned on this list). Can you briefly compare and contrast this with TensorFlow XLA? -- Sean

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.

I'm trying to get Jade Dynasty to work properly with Wine. Most of the bugs I've fixed or are to minor to worry about. There is one problem I can't seem to fix. Glows on weapons are bright white from most camera angles. It appears the glows are reflecting something, but I don't know what. Certain monsters glow white as well. The bug does not happen on the character select screen

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.

What version of SIP do Asterisk 1.4.x uses. 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 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 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