search for: rajdev

...e 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

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

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 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

...1, after that executed the following commands ./configure make clean make make install Went to asterisk folder ./configure make clean make make upgrade But could not get chan_zap.so then did the make install of asterisk. still missing the chan_zap.so Can someone please help. Regards, Sanjay Rajdev

...eturns 1 1 rows fetched Here are the details of the stuff I am using OS :- fedora core 6 kernel 2798 (Was able to build asterisk on it) asterisk-1.4.1 libpri-1.4.0 zaptel-1.4.0 asterisk-addons-1.4.0 (Also tried with or without) Can someone please help, I am very new to asterisk. Regards, Sanjay Rajdev Phone : +1 (877) 342 2329 x 1702 Fax : +1 (815) 261 5907 http://www.featherstoneinformatics.com Communications from Featherstone Informatics Group ("FIG") may transmit information that is confidential and privileged information of Featherstone Informatics Group ("FIG"). Unle...

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

...ta error!\n[%s]\n\n", sql); unlink(fullpath); goto free_res; } } } } close(fd); SQLFreeHandle(SQL_HANDLE_STMT, stmt); The value of colsize printed on CLI is 64512, Is there some limitation somewhere in FREETDS or ODBC. Can anyone please help me to get this fixed? Regards, Sanjay Rajdev -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080517/b1928701/attachment.htm

...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 => 1234,2,Monitor(gsm,/recordings)/${UNIQUEID},m) Do I have to upgrade and check or is their some other thing I can check? Regards, Sanjay Rajdev -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080421/3ea4fd1d/attachment.htm

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

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 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

...ication 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 Date: Fri, 28 Mar 2008 20:54:51 +0530 (IST) From: sanjay.rajdev at featherstoneinformatics.com Subject: Re: [asterisk-users] how to register IAX user without password To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Message-ID: <16916992.1331206717891511.JavaMail.root at mail> Content-Typ...

...ue table name=test timeout=15 monitor_join=t (yes) monitor_format=wav announce_frequency=60 retry=5 wrapuptime=20 maxlen=0 servicelevel=120 strategy=rrmemory eventwhencalled=t (yes) reportholdtime=t (yes) memberdelay=0 weight=0 Does anyone have idea what is wrong. Please suggest. Regards, Sanjay Rajdev

Hi, I just compiled and installed Asterisk-1.4.2 along with zaptel-1.4.1 and libpri-1.4.0 on a Debian machine with a TDM400P card. Everything goes ok but when I try to make a call through the ZAP channel get an error message about NO ZAP CHANNEL AVAILABLE. Ztcfg and zttool show the card correctly installed. When I tried to use the debug command ZAP SHOW, it was not present in the CLI. My

...ologoff=1000 autologoffunavail=no but it does not work It only ring twice in this case also and the caller keeps on hearing, you are first in line. If the agent re login the queue starts again. I have Asterisk 1.4.2 with zaptel 1.4.1 Can anyone Please help. Thanks in advance. Regards, Sanjay Rajdev

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

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.