Re: [asterisk-users] Call Placed through Manager connecting before the call connects.

I am using Asterisk 1.4.19.2 on Fedora Core 8, with a Sangoma A200 card. Regards, Sanjay Rajdev - Original Message - From: Sanjay Rajdev [EMAIL PROTECTED] To: Mailing List Asterisk asterisk-users@lists.digium.com Sent: Thursday, May 22, 2008 6:16:17 AM GMT +05:30 Chennai, Kolkata

Re: [asterisk-users] Call Placed through Manager connecting before the call connects.

what can be wrong? Regards, Sanjay Rajdev - Original Message - From: Sanjay Rajdev [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Cc: Mailing List Asterisk asterisk-users@lists.digium.com Sent: Thursday, May 22, 2008

Re: [asterisk-users] Call Placed through Manager connecting before the call connects.

Is there no one who can even comment on below? Regards, Sanjay Rajdev - Original Message - From: Sanjay Rajdev [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, May 22, 2008 8:47:13 PM GMT +05:30 Chennai

[asterisk-users] Fax solution for Asterisk

was not able to find anything that could answer my questions. Regards, Sanjay Rajdev ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http

Re: [asterisk-users] Fax solution for Asterisk

How about outbound faxing. Regards, Sanjay Rajdev - Original Message - From: Steve Totaro [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, May 21, 2008 8:04:46 PM GMT +05:30 Chennai, Kolkata, Mumbai

Re: [asterisk-users] Fax solution for Asterisk

of SpanDSP. Regards, Sanjay Rajdev - Original Message - From: Steve Totaro [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, May 21, 2008 8:12:20 PM GMT +05:30 Chennai, Kolkata, Mumbai, New Delhi Subject: Re

Re: [asterisk-users] Fax solution for Asterisk

Further more I think we will have to license it to for using it on more than one channel. I am looking for something totally open source. Regards, Sanjay Rajdev - Original Message - From: Sanjay Rajdev [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion

Re: [asterisk-users] Asterisk Database Handling

logs to see if Asterisk has thrown any exception. they can be found on Linux at /var/log/messages Regards, Sanjay Rajdev - Original Message - From: Alex Balashov [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent

[asterisk-users] Call Placed through Manager connecting before the call connects.

. Regards, Sanjay Rajdev ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Fetching Binary data from SQL Server

in FREETDS or ODBC. Can anyone please help me to get this fixed? Regards, Sanjay Rajdev ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http

Re: [asterisk-users] Fetching Binary data from SQL Server

Tilghmanm, Thanks a lot, I have changed the value in FREETDS and it worked. Regards, Sanjay Rajdev - Original Message - From: Tilghman Lesher [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, May 17

Re: [asterisk-users] x100p card or similar in India

We have been using Sangoma A200 for about an year now with BSNL connection. I don't know if you can get it in India directly as in our case it was brought from US directly. Regards, Sanjay Rajdev - Original Message - From: Amit Patel [EMAIL PROTECTED] To: asterisk-users

[asterisk-users] Is there a way to have Manager Bridge Channel without being connected

call originated by manager even if a telco message is played. Regards, Sanjay Rajdev ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http

Re: [asterisk-users] Best Linux distribution to use in Asterisk server

I have been using FC6 for the past 1 year without any problem. Regards, Sanjay Rajdev - Original Message - From: equis software [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, May 9, 2008 8:49:23 PM GMT

Re: [asterisk-users] Basic modules of Asterisk

Thank Russell, I will try to manage it through the modules.conf file. Regards, Sanjay Rajdev - Original Message - From: Russell Bryant [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, May 8, 2008 4:11:00

Re: [asterisk-users] Asterisk Restarting due to segfault

the branches in it, but if someone from your side can take the effort to change this It would be great help for others. Regards, Sanjay Rajdev - Original Message - From: Russell Bryant [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com

[asterisk-users] Which Cepstral Voice to license

Which Cepstral voice is best for Asterisk? We need to license one. Regards, Sanjay Rajdev ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http

Re: [asterisk-users] Which Cepstral Voice to license

We are looking for a female voice. Regards, Sanjay Rajdev - Original Message - From: Matthew Gibson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, May 9, 2008 2:35:24 AM GMT +05:30 Chennai, Kolkata

[asterisk-users] Basic modules of Asterisk

I just want to Run Asterisk with the basic required modules, What can I do to achieve so? My only requirement is to run SIP clients and the Dictate Module. Regards, Sanjay Rajdev ___ -- Bandwidth and Colocation Provided by http://www.api

Re: [asterisk-users] Asterisk in Production ?

We are using Asterisk 1.4.13 in production, were we have almost 30 SIP users on a Asterisk box, we are also using IAX to communicate between main Asterisk server and the other. we use Queues, Conference too. Regards, Sanjay Rajdev - Original Message - From: Benoit Plessis [EMAIL

Re: [asterisk-users] Mixmonitor recording issue

I had a similar problem. In my case Asterisk was crashing due to MixMonitor() and then automatically restarting. I have never found a alternative solution to record the calls. Regards, Sanjay Rajdev - Original Message - From: Rahul Yadav [EMAIL PROTECTED] To: asterisk-users

[asterisk-users] Asterisk Restarting due to segfault

to the root of the problem or fixing it? Thanks in advance. Regards, Sanjay Rajdev ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com

[asterisk-users] Monitor not merging calls

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 ___ -- Bandwidth

Re: [asterisk-users] Monitor not merging calls

John, Is their something that I can change on my side to get this working ? Jared, I thought MixMonitor() was for Queue, Can you let me know how to use it? Thanking you for replying. Regards, Sanjay Rajdev - Original Message - From: John covici [EMAIL PROTECTED] To: [EMAIL

[asterisk-users] Monitor v/s MixMonitor

What is good for recording all the incoming and outgoing calls, Monitor() or MixMonitor(). Regards, Sanjay Rajdev ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options

[asterisk-users] C# SIP API to Comiunicate with Asterisk

Do anyone has an idea about an open source SIP API written in C# that can communicate with Asterisk, to call out? Regards, Sanjay. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE

Re: [asterisk-users] C# SIP API to Comiunicate with Asterisk

This work with Asterisk Manager Interface. I want to implement basic phone functionality in C#. Regards, Sanjay. - Original Message - From: Matt Watson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, April 4,

Re: [asterisk-users] C# SIP API to Comiunicate with Asterisk

Can you Please refer me to any, the one that I found are all either in Java/C. Or if they are in C# they are not opensource. Regards, Sanjay. - Original Message - From: Grey Man [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion

Re: [asterisk-users] C# SIP API to Comiunicate with Asterisk

Thanks a lot, will try this out. Regards, Sanjay. - Original Message - From: Grey Man [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, April 4, 2008 4:58:56 AM (GMT+0530) Asia/Calcutta Subject: Re:

Re: [asterisk-users] Simple Question

No It does not require. Regards, Sanjay. - Original Message - From: Drew Miller [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, March 31, 2008 9:17:19 PM (GMT+0530) Asia/Calcutta Subject: [asterisk-users] Simple Question Does AMD (answering machine detect) need

Re: [asterisk-users] how to register IAX user without password

Create a User and a Peer on both the machines for each other. e.g IAX.conf on PCa [pca2pcb] type=peer host=[IP OF pcb] username=pca2pcb serect=pca2pcb12345 qualify=yes [pcb2pca] type=user context=default auth=md5 secret=pcb2pca12345 deny=0.0.0.0/0.0.0.0 permit=[IP of pcb] qualify=yes ON PCb

[asterisk-users] How is uniqueid computed

Can anyone let me know how the uniqueid for a call is computed in asterisk? Regards, Sanjay. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:

Re: [asterisk-users] How is uniqueid computed

Thanks Mindaugas. Regards, Sanjay. - Original Message - From: Mindaugas Kezys [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, March 18, 2008 10:26:37 PM (GMT+0530) Asia/Calcutta Subject: Re: [asterisk-users]

[asterisk-users] Logs for Call generated by Manager API

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

Re: [asterisk-users] Logs for Call generated by Manager API

, Sanjay. - Original Message - From: sanjay rajdev [EMAIL PROTECTED] To: asterisk-users asterisk-users@lists.digium.com Sent: Friday, March 14, 2008 7:26:00 PM (GMT+0530) Asia/Calcutta Subject: [asterisk-users] Logs for Call generated by Manager API I am generating an outbound call through

Re: [asterisk-users] Logs for Call generated by Manager API

Thanks Lee, Will try to match on Parameter received in message. Regards, Sanjay. - Original Message - From: Lee Jenkins [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, March 14, 2008 11:27:51 PM (GMT+0530)

Re: [asterisk-users] Logs for Call generated by Manager API

I would just try parsing the message in my code to make it work. I know this is not a full proof solution but is ok for now, till I get something better. Regards, Sanjay. - Original Message - From: Mark Hamilton [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial

[asterisk-users] Best alternative for getting prompts recorded.

What is the best alternative for getting the IVR and other prompts recorded for Asterisk. Regards, Sanjay. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:

Re: [asterisk-users] Best alternative for getting prompts recorded.

Thanks everyone for the reply. Till now we had simple IVR so we recorded it ourself. Now I have a requirement where customer needs a customized message to be played to customer. I am basically looking for some Text to Speech software that would be cost effective (most probably a open source)

Re: [asterisk-users] Best alternative for getting prompts recorded.

Thanks a lot everyone, I will go ahead and try Cepstral. Regards, Sanjay. - Original Message - From: Steve Edwards [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, March 12, 2008 2:54:50 AM (GMT+0530)

[asterisk-users] Want to know Frequency and lenght of Frame

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. ___

Re: [asterisk-users] Detecting SIT (Special Information Tone) on outbound calls

For reference of SIT please check http://en.wikipedia.org/wiki/Special_information_tone Regards, Sanjay. - Original Message - From: sanjay rajdev [EMAIL PROTECTED] To: asterisk-users asterisk-users@lists.digium.com Sent: Friday, February 29, 2008 8:35:08 PM (GMT+0530) Asia/Calcutta

[asterisk-users] Detecting SIT (Special Information Tone) on outbound calls

Is there a way to detect SIT (Special Information Tone) when making an outbound call. Regards, Sanjay. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:

[asterisk-users] Can AMD detect Service Provider Message.

Is there a way to detect Service Provider message such as invalid number, using AMD or some other application. Regards, Sanjay. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or

[asterisk-users] Question regarding AGI

I have questions about AGI. 1. When Using CONTROL STREAM FILE command with all the parameter, I could not find any way to * or # in the DTMF, it only returns if any digit is pressed, even if I set forward and rewind digits to BLANK () 2. When I call out using ZAP, is their a way to find if

[asterisk-users] Sip Version

What version of SIP do Asterisk 1.4.x uses. Regards, Sanjay. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit:

[asterisk-users] Transfering IAX context

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

[asterisk-users] Detecting card on the PCI Slot

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 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk

Re: [asterisk-users] Detecting card on the PCI Slot

I have installed FC6 on it, want to configure it with Asterisk. It had some driver earlier but the machine has been formatted yesterday, so no idea. Also I am new to Linux. Regards, Sanjay Rajdev Phone : +1 (877) 342 2329 x 1702 Fax : +1 (815) 261 5907 http://www.featherstoneinformatics.com

Re: [asterisk-users] Detecting card on the PCI Slot

Thanks for the suggestion, I figured out the cards. I have 2 Digium TDM400P card and a Sangoma A101 single port card on the machine. Any suggestion on installing them. Regards, Sanjay Rajdev - Original Message - From: Gordon Henderson [EMAIL PROTECTED] To: Asterisk Users Mailing List

[asterisk-users] Can two card be configured on same machine.

I have a 2 sangoma cards that need to be configured on a same server, one is a T1 and another is a for PSTN line. Is this possible, if so please help. Regards, Sanjay Rajdev ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users

Re: [asterisk-users] Help with IAX

Can you send IAX.conf of both the systems Regards, Sanjay Rajdev - Original Message - From: Malcom Kemp [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, 30 May, 2007 9:05:34 PM (GMT+0530) Asia/Calcutta Subject: [asterisk-users] Help with IAX

Re: [asterisk-users] socket_process: Received mini frame before first full voice frame

Never received a response for this from anyone. This is being seen more frequently now. Please Suggest. Regards, Sanjay Rajdev - Original Message - From: Sanjay Rajdev [EMAIL PROTECTED] To: asterisk-users asterisk-users@lists.digium.com Cc: asterisk-dev [EMAIL PROTECTED] Sent: Friday

[asterisk-users] AgentCallbackLogin not working with 1.4.4

that no agent is online. Is it so that we require to install a T1 Card (or some other) to make this work, just wanted to ask this because I was not able to make 1.4.1 Meetme work until we install a T1 card. Regards, Sanjay Rajdev Phone : +1 (877) 342 2329 x 1702 Fax : +1 (815) 261 5907 http

[asterisk-users] socket_process: Received mini frame before first full voice frame

Anyone any idea why do we keep on getting chan_iax2.c:7535 socket_process: Received mini frame before first full voice frame Regards, Sanjay Rajdev ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list

Re: [asterisk-users] Iaxy clicking

What OS (operating system are you using) Regards, Sanjay Rajdev - Original Message - From: Matthew Yingling [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, May 11, 2007 2:59:01 AM (GMT+0530) Asia/Calcutta Subject: [asterisk-users] Iaxy clicking Hi, I have three

[asterisk-users] Call In queue stucks

. 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 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE

Re: [asterisk-users] BSNL caller ID (India)

Yes, As I have mentioned below I tried the link http://bugs.digium.com/view.php?id=6683nbn=24 but was not able to make it work. Regards, Sanjay Rajdev - Original Message - From: Steve Murphy [EMAIL PROTECTED] To: sanjay rajdev [EMAIL PROTECTED] Sent: Thursday, April 19, 2007 4:37:39 AM

Re: [asterisk-users] BSNL caller ID (India)

Tzafrir, Can you Please let me know if the zapata.conf below is correct, or do I have to change something. Regards, Sanjay Rajdev - Original Message - From: Sanjay Rajdev [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Cc

Re: [asterisk-users] Problem with queue

Thanks Philipp, I tried making it 5000, and it worked. Once again thank for your help. Regards, Sanjay Rajdev - Original Message - From: Sanjay Rajdev [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Cc: philipp kempgen

Re: [asterisk-users] BSNL caller ID (India)

Tzafrir, I am sure about both of them in my zapata.conf. I am on Asterisk 1.4.2 and the zapata.conf is in /etc/asterisk directory with all other asterisk configuration files Do you have any other idea which can help me finding out what is wrong. Regards, Sanjay Rajdev - Original Message

[asterisk-users] Question regrading IAX

and coming back. Regards, Sanjay Rajdev ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] queues

You can have the agent login once and newer log out. You can certainly set up your asterisk box to persit the login over the reload and the restart. persistentagents=yes Regards, Sanjay Rajdev Phone : +1 (877) 342 2329 x 1702 Fax : +1 (815) 261 5907 http://www.featherstoneinformatics.com

[asterisk-users] BSNL caller ID (India)

, Sanjay Rajdev ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] ${CALLERIDNUM} DEPRECATED in 1.4.2

${CALLERIDNUM} is DEPRECATED in 1.4.2 what is the alternative to find the same in extensions.conf for setting a proper dialplan. Please Suggest Regards, Sanjay Rajdev ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing

Re: [asterisk-users] BSNL caller ID (India)

channel = 4 Regards, Sanjay Rajdev - Original Message - From: Tzafrir Cohen [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, April 17, 2007 2:40:32 AM (GMT+0530) Asia/Calcutta Subject: Re: [asterisk-users] BSNL caller ID (India) On Mon, Apr 16, 2007 at 11:18:46PM +0530

[asterisk-users] Problem with queue

) reportholdtime=t (yes) memberdelay=0 weight=0 Does anyone have idea what is wrong. Please suggest. Regards, Sanjay Rajdev ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit

Re: [asterisk-users] Problem with queue

Regards, Sanjay Rajdev Tha i did because i dont want any call to get disconnected. Can you let me know what can be the problem doing so. - Original Message - From: Philipp Kempgen [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users

Re: [asterisk-users] Problem with queue

I waited for almost 5 minutes but still did not receive the call. Regards, Sanjay Rajdev - Original Message - From: Philipp Kempgen [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, April 17, 2007 5:36:45 AM (GMT

Re: [asterisk-users] Zap failure: cause 66 - Channel not implemented

Which version of Zaptel and Asterisk are you using. If you have compiled Asterisk 1.4.2 with Zaptel 1.4.0 or a lesser version of Zaptel, you may face this problem. Regards, Sanjay Rajdev - Original Message - From: Greg Woods [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non

[asterisk-users] LED does not glow on new Voicemail

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 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk

Re: [asterisk-users] LED does not glow on new Voicemail

on my asterisk box with SIP Firmware version 8.0.1 Any ideas Also it would be great if someone could tell me how to configure MWI from step 1, so that I can check if I am missing something. Regards, Sanjay Rajdev - Original Message - From: Matt [EMAIL PROTECTED] To: Asterisk Users

[asterisk-users] missing chan_zap.so

. Regards, Sanjay Rajdev ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] ZAP does not disconnect

I have a ZAPTEL interface card with 4 channel. If I call out through the zap channel to my mobile, the mobile starts ringing, but If I disconnect the internal phone that is my SIP client the mobile does not stop ringing. Anyone any suggestion of what am I doing wrong. Regards, Sanjay Rajdev

[asterisk-users] Require only GSM Codec

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 ___ --Bandwidth and Colocation provided by Easynews.com

Re: [asterisk-users] Problem while using asterisk Realtime

so as I have already mentioned below that the odbc connectivity is working fine. Also I have checked other option in Menuselect everywhere it says same for odbc. Can someone please let me know what I is wrong here. Regards, Sanjay Rajdev - Original Message - From: Sanjay Rajdev [EMAIL

[asterisk-users] Problem while using asterisk Realtime

(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

Re: [asterisk-users] SIP RTP Tunnel

Try setting canreinvite = no in sip.conf or the database (where you have sipuser setting). Regards, Sanjay Rajdev - Original Message - From: kalle odenthal [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, March 30, 2007 5:52:47 AM (GMT+0530) Asia/Calcutta Subject