Re: [asterisk-users] asterisk-users Digest, Vol 72, Issue 42
thanks for your replay, but i am not able to set this fecility in agent phone. any other solution ? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Queue
hi, all Is ther any way to set 3-way conference using queue app other other way using queue app. scenario: custmore call to queue , agent answered than agent transfer to third persion, so the three call communicate with each other. Regards, -- Bhrugu Mehta Sr. S/W Engineer (DD) VOIP,Telephony Team India -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] call-waiting
hi, all Is ther any way to set up call-waiting feature in asterisk using dialplan or any other ways. I want to use only asterisk for that not any other gui. I am using asterisk 1.4.28. Regards, -- Bhrugu Mehta Sr. S/W Engineer (DD) VOIP,Telephony Team India -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] get hold event
Hi, all how to get hold event in asterisk. is it possible, when user1 put on hold in queue moh1 file played. when call transfer to agent and answered agent put hold at that time moh2 file played ? I have used asterisk 1.4 version. Regards, -- Bhrugu Mehta Sr. S/W Engineer (DD) VOIP,Telephony Team India -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sip send image
Thnks for ur reply, SendImage() doesn't work with asterisk sip channel. any other solution? Regards, -- Bhrugu Mehta Sr. S/W Engineer (DD) VOIP,Telephony Team India -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sip send image
hi, all is there any way to send image on sip channel ? Regards, -- Bhrugu Mehta Sr. S/W Engineer (DD) VOIP,Telephony Team India -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] astdb
Hi, all What is the use of astdb? Is it used to store realtime values like sip etc. Regards, Bhrugu Mehta -- _ -- 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
Re: [asterisk-users] astdb
hi, all thanks for reply, but actually i have configured sip to realtime and i got this message SIP Seeding peer from *astdb*: 'sip_ext' at sip_...@asterisk_ip:5060 for 60 so i have to know that my sip ext is stored in astdb or not. any other suggetion ? Regards, On Wed, Jan 27, 2010 at 4:37 PM, bhrugu mehta mehtabhr...@gmail.com wrote: Hi, all What is the use of astdb? Is it used to store realtime values like sip etc. Regards, Bhrugu Mehta -- Bhrugu Mehta Sr. S/W Engineer (DD) VOIP,Telephony Team India -- _ -- 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] queue
Hi, all Is ther any way to pass channel queue such a way Queue(SIP/1001SIP/1002SIP/1003) thanks, Bhrugu Mehta -- _ -- 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
Re: [asterisk-users] PBX deployment big problems: Voip traffic analysis
hi, Yes, there are many problem to implement and setup asterisk in a callcenter. but , all these problem can be remove if you set up your hardware and your LAN network verywell. Generaly, your server Configuration should be greater and your LAN also. You have to use Proper Codecs for voice. Generaly , g729 is greater. regards, Bhrugu Mehta On 5/16/08, gincantalupo [EMAIL PROTECTED] wrote: Hi, hope not to be OT :) after more than 3 years of PBX installations we can adfirm Asterisk is stable enough to be considered a good product but still we encounter a lot of problems when deploying a new PBX. It seems that the biggest problems are all networking related: one way voice (also inside a LAN), calls drops, etc... How do you face this kind of problems? Which diagnose tools/methods do you use? Thank you. Giorgio Incantalupo ___ -- 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 ___ -- 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
Re: [asterisk-users] Asterisk for Larg
hi, I have not tested that but I have seen 100 agents configure with asterisk. thnks Bhrugu mehta On 5/15/08, gmail [EMAIL PROTECTED] wrote: Is Asterisk practically stable and reliable for a larg Enterprise has say a 1 phones , is there any case study like this? ___ -- 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 ___ -- 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] queue
hi, all is there any way in queue app. to execute asterisk app. after Queue() app. i.e [myplan] exten = _X.,1,Answer exten = _X.,n,Queue(myqueue) exten = _X.,n,Background(file-to-play) exten = 1,1,Playback(thnks) exten = 2,2,Playback(by) Is these possible above situation , how thnks, Bhrugu Mehta ___ -- 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] callerid problem
hi, all i am using zma800p card( from zapmicro). i create small ivrs. when i call on fxs channel calls lended and ivrs start on that channel. but when i use callerid app. from asterisk , doesn't displayed any callerid on asterisk. any suggestion. thanks in advance. Bhrugu mehta ___ -- 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] fxo tdm400p issue
hi, all I have configure tdm400p analog fxo card. that's ok. but how to chek that is working properly or not. i chek with ztcfg - and zttool . that's ok. i want to dial from my fxo port to another extesion. zaptel.conf -- fxsls=1,2,3,4 defaultzone=in loadzone=in zapata.conf context=mycontext signalling=fxl_ls group=1 channel=1-4 thanks' in advance. Bhrugu mehta ___ -- 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] chanspy doesn't work properly
HI, all I have tested chanspy app. to monitoring agent and customore conversation. if customer and agent are already in conversation , using spy we can'nt here anything on that extension(agent extension). if next time calls come to that chan we listen that conversation. any idea? thnks, in advance Bhrugu Mehta ___ -- 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
Re: [asterisk-users] Communication between two asterisk server
hi,preeta you have to change sip.conf in both server. suppose, server 1 and server 2 both are asterisk server. you want to call from server 1 to server 2. then, in ser-1, sip.conf [general] register= user:[EMAIL PROTECTED] [user] type=friend fromuser=user username=user secret=pass host=ipofserver2 context=any in server2, sip.conf [user] type=friend username=user secret=user host=dynamic context=anyyouwant Bhrugu Mehta (SAI INFO SYSTEM LTD.) On 2/15/08, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi, I want that an sjphone registered using serverA can call to an sjphone registered using serverB and vice vers. I want to know how two asterisk server communicate to each other. Please let me know, for that, what configuration file I have to change. Thanking you, Regards, Preeta Pandey The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com ___ -- 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] two zaptel card
hi, all I want to use two zaptel card(TE210p) in pc for asterisk. Is there any special requirement for this configuratin. any suggestion. thanks , Bhrugu Mehta ___ -- 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
Re: [asterisk-users] Asterisk crashed..
hi, I have used asterisk 1.2.12.1 and using linux 4 enterprise edition. Bhrugu Mehta On Jan 22, 2008 11:33 AM, ram [EMAIL PROTECTED] wrote: On Jan 22, 2008 9:36 AM, Bhrugu Mehta [EMAIL PROTECTED] wrote: hi, all I set up asterisk with 5 to 6 agent . in these all are going well. but when i increase agent 12 to 13 asterisk crashed. Any suggetion. thnks Bhrugu mehta Hi what version of asterisk what is the hardware config and OS ram ___ -- 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 ___ -- 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 ___ -- 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] Asterisk crashed..
hi, all I set up asterisk with 5 to 6 agent . in these all are going well. but when i increase agent 12 to 13 asterisk crashed. Any suggetion. thnks Bhrugu mehta ___ -- 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] Developing Help
hi, all, can anybody tell me how to be a part of asterisk developer team. I am so much intersted. thnks in advance. Bhrugu Mehta ___ -- 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] zaptel digit problem
hi, all I am using asterisk 1.2.12.1 and zaptel 1.2.7 and libpri 1.2.1 version. I have created Ivrs(very big) .It works fine in sip phone , but when i call through zaptel digit sens problem occured. Asterisk doesn't sens any digit pressed.Our pstn is CORAL pbx. any suggesion.. thnks, Bhrugu Mehta(india, gujarat) ___ -- 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] zaptel programming
hi, all I am new to zaptel programming. can anybody help me how to start this. or any ref. site or matirial availabel. i want to use c lang. for this. thnks in advance. Bhrugu Mehta ___ --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] ASTERISK cd-rom
hi, all i want to create cd-rom with asterisk. how it possible. when i put disk in cdrom it boot automatifcally and auto-start installation like TRIXBOX. any idea. thnks, Bhrugu Mehta ___ --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
Re: [asterisk-users] auto dial and IVR
hi, easy below i have done, // 1.call file Channel: SIP/your_exten MaxRetries: 2 RetryTime: 60 WaitTime: 30 Context: ivrs_context_to_play Extension: your_exten Priority: 1 then move this file to your outgoing dir to generate outgoing call. asterisk auto handle all things of ivrs which you have created. enjoy Bhrugu Mehta On Jan 2, 2008 3:59 PM, Rilawich Ango [EMAIL PROTECTED] wrote: Hi, Is it possible to let asterisk auto dial out and play the IVR? How? i.e. -asterisk auto dial out (use outgoing folder?) -user pick the call -play IVR (1-for English, 2-for Chinese) -Then user can press the number to go through the level of IVR. ___ --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 ___ --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] app_echo.c
hi, all I have test echo application for just fun. I can'nt understand why this is used below in .c file, format = ast_best_codec(chan-nativeformats); ast_set_write_format(chan, format); ast_set_read_format(chan, format); without this this application work fine. then why this is used. any suggestion?? Bhrugu mehta ___ --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
Re: [asterisk-users] Directories Used by Asterisk
hi, first of chek you have permission on appropriate folder. /etc/asterisk: directory not created automatically. type command in asterisk source /usr/src/asterisk dir prompt # make samples this creates .conf files in /etc/asterisk dir. enjoy!! Bhrugu mehta On Dec 29, 2007 2:49 PM, broadband Voice [EMAIL PROTECTED] wrote: I successfully obtained the Asterisk code and extracted them into /usr/src. When I make and install asterisk, zaptel, libpri etc. Are they supposed to move automatically into their respective directories? I cannot find: /etc/asterisk/ /usr/lib/asterisk/modules/ /var/lib/asterisk Do I have to manually create them or this is failed install? Thanks. ___ --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 ___ --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
Re: [asterisk-users] application not load
hi, thnks 4 reply, actully i am using asterisk 1.4.15 and that is defined in menuselect file.(xml file) so no need to add entry in module.conf Bhrugu mehta On Dec 27, 2007 7:37 PM, dave cantera [EMAIL PROTECTED] wrote: bhrugu, did you try and load it manually? Modules are compiled in to shared object (.so) files. They are installed to /usr/lib/asterisk/modules and can be turned on and off from loading by editing /etc/asterisk/modules.conf. Modules must include asterisk/modules.h. Modules must also export several functions. The following functions generally return 0 on success and non-zero on failure. Do not define any of these functions as static. http://www.lobstertech.com/doc/ast-12-func/#funcmod daveC Bhrugu Mehta wrote: hi, all I creat new application app_myapp.c for asterisk 1.4.15. I add this in asterisk/apps dir. to load. after compiling asterisk app_myapp.o and app_myapp.so has been created but when i run show applications at cli . my application not displayed. what's wrong??? any suggestion!!! thanks Bhrugu Mehta ___ --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 -- My wife's sister is in California. I should buy her a Videophone2008! Truly, The Next Best Thing to Being There! -- WorldWideVideoPhones.com 856.380.0894 ___ --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 ___ --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] application not load
hi, all I creat new application app_myapp.c for asterisk 1.4.15. I add this in asterisk/apps dir. to load. after compiling asterisk app_myapp.o and app_myapp.so has been created but when i run show applications at cli . my application not displayed. what's wrong??? any suggestion!!! thanks Bhrugu Mehta ___ --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] zap transfer
hi, all I want to transfer my zap incoming call to another hard phone. is there any way to transfer call. our company is using CORAL EPBX. thnks for any suggestion Bhrugu Mehta ___ --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] autoservice.c
hi, all actually i can't understand what is the use of autoservice.c file. can anybody help me. thnks in advance. Bhrugu mehta ___ --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
Re: [asterisk-users] Soundcard necessary on an asterisk server to get output of playback()??
no , not at all, there is no need to install sound card in asteirsk system. I am using asterisk server without soundcard. so there may be antoher problem may in configurtion of zapata or other. cheers!!! Bhrugu mehta On Dec 3, 2007 11:31 PM, Stefan Guenther [EMAIL PROTECTED] wrote: Hi, I' still fighting the problem, that I can talk from one SIP phone to another, but I can't hear the output of the playback or similar applications: exten = 202,1,ANSWER() exten = 202,2,PLAYBACK(tt-monkeys) exten = 202,3,HANGUP() When I dial 202, asterisk show the following on the cli: -- Executing [EMAIL PROTECTED]:1] Answer(SIP/user1-0827ebe8, ) in new stack -- Executing [EMAIL PROTECTED]:2] Playback(SIP/user1-0827ebe8, tt-monkeys) in new stack -- SIP/user1-0827ebe8 Playing 'tt-monkeys' (language 'de') Yes, the file tt-monkeys exist in /var/lib/asterisk/sounds and the subdirectory de. No, there is no error message even if turn on debugging. :-( Besides this strange behaviour, I was wondering whether the asterisk server needs an soundcard to send the output of e.g. the playback application to the phone. BTW, this is asterisk 1.4.13 I would be really happy, if someone has an idea how to solve this problem. Thanks in advance, Stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Geschaeftsfuehrer Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice-over-IP-Loesungen ___ --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 ___ --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] call-limit in database
hi, all proble: I have add CALL-LIMIT field in my sip table in mysql. but when i call using sip same error occurred when use simple sip.conf file. is this possible to add CALL-LIMIT field in sip realtime table in mysql. if yes than how Bhrugu Mehta ___ --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
Re: [asterisk-users] GUI for Asterisk: Call Flow
hi, ya, there is one s/w whiche is freely available for linux os as * events.tar * . it is in php. you can use this. regards, Bhrugu Mehta ___ --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] DeadAgi
hi, all I am new to use DeadAgi, can anybody help me how to use DeadAgi, actually i want this, when caller hangup his/her phone, i want to send packet to my other app that check caller hung up done. ___ --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] Oracle and asterisk
hi, all I want to connect asterisk with oracle database. how to start this , that's i dont know . any pls help me thnks in advance Bhrugu mehta ___ --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
Re: [asterisk-users] Oracle and asterisk
thnsk for giving me reply, Bhrugu mehta On Dec 3, 2007 12:41 PM, Tilghman Lesher [EMAIL PROTECTED] wrote: On Monday 03 December 2007 00:48:55 Bhrugu Mehta wrote: I want to connect asterisk with oracle database. You'll need to install the Oracle ODBC driver for Linux. One word of warning, though: the ODBC driver linked against the InstantClient library has a very nasty resource leak in the library itself. Specifically, on every connection, it leaks 2 file descriptors and fails to close cursors properly on each statement executed. Therefore, be sure that you're linking to the other Oracle client library, not the InstantClient. http://home.fnal.gov/~dbox/oracle/odbc/ -- Tilghman ___ --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 ___ --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
Re: [asterisk-users] MSSQL ODBC Connections
hi, thnks for reply I have already upgrade my odbc connector , but same error come. Bhrugu Mehta ___ --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
Re: [asterisk-users] MSSQL ODBC Connections
hi, I want to create connection using odbc for mysql i have used cdr_odbc module for that. but when asterisk insert record to my mysql database arise segfault error. any suggetion, pls give me tnks Bhrugu Mehta ___ --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] Queue
Hi all I want to create Ivrs using dialplan and aslo want to transfer call to agent using Queue app in asterisk. Is there any way to get IP ADDRESS of free agent which is found by asterisk thnks , Bhrugu mehta ___ --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
Re: [asterisk-users] Need Reference sites
Hi, Various site available for asterisk,listed below, www.asterisk.org www.voip-info.com www.digium.com and best is search in www.google.com On Nov 5, 2007 5:22 AM, Michael Davidson [EMAIL PROTECTED] wrote: Hi, I'am comparative newbie to the world of Asterisk. I'd like to introduce an Asterisk based PBX into my company but need to convince my executive of it's worthiness. I need some reference sites to quote in my discussion, preferably well known companies of course. I have surfed the net but not come up with anything of note, if anyone can help it would be greatly appreciated. Thanks, Mike D. ___ --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 ___ --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] Autodialing
hi,all I want make Autodialer in c++ using Asterisk Mangager Interfase; how to syncronize originate action i.e. at a time one call made and this time asterisk wait for some second to generate new call. thnks in advance. Bhrugu Mehta (SAI INFO SYSTEM) ___ --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] DTMF DIGIT PROBLEM
hi, all I have problem to sense digit in my ivrs. scenario is below: I am using zaptel T410P digium card to competible with my PSTN(CORAL) [ivrs] exten = s,1,Background(welcome-ivrs) exten = 1,1,Playback(welcome) exten = 2,1,Playback(goodby) sound file are .wav files. when i dial no. from analog phone to launch ivrs welcome-ivrs.wav file plays and when i press digit 1 play wecome and 2 play goodby. but some time asterisk server doesn't sense which digit i pressed so welcome-ivrs file continue with playing. Doesn't stop playing thnks, regard Bhrugu Mehta(SIS) ___ --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
Re: [asterisk-users] Asterisk cli
Hi, 1. If you are connecting to remotly with asterisk server you have to use asterisk -vvvrc 2. if your asterisk server is your pc then you have to use asterisk -c ok enjoy Bhrugu mehta ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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
Re: [asterisk-users] Looking for Asterisk Consultant in San Franicsco
HI, I have read your mail. I get ready for that but pls tell me what i do at remote support. thnks for sending mail. Bhrugu mehta ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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
Re: [asterisk-users] How to make call from asterisk?
To make call to X-lite or any sip phone , 1. create extension in sip.conf for soft phone. 2. register sip phone with that exentension which is in sip.conf -give your asterisk server ip in softphone and also username and password. 3. wait for some time 4. if all are good then your phone has been registered. ___ --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
