similar to: Help with IAX

(missed one thing) I have made some changed to your config: extensions.conf from 192.168.253.21: ; ; Create an extension, 205, for trunk ; exten => 205,1,Dial(IAX2/ <mailto:IAX2/192.168.253.20/205@iax-trunk> tecinfo1/205) iax.conf from 192.168.253.21: [tecinfo1] type=peer username=tecinfo2 secret=secret host=192.168.253.20

I am new at this. I have read "Asterisk: The Future of Telephony" and have installed AsteriskNOW (beta 4, due to the dual processor problem in beta 5). The GUI interface does not seem to provide the capability that I need, although I have modified the *.conf files to successfully create what I need. Given this, I would like to install Asterisk on a distro. I am most familiar with

In looking at the safe_asterisk script, it would appear that it is encouraging the running of the Asterisk application as root user. My natural inclination is to run it as a non-privileged user. What is recommendation? +++++++++++++ This e-mail was checked by the TecInfo Content Scanning Service for potentially harmful content, such as viruses or Spam For more information, call 800.863.5415

Thanks to all that have helped me so far. I have made a lot of progress. I am able to make prilib and zaptel. Now to Asterisk... After installing the kernel source, I have: # cd /usr/src/linux # make cloneconfig # make prepare-all Then I have run ./configure in the asterisk-1.4.4 directory. I have: # make clean # make Which goes through a number of compiles and then ends

From: Malcom Kemp Sent: Wednesday, April 25, 2007 11:19 AM To: 'asterisk-users@lists.digium.com' Subject: AsteriskNOW generation of zapata.conf file I am a new user of Asterisk, and an trying to use AsteriskNOW. The test system is dual processor, so I am using the beta 4 version. I am currently trying to manually configure, as the GUI does not seem to let me accomplish what I need.

I am installing asterisk on a second box with OpenSuSE 10.2. I have installed libpri, run menuselect/configure and then make. The make stops at the last line shown below. Looking at the processes, the current process is running sed. Not sure from where. Any ideas? ... checking for ar... /usr/bin/ar checking for cp... /bin/cp checking for ln... /bin/ln checking for mkdir... /bin/mkdir

What is the difference between the 1.2.x and the 1.4.x branches? I am starting to deploy asterisk from scratch. I expect that I should use the 1.4.x branch, but don't know the rationale for the two. +++++++++++++ This e-mail was checked by the TecInfo Content Scanning Service for potentially harmful content, such as viruses or Spam For more information, call 800.863.5415 or visit

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

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

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.

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

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

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

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 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 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.

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