similar to: Asterisk Realtime extensions configuraton..

Displaying 20 results from an estimated 10000 matches similar to: "Asterisk Realtime extensions configuraton.."

2005 May 13
2
Asterisk extensions from Mysql
Hello I was just stuck around as to how I configure my Asterisk to access extensions from Mysql. I have made all the necessary changes in the extconfig.conf, the extensions.conf, res_mysql.conf, res_config_odbc.conf,res_odbc.conf as they have mentioned on the site www.voip-info.org <http://www.voip-info.org/> . But still I am getting the error as May 13
2006 Jan 12
0
Configuration of SIP Mysql peers.
Hi all, I have configured Asterisk using Mysql database. The peers that I have mentioned in the database are successfully registered to Asterisk, But I am getting a warning stating "Mysql realtime: Failed to query the database" Below I am pasting error. Could anybody please let me know what's the cause for this warning.... *CLI> -- Registered SIP 'bharat'
2005 May 25
5
C files of Asterisk
Hello Everybody, I was going thru the C code of Asterisk. Does anybody know how does one go about modifying the C code of Asterisk? Please do reply. Regards, Bharat M. Sarvan EZZI BPO Pvt Ltd., PUNE. -------------- next part -------------- An HTML attachment was scrubbed... URL:
2005 May 19
1
GOTO statement in Realtime-Extensions not working like expected
Hi .. When I use the GoTo statement in realtime to goto a priority only ... E.g. Goto(3) then there's no problem But, If I try to jump to another context ... E.g. Goto(othercontext,${EXTEN},3) then it doesn't work If I process the same statement in extensions.conf things go well Are there things broken regarding GoTo in combination with Realtime Extensions ?
2005 May 19
1
Asterisk real time extensions problem...
Hello everybody, I have setup asterisk real time extensions and its working pretty well. But the problem is when I am jumping between the contexts using the Goto statement in the database. I am getting a error = Parsing '/etc/asterisk/sip_notify.conf': Found -- SIP Seeding peers from Astdb: 'ezzibpo4' at ezzibpo4@210.211.246.47:5061 for 60
2005 Mar 17
4
Hi there..
Hello Everybody, This is Bharat here. I am on the way of learning Asterisks, and I just wished to know how I go about if got to write dailplans for outbound calls and inbound calls. If you could provide me with a simple example, I could get thru. Waiting for your response Regards Bharat M. Sarvan -------------- next part
2006 Feb 08
0
Asterisk returning 403 Forbidden response
Hi all, I have configured Asterisk using database. The real time is working pretty well. That is asterisk is picking up details of peers and the extensions as well properly from the MySQL database. -- Executing Wait("SIP/bharat-f720", "2") in new stack -- Executing NoOp("SIP/bharat-f720", ""Welcome to Asterisk"") in new stack
2005 May 13
1
Help needed on setting up realtime
I installed Asterisk CVS-NHEAD-05/13/05-01:59:30 and placed few call in and through successfully. I was trying to set up the Realtime - picking the sip.conf and extensions.conf from mysql. I was going through some wiki pages, but what i don't understand is - which configuration change makes asterisk stop looking at extensions.conf and sip.conf for sip peers and pick the same from database.
2005 Aug 04
3
SIPPeersAction class file not found in the Asterisk-java.jar file
Hello Everybody, I am working on Fastagi and I am making use of Asterisk-java. But I don't find the class file for SIPPeersAction. Hence I am getting the error message when compiling my java code. ---------------------------------------------------------------------------- ------------------------------------------------ [root@localhost asterisk-java-0.1]#
2005 Sep 15
2
Help on RealTime Extensions on Oracle DB
Does someone here configured RealTime Extensions using ODBC connecting to Oracle DB? Im having a problem in dialplan patterns, it doesnt work. Pls. help! -Chris __________________________________ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com
2009 Mar 19
0
Extensions not found and 401 Unauthorized in realtime configuration (Long post)
Hi to all the ML. I'm new here. I start to use asterisk with realtime configuration, with pgsql backend connected via odbc. The connection between asterisk and pgsql works fine. I create a table sip_conf with 2 user (for testing purpose), 1401 and 1501. Those are the records: asterisk=> SELECT name,host,type,context,secret,defaultuser from sip_conf; name | host | type | context |
2007 Jan 14
0
realtime mysql db performance difference with matching extensions
Hi All, I'm testing different ways to implement a LCR/OCN tabe to shift calls to multiple carriers for better rates. I'm using realtime mysql database access, asterisk 1.2.9.1 with mysql 3.23. Scenario 1: I send outgoing calls with with a Goto statement into the context with the realtime switch to dip into the mysql lcr database, currently I have ~13K records/routes. The database
2005 May 12
0
Making Asterisk run on Mysql backend
Hello there, I have configured my asterisk to run on Mysql backend. But the Asterisk was unable to pick the peer details from the database. This is how I configured the Asterisk to run with mysql on the backend. Edit /usr/src/asterisk/channels/Makefile, change it to enable the MYSQL_FRIENDS USE_MYSQL_FRIENDS=1 USE_SIP_MYSQL_FRIENDS=1 cd /usr/src/asterisk make
2005 May 23
0
Modifying Asterisk's C files
Hello Everybody, I wished to know that the .c files in Asterisk in the /usr/src/asterisk directory, can they be modified to change the behavior of Asterisk? If yes, could you please tell me as how does one go about modifying the code and also which are the files that are modified? It would be very kind of you if could mail me a sample file with the changes made and the
2009 Dec 05
2
How to use SIP hints and BLF for realtime extensions on Aastra phones?
Hi, I need to make use of BLF feature on Aastra 6757i phones but its an Asterisk 1.4 using realtime architecture. Extensions are defined in realtime database and dial plan is in AEL. I am able to correctly setup hints in the dialplan, but they don't work. Did some research and found out that hints don't work work with realtime extensions. Is there any work around? On voip-info I read
2005 Jan 20
0
Dial plan problems with realtime extensions ...
Hi, Case1: --------- --> extensions.conf exten => 1023,1,Voicemail(101) exten => 1023/101,1,MeetMe(200) Case2: --------- -> extensions table (using realtime extensions) +----+---------+----------+--------+----------+---------+ | id | context | exten |priority| app | appdata | +----+---------+----------+--------+----------+---------+ | 29 | default | 1023 | 1 |
2005 Feb 07
0
RealTime Configuration for extensions.conf
Hi All, I've been fiddling around with the RealTime configuration. For SIP and IAX it's really cool, and the switch thing is cool too. But I've tried performing a GOTO from one RealTime context, to a second RealTime context. That didn't really work. Any idea how to make it work ? apart from simply defining contexts in /etc/asterisk/extensions.conf manually mapping the
2006 Dec 18
1
MWI, Realtime SIP, Voicemail and Extensions, UAs registered with SER
I have the following setup: - UAs registered with SER/OpenSER - SIP peers (non cached), extensions, voicemail setup (not message storage) defined in Asterisk 1.2 using Realtime When a message is left in the user's mailbox, no Notify message is sent to SER. 1. If the SIP peer is defined in sip.conf with a host=ser.domain.com then the notfy is sent to SER. 2. If realtimecache=yes is set in
2007 Feb 26
2
Ex-Girlfriend syntax and RealTime Extensions
As seen in the following URL: http://www.voip-info.org/wiki/view/Asterisk+RealTime+Extensions and as I also tested some time ago with an old release of Asterisk, RealTime Extensions didn't support the Ex-Girlfriend syntax. Is it already working in recent 1.4 or 1.2.15 releases? Is there any other way that I can use to do the same thing but only using contexts, for example? If yes, please
2008 Feb 07
2
Goto in Realtime extensions
Hello, I'm having troubles while using the "Goto" function in a realtime extension. Here is the error message : -- Executing Goto("SIP/siemens1-081f56b0", "script_13_0|s|1") -- Goto (script_13_0,s,1) [Feb 7 13:24:21] WARNING[28666]: pbx.c:2455 __ast_pbx_run: Channel 'SIP/siemens1-081f56b0' sent into invalid extension 's' in context