similar to: Struggling with Extensions in Realtime

Hello Everyone, I have gone through a few really good tutorials from the OpenSIPS site, Asterisk resources etc.. The unanswered question (and final piece of our puzzle) is if it's possible to have a register free environment in an OpenSIPS/Asterisk integration. Most approaches have OpenSIPS relay the UA's REGISTER request to Asterisk which has "host=dynamic" set for the

hi, all i try to confiture realtime queue, but not work, details as below: Insert into queue_table(name)value('95040654321'); INSERT INTO queue_member_table VALUES ('', 'Zhang Shukun', '95040654321', 'SIP/1001', 2, 1); INSERT INTO queue_member_table VALUES ('', 'Li Aiwei', '95040654321', 'SIP/1002', 2, 1); INSERT INTO

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

I am struggling with how to use the date extension for Dovecot/Pigenhole implementation of Sieve email filtering language. I have read RFC 5260 several times and tried different things with no luck. I would like to place all messages this year up till the end of March into the specified mail folder. What the snippet below actually seems to do is take any messages this year (whether before or after

Hello Everyone, We are getting some rather poor results (relative) with our Asterisk setup. Not sure if we are using the sipp correctly etc.. but nevertheless, is there any documentation that describes how we can get the most our of our Asterisk box. For example when we hit the "too many file" error, and fixing it using ulimit..... Also, is there any way we can allocate sufficient

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

I've gotten canreinvite=yes to work with a sip device behind NAT!! You *MUST* port forward the SIPPort to in your gateway router to your phone. This is a MUST. Okay, now on to my problem.. I have people who will be using ulaw, and I have people who will be using g729.. I want to set it up so that canreinivte will work.. I have a single cisco gateway.. Asterisks isn't handling the

Okay, now on to my problem.. I have people who will be using ulaw, and I have people who will be using g729.. I want to set it up so that canreinivte will work.. I have a single cisco gateway.. Asterisks isn't handling the negotation between the 2 devices very well.. For example.. [gateway] type=friend host=1.2.3.4 canreinvite=yes qualify=200 dtmfmode=rfc2833 context=default disallow=all

Hello Everyone, We are currently experiencing some higher load on our servers, and since signaling comes into our servers on G729, we would like to implement G729 pass-through. A few questions arise, do we need to convert all the recording to the codec, and what about voicemail? We are also using A2Billing (hope I am not violating any thread rules), and would like to convert all that recording

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 ?

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

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

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

Hi I am having a problem with extension matching in asterisk (I am using asterisk 1.6.0.6). Is there a difference between extensions matching in realtime architecture and extensions matching in extensions.conf file. For example when I have these two extensions: -- _0699[134]XXXXX -- _06[069]XXXXXXX that are in the database and number 0699123123 comes in asterisk will always choose exten

I cannot seem to find any reference to labels in realtime extensions - using 1.4. I've googled until my eyes have bled, and also scoured voip-info.org. Is there anything that helps me here ? Many thanks. Julian

Hello, Is it possible to use the switch => statement in extensions.conf (http://www.voip-info.org/wiki/view/Asterisk+RealTime+Extensions) to point to a database and in the database use the include-statement ? In extconfig.conf I would have : extensions => mysql,asterisk,extensions_table In extensions.conf I would then have : [includecontext] switch => Realtime/includecontext at

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 |

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

Is it possible to use the "include" command to include other contexts if you are using realtime for extensions? I've searched voip-info and some people were asking about it, but I didn't find a real answer anywhere.

Hello everybody, Does anybody know how do we configure the asterisk realtime extensions using Mysql or ODBC. I have gone through the information given on the site www.voi-info.org <http://www.voi-info.org/> , but didn't get thru. Would you please tell me as to how do I go about configuring the asterisk realtime extensions. Thanks in