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Please could someone advise good manual for using lua for asterisk dialplan. There is not much docu about it. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110218/42642787/attachment.htm>

Hi , I know it's might not the right way to asking such stupid question. But I want to take help from experts into VoIP fields so I have to decided to post here. Please help me which will be the best VoIP conferencing phone which will cover 10 Persians into conferencing with best audio support ? -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -------------- next

Stop advertising. Le 26 sept. 2010 09:46, "Faisal Hanif" <faisal at vopium.com> a ?crit : Hi Abdul-Basit, If you need only different intervals of billing you can easily do it using any AGI as we are doing it in Perl AGIs using post call billing. But if you need realtime billing then the most stable and flexible option is to use FastAGI+ AM...

Hi, I have 8 E1 PRI Lines and i have 200 phone numbers and 200 channels (25 channels per PRI). is there a utility available in Asterisk to dial out 200 numbers and run a campaign for 200 numbers concurrently and play a mp3 file ? Please suggest/guide Regards Kaushal

...e using Asterisk 1.6.2 and it is continually failing to resolve Verizon SRV and sending following message, WARNING[23042]: acl.c:392 ast_get_ip_or_srv: Unable to lookup 'whsvoip.globalipcom.com' DNS settings on OS level is working fine. Can anyone have an idea about it? Regards, Faisal Hanif

Hi Everyone Asterisk 1.4.33 is running with Sangoma/Dahdi for analogue lines to Bell Canada. User claims that call hangup without any interferance to the phone set. Is there ANYWAY to find out which party hang-up the call or if the call was cut-off due to other reasons? I checked the *"asteriskcdrb"* table and it's pretty much useless in this case as it only logs the duration and

Hi, The subject says it all. Is it possible to put the IP address of the peer in the CDR records? Using AGI maybe? -- Kind regards, Signet bv Remco Bressers T 040 - 707 4 907 F 040 - 707 4 909 E rbressers at signet.nl

Hello everyone. I need a quick help on how to capture who answered the call with agi. Here is an example: -- Zap/32-1 is ringing -- Zap/33-1 is ringing -- Zap/34-1 is ringing -- Zap/35-1 is ringing -- SIP/operator1-e77f answered Zap/23-1 So how can I capture this value and write it to mysql? I already have this:

I need graph the utilization of my t1s. Does anyone know of a plug-in, code, or web interface I can use to help do this. I am currently using Asterisk 1.4 Tony -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100726/c1463f31/attachment.htm

Hi All, I have a quick question with regards the pbx_lua module. Would the lua dialplan have direct access to the odbc configuration within Asterisk, those odbc connections/dsn's that are defined in res_odbc.conf/extconfig.conf/cdr.conf? Thanks Bruce

Hello, I was wondering: What does Dial(Local/...) offer that a Goto() doesn't? For instance: ======== ;exten => h,n,Goto(callback,start) exten => h,n,Dial(Local/start at callback) [callback] exten => start,1,Verbose(In callback) ======== Thank you.

Hi all, Trying to find where i got wrong in my config.... Is the "realm" parameter in sip.conf only used for possible autentication? The thing is, i got my box more-or-less working as i wanted, but i can only reach internal functions (like echo-test and so on) and other sip-clients if i dial "1234 at fqdn", while i was expected to be able to just dial "1234" I

...t;= 2'. You can think of drawing 'trunc(size*prob)' observations deterministically, then drawing the remaining 'size - sum(trunc(size*prob))' observations without replacement, with an adjusted prob vector. The algorithm I used is relatively simple. It is one of the Brewer and Hanif algorithms (though I don't recall if they used the final random shuffle). Here's one description, or you may prefer the description in Pedro J. Saavedra (2005) Comparison of Two Weighting Schemes for Sampling with Minimal Replacement http://www.amstat.org/Sections/Srms/Proceedings/y2005/F...

Anyone here who has configured zaptel/dahdi for a singtel E1 line? What are the settings for coding, framing, line type and switchtype? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110210/4288ed84/attachment.htm>

How should I configure my asterisk server so that I can receive calls from an unregistered peer from whom I also receive registrations of sip phones? I'm asking you this, because with my actual configuration, when I register a contact from that peer's IP, no more inbound calls are accepted from that peer, as my asterisk rejects those INVITEs with "407 Proxy Authentication

Hello guys, I need to be able to throw cdrs on more than one servers at a time. Please let me know how this can be done. Thanks

Hello, trying to load ael module in asterisk ver 1.6.2 got the following: asterisk*CLI> module load pbx_ael.so Unable to load module pbx_ael.so Command 'module load pbx_ael.so' failed. [Feb 18 11:25:47] WARNING[7412]: loader.c:449 load_dynamic_module: Error loading module 'pbx_ael.so': /usr/lib/asterisk/modules/pbx_ael.so: undefined symbol: ast_compile_ael2 [Feb 18 11:25:47]

Hello, I wish to make a simple system to limit peers at x minutes depending of buyer voip packet. Can someone help me with some directions? I intend to make a separate dial plan and every calls to be in cdr table in mysql. Is any chance to make some scripts to drop calls after peer used all minutes? I use asterisk 1.4.34 + mysql + cdr + asterisk-gui administration interface. I don't really

Hi guys, is there a way to see how many channels of an specific tecnology are being used? Like, i have a zap card, e1 (30 channels), and there are 10 channels being used at this moment. When the E1 reaches 15 busy channels I need to receive a call or something like this, telling me that 15 of 30 channels are busy. How can I do this? Thanks! -------------- next part -------------- An HTML

Hello, Are there any gateways which allow me to hook a cellphone to Asterisk and use that line for routing my calls? Basically, I'm looking to play around a bit and if I can get to connect a cellphone with Asterisk then that would be great. Thanks, Hitesh PS: I have tried to search on the web, but didn't find any pointers on how to do so. -------------- next part -------------- An HTML