[asterisk-users] Different box for SIP and RTP
Hello, Is there way I can use two Asterisk box, one to maintain SIP packets and other for RTP traffic? Thanks, Mohammad -- _ -- 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
Re: [asterisk-users] Different box for SIP and RTP
Can't that third-party be an asterisk box? After hand off RTP processing, does the first box (who, hand off) still in charge of SIP packets? On Mon, May 16, 2011 at 9:13 AM, Alex Balashov abalas...@evaristesys.comwrote: On 05/16/2011 09:00 AM, Mohammad Khan wrote: Is there way I can use two Asterisk box, one to maintain SIP packets and other for RTP traffic? No, the signaling and bearer plane are integrated in Asterisk. But you can use reinvites to hand off RTP processing to third-party endpoints and bypass Asterisk, in qualifying call scenarios and network topologies. -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- 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 -- _ -- 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
Re: [asterisk-users] Best Scripting Language
I am using Ruby, per call I have 2-4 agi scripts that execute. Each take 0.02 to 0.08sec On Mon, Apr 4, 2011 at 3:19 AM, Thorsten Göllner t...@ovm-group.com wrote: Am 01.04.2011 14:27, schrieb Roger Burton West: On Fri, Apr 01, 2011 at 05:27:20PM +0530, Gopalakrishnan A.N wrote: Can anyone suggest which is the best scripting language for Asterisk or any telecom device? Depends on the other parameters. Perl is great for rapid development, but I wouldn't run it per-call on a box taking hundreds of calls per second. (Ditto Ruby and Python.) C will be much faster, but it's more effort to write and debug. Another solution could be a combination of PHP and HipHop. Easy to develop and after transaltion with HipHop very performant. -- _ -- 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 -- _ -- 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
Re: [asterisk-users] pbx.c: We were unable to say the number
Finally, I have uncovered the mystery!
Channel terminated just at the time when it was executing SayNumber, ended
up the warning in my log!
Thanks everybody for your time on this thread.
Mohammad
On Sun, Mar 27, 2011 at 5:32 PM, Sherwood McGowan
sherwood.mcgo...@gmail.com wrote:
Oh crap, you're right, my bad. Yes, I also agree, it's most probably the
language and/or missing files
On Sun, Mar 27, 2011 at 4:30 PM, Jeff LaCoursiere j...@sunfone.comwrote:
On Sun, 2011-03-27 at 16:14 -0500, Sherwood McGowan wrote:
On Sun, Mar 27, 2011 at 2:50 PM, Mohammad Khan beepl...@gmail.com
wrote:
Here is the dialplan in macro:
exten = s,n,SayNumber($[${ARG1} % 100])
when 662 was passed as ARG1, I had the following at log:
WARNING[15217] pbx.c: We were unable to say the number 62, is
it too large?
Do you see any odd in my dialplan?
662 % 100 = 66.2, not 62. It seems to me that there's more going on
here..Maybe Asterisk is being confused by actually getting 66.2? I'm
not readily able to look into the source, but I think that Asterisk
(or at least, SayNumber) cannot handle a number with a decimal point,
but please don't take that as gospel.
'%' is 'modulus', and 62 is the correct result. I am betting it is the
language setting, and missing audio files.
j
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Re: [asterisk-users] pbx.c: We were unable to say the number
Here is the dialplan in macro:
exten = s,n,SayNumber($[${ARG1} % 100])
when 662 was passed as ARG1, I had the following at log:
WARNING[15217] pbx.c: We were unable to say the number 62, is it too large?
Do you see any odd in my dialplan?
On Sat, Mar 26, 2011 at 2:44 PM, Sherwood McGowan
sherwood.mcgo...@gmail.com wrote:
Again, the relevant dialplan code is important. It is quite possible that
there's an issue with the dialplan code that you (as the person who's
dealing with the issue) may have missed. It happens all the time.
On Sat, Mar 26, 2011 at 1:25 PM, Mohammad Khan beepl...@gmail.com wrote:
I am using asterisk 1.4.38
I am getting this warning occasionally when executing SayNumber in a macro
with argument which is less than 100.
On Sat, Mar 26, 2011 at 11:03 AM, Sherwood McGowan
sherwood.mcgo...@gmail.com wrote:
On Fri, Mar 25, 2011 at 11:15 PM, Mohammad Khan beepl...@gmail.comwrote:
Hello,
Occasionally, I get the following warning in my asterisk log,
pbx.c: We were unable to say the number [n], is it too large?
n is two or one digit number, which doesn't look like large to me!
Could anybody please tell more about this warning, like in what scenario
I may have this warning.
Thanks,
Mohammad
Please post the relevant context that is being executed, that'll give us
not only the actual application, but more info as to how it's being passed.
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Re: [asterisk-users] pbx.c: We were unable to say the number
I am using asterisk 1.4.38 I am getting this warning occasionally when executing SayNumber in a macro with argument which is less than 100. On Sat, Mar 26, 2011 at 11:03 AM, Sherwood McGowan sherwood.mcgo...@gmail.com wrote: On Fri, Mar 25, 2011 at 11:15 PM, Mohammad Khan beepl...@gmail.comwrote: Hello, Occasionally, I get the following warning in my asterisk log, pbx.c: We were unable to say the number [n], is it too large? n is two or one digit number, which doesn't look like large to me! Could anybody please tell more about this warning, like in what scenario I may have this warning. Thanks, Mohammad Please post the relevant context that is being executed, that'll give us not only the actual application, but more info as to how it's being passed. -- _ -- 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] pbx.c: We were unable to say the number
Hello, Occasionally, I get the following warning in my asterisk log, pbx.c: We were unable to say the number [n], is it too large? n is two or one digit number, which doesn't look like large to me! Could anybody please tell more about this warning, like in what scenario I may have this warning. Thanks, Mohammad -- _ -- 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
Re: [asterisk-users] Play one audio file to the called part before the Dial() command
*A(**x**)*: Play an announcement (*x*.gsm) to the called party. 2011/2/16 Faisal Hanif fai...@vopium.com You can do it using callback files or AMI. *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Songtao Yu *Sent:* Wednesday, February 16, 2011 6:41 PM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] Play one audio file to the called part before the Dial() command Hi, I am not sure if it is doable: 1. We originate one call from Asterisk 2. Asterisk plays one audio file to the called part when the called part picks up the phone. 3. Asterisk establish one real connection between the caller part and the called part. Thanks, Songtao Yu -- _ -- 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 -- _ -- 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] ruby-agi-1.1.2 released
Release notes of ruby-agi-1.1.2 March 07, 2006 In this release bug # 3722 has been fixed Details of this can be found at http://rubyforge.org/tracker/index.php?func=detailaid=3733group_id=883atid=3477 http://rubyforge.org/tracker/index.php?func=detailaid=3733group_id=883atid=3477 Feedback, suggestion, feature request, bug report is always appreciated. For more information, please visit projects homepage: http://rubyforge.org/projects/ruby-agi/ To install ruby-agi, % gem install ruby-agi and to update exiting ruby-agi % gem update ruby-agi Thanks, Mohammad Khan info AT beeplove DOT com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ruby-agi-1.1.1 released !
Release notes of ruby-agi-1.1.1 February 09, 2006 This is a bug fix release of ruby-agi. Below two bugs have fixed in this release. -- ReturnStatus#timeout? was not functional, which has fixed. -- AsteriskVariable#init_caller_variable updated to fix callerid bug which was returning 'nil' for number only callerid. method init_caller_variable is a private method that manage callerid, calleridname and calleridnumber Feedback, suggestion, feature request, bug report is always appreciated. Thanks, Mohammad Khan info AT beeplove DOT com February 09, 2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ruby-agi 1.1.0 released
I just released ruby-agi-1.1.0 Here is the release notes: addition of method 'jump_to' jump_to would take three arguments ex. jump_to(context, extension, priority) enhanced callerid, calleridname and calleridnumber. calleridnumber is an addtional method to this release. Regardless of Asterisk version above three caller methods would return uniform value. Such as, callerid would return John Smith 1234567890 or empty string, if unidentified calleridnumber would return number part of callerid (as string) ex. 1234567890 or empty string, if unidentified calleridname would return name part of callerid ex. John Smith or emtpy string, if unidentified To install ruby-agi via gem % gem install ruby-agi or, to update % gem update ruby-agi Please feel free to submit bug report, feature request at info AT beeplove DOT com Your feedback is always welcome and appreciated. Thanks, Mohammad Khan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ruby-agi 0.0.2 released
Hello, I have released Ruby Asterisk Gateway Interface (ruby-agi) v0.0.2b. Any feedback, bug report, suggession, feature request is most welcome. ruby-agi homepage: http://www.rubyforge.org/projects/ruby-agi/ Download ruby-agi v0.0.2b here: http://rubyforge.org/frs/download.php/5965/ruby-agi_v0.0.2b.tgz Thanks, Mohammad Khan ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] first beta of ruby-agi is out !
Hi, I just released first beta version of ruby-agi (ruby-agi-0.0.1). ruby-agi is an Asterisk Gateway Interface (AGI) written in Ruby. Project homepage: http://rubyforge.org/projects/ruby-agi/ you can download ruby-agi-0.0.1 from: http://rubyforge.org/frs/download.php/5840/ruby-agi-0.0.1.tgz Feel free to try it. Your feedback, bug report, feature request, suggession would be greatly appreciated. Thanks Mohammad Khan ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
