similar to: Concurrent Calls via Manager Originate

Been researching connecting over TCP\IP to Asterisk Manager API to initiate several concurrent calls to dial out. Prefer not to generate ASCII .call files. Question : I read in places that you use "originate" command and wait for an event back, does that mean you cannot place another "originate" until the event comes back ? Is it true that multiple API connections to Asterisk

Why is using ulaw or alaw an unlikely scenario? I wouldn't use anything but ulaw\alaw. The Bells can compete on price and will if they have to. Where they CAN'T compete is quality. If there were something better than 711, I'd offer that. Well, there is 722, but not many things support it. ---- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ----- Original

What about IAX - SIP or IAX - IAX? ---- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ----- Original Message ----- From: <asterisk-users-request@lists.digium.com> To: <asterisk-users@lists.digium.com> Sent: Saturday, January 28, 2006 5:43 AM Subject: Asterisk-Users Digest, Vol 18, Issue 185 > Send Asterisk-Users mailing list submissions to >

There you go. "if it is doing no other work" is key phrase. A lot of PC can do that these days if all it has to do is re-route packets to different destinations, and guess what, if you make sure silence compression is turned on at the endpoints, you can claim even more streams can be passed through. The trict here is how * stores the mapping pair and how effiecent its lookup process is.

Hello, You can do either, you can send multiple Originate actions in a long line without waiting for a response back(although the responses do usually come back very fast) or you can open multiple connections using each one to Originate a new call. We use the multiple connection method in the astGUIclient suite because if you get a pause or lag in Manager output on a single connection(which does

Isn't it ridiculous that Hammer charges an arm and a leg for any work they do. For systems as large as that one, we just setup a seperate one, connect them back to back and run automated script to burn it in. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of William Boehlke Sent: Thursday, February 02, 2006

This is my first time setting up Asterisk in production and we are buying the Digium TE121-card for use with an ISDN-30 connection. We are considering buying a Fujitsu-Siemens Primergy TX200 S4 - http://www.fujitsu-siemens.com/products/standard_servers/tower/primergy_tx200s4.html - for handling the calls. Quad-Core Xeon, 2.5 GHz / 2 x 6 MB / 1333 Mhz. 2 GB RAM, 3.5" SATA II discs. I know this

Hi, Does anyone know what is the amount of max concurrent calls that can be made in one Asterisk box? I heard that it is 256 and it doesn't depend on how good your machine is. It is the program constraint. What can I do when I need to have more calls than that. I read about connecting Asterisk boxes with IAX. Is it a good solution? Does anyone have other proposals? Cheers Andrew

Hello, Can anyone provide me with an example of how they have set up their dialplan and handset for a receptionist desk that handles a large volume of concurrent calls? I'm having a problem with transferring calls while several calls are either answered or coming into a receptionist's telephone at the same time. Thanks, Colin -------------- next part -------------- An HTML attachment

Hello, I have Asterisk 1.6.1.12 with FAX For Asterisk Components: Applications: 1.6.1.5_1.1.6 Digium FAX Driver: 1.6.1.5_1.1.6 (optimized for core2_32) If I use call file with spool -------------------------------------------- Channel: SIP/IP/DEst No MaxRetries: 0 RetryTime: 10 WaitTime: 50 Application:SendFAX Data:/var/spool/asterisk/test.tif

Hi, Asterisk 1.6.2.9 PHP 5.3 Mysql 5.0 Can anyone suggest hardware specification for 500 hundred concurrent SIP only calls, no codec transcoding, no IVR, no Voicemail or so. Just plain switching. There is only one requirement is to execute one php script on call hangup (h extension) which will do some calculation and update the CDRs. Thanks, Kamlesh -------------- next part

Hi guys. Does somebody knows how to get the concurrent calls from the dial plan? Or. How can i control not to run more than n simultaneus agi applications? Thanks in advance. rv -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140710/70b38e21/attachment.html>

Hi all, I'm looking at some suggestions from you techies out there. Let me explain my scenario. Im a reseller to callshops. I need to take around 100 concurrent calls. Almost all endpoints are sending G723 codec and my peers take G729. Can anyone recommend the Server Specs that is ideal for this scenario. Im planning to lease a server. Calls are purely SIP or IAX2 only. Thanks in advance.

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Can someone please show me what the proper syntax is for Variable: in an Originate. Basically, I'm looking to pass a string into my dialplan as ${MYSTRING} so it's available from whatever Exten => I originate to. Thanks - -jwb -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.2.4 (GNU/Linux)

Hi. I'm having trouble figuring out why I'm not able to make many concurrent VoIP calls on my system. I'm not aiming for a huge number, because I have purposely bought a low powered system, but I would think that I could get more. Here are the details: I have a small-form-factor Asterisk server with an Intel Atom 230 CPU (1.6 GHz, 533 MHz FSB) and 512 MB DDR2 533. It is running Ubuntu

Try Capitalizing Your Actions And The Parameters For Them: Action: Login Login: admin Secret: mypass Action: Originate Exten: 200 Context: stations Channel: SIP/agent07 Priority: 1 Callerid: James Bond Calling That should work. We use manager actions extensively in our applications and the managerAPI is sometimes finicky for capitalisation. and a capitalised first letter of the action and

hello all, does anyone has experiencie using asterisk with a digium CARD using G729 managing 120 concurrent calls with SIP and/or H323??? I wanna know if Asterisk is stable doing this....because we wanna implement it in some locations!! Thanks All!! Sebastian. -------------- next part -------------- An HTML attachment was scrubbed... URL:

<<I am getting "Message: Originate failed" even the phone is ringing on the other end of the line.>> Originate will ring your own extension first and when you pick up, call the other number. If you don't pick up your extension, you will receive the message you see. Bill Seddon ________________________________ From: asterisk-users-bounces@lists.digium.com on behalf

Is it possible to restrict the number of concurrent calls made to a SIP peer? Or maybe the number of concurrent calls made to a particular extension. This way I can avoid asterisk trying to make more voice calls to my remote SIP gateway then I have bandwidth to handle. /davidh

Hi all, I would like to track/store concurrent call usage per user by day/week/month and get server totals by day/week/month. Google comes up with mostly info regarding concurrent call limits, though my goal is to calculate actual concurrent channel usage and add it into reporting. I'm using * 1.6.2 + mysql - realtime (no gui). Any suggestions / open-source / AGI on where to start looking