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...e/store data. So one call - 2 connections to database. So I want to do like this: 100 simultaneous calls , make 200 queries per one mysql connection. WEB developers uses singleton to avoid this issue. Maybe somebody has experience with singleton and phpagi. thanks... -- Pagarbiai / Best Regards, Giedrius Augys -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081005/44b48149/attachment.htm

Hello, I have one problem and I need to disable dtmf (disable rfc2833, info and inband) on one (other peers must support dtmf) SIP peer . Is it possible? Workaround would be use g729 codec with dtmfmode=inband. Maybe there is better solution? Thanks for help. -- Pagarbiai / Best Regards, Giedrius Augys -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090925/cf527e7a/attachment.htm

...as originated, for example "<client's_number> -> Sales". This problem appears when one member can belong to couple queues. Work around would be setting calling name with such information. Maybe there is another way (setting SIP header) to do this? Thanks -- Best Regards, Giedrius -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091207/06244f65/attachment.htm

...n 9 for 'SIP/xxx.xxx.xx-082b9c80' -- Started music on hold, class 'default', on SIP/sip.call.lt-082b9c80 Is it possible to avoid this? I don't want that in this situation (after pressing Flash), asterisk starts Musing On Hold. Thanks for help -- Pagarbiai / Best Regards, Giedrius -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090818/fb15ebda/attachment.htm

...8]: --> [2:OK] Oct 17 07:39:31.86: [22428]: MODEM set DTR OFF Oct 17 07:39:31.86: [22428]: MODEM set baud rate: 0 baud (flow control unchanged) Oct 17 07:39:31.86: [22428]: STATE CHANGE: SENDING -> MODEMWAIT (timeout 5) Oct 17 07:39:31.86: [22428]: SESSION END -- Pagarbiai / Best Regards, Giedrius Augys -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071017/1b1b371a/attachment.htm

...certificate-------- blabla --end certificate---------- in some example a saw, that there should be three files, looks like pfx file should be splited to two separate files? how to prepare those two files(mycert.pfx and ca.cer) for using with HttpClient in ruby? How to use them with HttpClient? Giedrius -- Posted via http://www.ruby-forum.com/.

...0", "test/enter-conf-pin-number_8") in new stack -- <SIP/111-b4091d40> Playing 'test/enter-conf-pin-number_8.slin' (language 'lt') -- Auto fallthrough, channel 'SIP/111-b4091d40' status is 'UNKNOWN' Thanks -- Pagarbiai / Best Regards, Giedrius Augys -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090227/05b6a9b0/attachment.htm

...when the third user joins to conference, others hear "new user have join" and so on. I'll try to do this with meetme, but it always ask me for recording user name.... So is it possible to do that with meetme, or use another conference application? thanks -- Pagarbiai / Best Regards, Giedrius Augys -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081117/811784aa/attachment.htm

...ew user has joined..." announcement in russian language . I want that every user in the same conference number hears announcements in their chosen language (user A hears everything in english, user B in russian) and so on. Is it possible to do that... Thanks -- Pagarbiai / Best Regards, Giedrius Augys -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081126/e7947ae4/attachment.htm

Hello, Is it possible, that during the call one side , for examples clicks the button on the web, and this call starts recording? It's possible with asterisk feature automon and DTMF. So it is possible to start recording the channel using AMI or ... ? Thanks -- Pagarbiai / Best Regards, Giedrius Augys -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081216/64da52a4/attachment.htm

hello, I'm using Asterisk 1.6.0.5 . And I'm creating IVR, and I need that Read application accepts # sign, So is it possible? And maybe there is a workaround? Thanks -- Pagarbiai / Best Regards, Giedrius -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091106/2f2b443c/attachment.htm

...0", "MEMBERNUMBER=Local/123") in new stack -- Executing [s at check-record:2] Set("SIP/sip.call.lt-12d132d0", "MEMBERNUMBER=123") in new stack What I'm missing? Something wrong with ael syntax/structure ? Thanks in advance -- Pagarbiai / Best Regards, Giedrius Augys -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081217/9c8e97d7/attachment.htm

Hi, I have FRITZ!Card PCI card. I have installed misdn-1.1.0 on stable debian 3.1 with 2.6.8. kernel. Then I reboot system, and it doesn't boot, it stops near "Apache2 starting...". I started my system with "recovery" kernel, and tun off misd, then my system works fine. I think it's problem with memory. Has anybody debian and misdn working fine? Maybe you can

Hello, My situation is that , I can't make calls with asterisk, but with x-lite works fine. Asterisk shows , that successfully registers with another SIP server, asterisk sends invite, gets trying, and after 30 secs asterisk gets 408 Request timeout. And as I said , with x-lite no problems. I heard that for comercial purposes, this SIP server detects asterisk , and ignores him. Or maybe it

Hi, I have problem installing spandsp-0.0.3pre24 on FreeBSD 6.1. I get error: configure: error: "Can't build without libtiff" . But I have installed tiff from port tiff-3.8.2. I understand that the problem is about libtiff, and spandsp can't find these libs. So how to fix the problem? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, Is it possible to do like this: play a sound file (if needed play in loop) while php agi script finishes work ? And how to do this? When on my server is huge load , I don't want that client hears silent , but hears music. Thanks -- Pagarbiai / Best Regards, Giedrius Augys -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081124/be5f07cd/attachment.htm

...se function ODBC_FETCH. But how to get result-id variable and use ODBC_FETCH? And another question is, if I execute not SELECT , but stored procedure, and this procedure will return two, three tables? Is it possible retrieve these data from couple tables? Thanks -- Pagarbiai / Best Regards, Giedrius Augys -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081201/67a85573/attachment.htm

Hello, Maybe there is the easiest way to compile additional my module without recompiling all asterisk? Thanks -- Pagarbiai / Best Regards, Giedrius -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100107/cfe8f0b7/attachment.htm

...e peak , I can see : ODBC DSN Settings ----------------- Name: mydb DSN: MYDB Pooled: Yes Limit: 200 Connections in use: 93 Is it possible to free idle connections? When limit was 40, I had lost part of data. My asterisk version is 1.6.0.20 . Thanks -- Pagarbiai / Best Regards, Giedrius -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100121/4f6a9dad/attachment.htm

Hello, I have asterisk 1.6.0.20 and Is it possible to add Reason header on Hangup: Reason: q.850;cause=17 Thanks -- Best Regards, Giedrius -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100221/d29c02b8/attachment.htm