search for: augys

...ata. 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

...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

...; [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

..."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

...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

...as 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

...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

..."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

hello, I've callcenter and our queue members want to see on their IP phone's display queue's name , from which incoming call was 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

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

Does anyone know of any FOSS tools that can assist in benchmarking ISC DNS and DHCP services? I would like to simulate X number of users attempting to pull an IP, resolve DNS names, etc. I would also like to test TCP connections into a CentOS server. We have some software that communicates with equipment via a TCP connection and I want to simulate thousands of TCP connections coming into one

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

...on 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

...sas . -- Auto fallthrough, channel 'SIP/sip.call.lt-01993050' status is 'UNKNOWN' As I read documentation, function hash gives posibility to get values using column name. But my test was unsuccessful. Maybe somebody can help... Thanks -- Pagarbiai / Best Regards, Giedrius Augys -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081202/c992d0fa/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

Hi, I have debian etch 4.0 machine (2.6.18) with two TW-ISDN PCI (Hfc) cards. I use bristuff-0.3.0-PRE-1y-e (asterisk-1.2.17,libpri-1.2.4,zaptel-1.2.16). I also have patched zaphfc with zaphfc_0.4.0-test1_florz-13.diff.gz (I load module: insmod /usr/src/bristuff-0.3.0-PRE-1y-e/zaphfc/zaphfc.ko modes=1 debug=1). So i want to test two cards and make loop between them. So one card would be NT,

Hi, I have cable modems Arris with MGCP protocol. And I need PacketCable NCS patch for Asterisk. http://asterisk.urtho.net/ doesn't work! -- Pagarbiai, Giedrius Augys Siauliu Universitetas, IST IP telefonijos inzinierius Tel. 8 41 590408 Mob. Tel. 8 678 05790 el. pastas voipas@gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060614/27aef641/attachment.htm

Hi, I am testing Nokia E60 with Asterisk. And I noticed that if another side is busy, nokia is still calling (I hear alerting), it do not show that another side is busy. Maybe somebody has noticed the same problem too adnd solved this one. I made the same tests with Xlite and don't have any problems like nokia. Please help me -------------- next part -------------- An HTML attachment was

Hi, My older kernel was 2.6.18. Now I have compiled new kernel (2.6.20). Is it necessary to re-build zaptel drivers (I'm just using ztdummy). Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070305/b2d47ba2/attachment.htm