similar to: "+" dialplan

Hello, I have a problem with some variables in 1.8.6.0. I set on extension the following lines: exten => h, n, Set (CDR (LLP) = $ {CHANNEL (rtpqos, audio, local_lostpackets)}) ; lost packets by local end ** exten => h, n, Set (CDR (PCR) = $ {CHANNEL (rtpqos, audio, remote_lostpackets)}) ; lost packets by remote end exten => h, n, Set (CDR (ljitt) = $ {CHANNEL (rtpqos,

Hello, I intend to buy a FXO/FXS device from Linksys. I'm thinking about SPA3102. What you guys think about it. Is ok, is working with asterisk, can i use it like voip peer. Thank you for your advice. Jonson. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070117/93bc7fdb/attachment.htm

Hello, I just buyed a SPA3102 from Linksys. Can anyone help me or guide me to configure voice part on it. I cannot get it if I can use like peer for my asterisk. Please help me with some tips. Thank you guys. Regards, Jonson. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070126/a49e3bdb/attachment.htm

Hello, I just want to put all my sip accounts in mysql and asterisk use it from mysql. How can I do that, could you be more specific because I readed alot on wiki and i'm lost... I don't know what to modify in Makefile from channel directory. I use asterisk 1.4.4, that is already compiled and i also have CDR in mysql. I must create manny accounts and I want to realize that from mysql.

Hi, compiler-rt builtins currently doesn't build on MSVC 2019, I the problem is that compiler-rt\lib\builtins\int_math.h includes the header ymath.h. according to eg. https://docs.microsoft.com/en-us/cpp/c-runtime-library/reference/finite-finitef?view=vs-2019 the header to include is float.h also the ymath.h file contains the comment /* ymath.h internal header */ so probably shall not be

Hello I recently get a Cisco 7940G IP Phone and I try to make several things with it and I en counted many difficulties: 1.) I tried to unlock the phone and to set manually IP Address, Netmask, Gateway etc. I don't get any luck. 2.) I tried to upgrade firmware like they said with tftp server... I downloaded: P0S3-08-11-00.zip and I uncompressed the files in tftpboot directory. I don't get

Hello, i hev a subscription to a international voip provider and I want all calls for numbers _001xxxxxxxxxx to go through my voip provider. I tried many settings in sip.conf , extensions.conf and iax.conf. Please give me some simple example for how can i transfer the specified calls to my external voip provider. What may I put and where in witch file. Thank you for your support. --------------

Hello, I'm looking for some hints on how to run pagemaker succesfully in wine. I run debian unstable and wine daily builds from http://gluck.debian.org/%7Eandreas/debian wine main When I installed Pagemaker 6.5 I had a lot of trouble with the dialogs, some could not be selected sometimes, stuck under other windows. It crashed two times when i tried custom install, as soon as i selected som

Hello, I want to limit calls per sip account user. How may I realize this setting? For example I want to limit to 10 min all possible calls from an account or to limit external calls to 10 min and local call remain unlimited. Thank you for support guys. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I have been researching Asterisk and have a few questions. Is it possible to allow users to only call certain area codes? Reason for this is that the plan from BroadVoice allows unlimited calling to certain countries. Any country outside of this plan would be charged on a per minute basis and I am trying to avoid that. Another question is if there is a way to distinguish a land line

Hello, I want to use an Huawei stick model K3765 which support voice with asterisk. I'm begginer with this kind of interaction from asterisk with external devices. Can someone guide me what should i configure to use this device? Thank you for support, Regards, Jonson. --- www.Mobile-Wi.Fi

Hello, i recently installed last version of asterisk (Asterisk 1.4.18.1 built by root @ h-gw on a i586 running Linux on 2008-03-23 00:26:44 UTC) and everything is ok but when i call an extension i cannot hear anything. I don't get any visible error on sip debug... i changed the codecs... everything is the same... Can someone help me with that? Thank you. Jonson. -------------- next part

Okay, i'll move my discuss to asterisk-users. Thank you. On 1/17/07, Tzafrir Cohen <tzafrir.cohen@xorcom.com> wrote: > > > On Wed, Jan 17, 2007 at 04:39:03PM +0800, ??? wrote: > > Jonson Player wrote: > > > Hello, I intend to buy a FXO/FXS device from Linksys. > > > I'm thinking about SPA3102. What you guys thik about it. > > > Is ok, is

Hello, I lookin' for a call in number from UK or USA. Can somebody offers me a peering for this or specify any sip provider that offers this thing? Thank you very much, Jonson.

This is my first time to use R.  For clarification: I made an oa design for 4 factors with the following levels: one with 4 levels, 2 with 3 levels, 1  with 6 levels.  Using DoE package, I have generated 72 runs (setting the columns="min3"). However, the numbers of generalized words of lengths 3 and 4 is still equal to 2.000000e+00 and 4.559372e-33, respectively.  I am concerned that

Hello, i have a SPA3102 and asterisk v. 1.2.18. I also hev a mysql database with cdr. Well all I want is to receive incoming calls from pstn on specified sip account (suppose 8000), and to initiate outgoing calls from all my asterisk sip accounts through SPA3102 device. Someone can explain me what may i set on SPA and asterisk to do this thing. Thank you for your support. -------------- next part

Hello, i just installed asterisk 1.2.15. I got this error message. Somebody can help me? Thank You. Feb 27 11:47:43 NOTICE[17086] cdr.c: CDR simple logging enabled. Feb 27 11:47:44 WARNING[17086] pbx.c: Already have an application 'Pickup' Feb 27 11:47:44 WARNING[17086] loader.c: app_directed_pickup.so: load_module failed, returning -1 Feb 27 11:47:44 WARNING[17086] loader.c: Loading

Hello, i just want to activate SMS service between my asterisk local sip accounts and between asterisk and local sip accounts. How can i do this thin? Also i tried smsq to an account but all i obtained is a error message: ---<Cut Here>--- May 22 13:09:37 WARNING[4829] pbx_spool.c: Unable to open /var/spool/asterisk/outgoing/smsq.motx.0.1179827912-4429.1: Permission denied, deleting May 22

Hello ppl, I want to make a special dial plan for routing calls to a peer which has an pin protection. Normally if you want to call through that peer you must first enter pin for example 1234# and after that you hear the tone from line and after that you can dial desired numbers. I tried something like that, but doesn't worked. Did somebody have some clues? exten =>

Hello, For the moment I have the following settings in my sip.conf. I want to optimize them to archive the following things: - for the moment all my users will re-register too often. I want that only lagged users to re-register quickly. - check from time to time all users but no too often to see if is logged and can be called. Overall i want only lagged users to reregister and users with good