similar to: Sample Config.

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 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 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 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. --------------

Hi all! I am a recent user of dovecot. First of all, the migration from UW-IMAP was definitely worth it. But now I've run into some troubles. Two of my users complain that they can't read their Inboxes, but other folders work fine. The error message I get in the logs look like: lynx imap(gheorghe.bancos): file buffer.c: line 357 (buffer_set_start_pos): assertion failed: (abs_pos

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'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 guys, I looking for some dial plan which can mach on +xxx numbers instead of 00xxx numbers. Some users of main use + instead of 00 for international dial. Is there any solution for this problem? As far as i readed in asterisk is some kind of replacement of characters in dial plan command. Could i use that for archiving this option? Thank you for help. Jonson. -------------- next part

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:

Hi! I did manage to load phone with SIP image : SIP70.8-0-3S, made SEP-MAC.cnf.xml, but phone never read the configuration from it. On the screen it's written "Unprovisioned", and phone is not trying to register with asterisk. Please help!! MihaelaMJ -------------- next part -------------- An HTML attachment was scrubbed... URL:

Dear R users, I'm trying to use the "strucchange" package to determine structural breaks in an ARFIMA model. Unfortunately I'm not so familiar with this topic (and worse, I'm a beginner in R), so I don't know exactly how to specify my model so that the "Fstats","sctest" and "breakpoint" functions to recognize it and to calculate the

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 I have two questions ! 1. What is the best speech recognition engine for asterisk? I have searched and asked on forums and found that lumen vox is best for asterisk bala bla bla 2. For TTS (text to speech) which TTS engine will be better to use ? I have tested Flite , cepstral (i have not buyed lisence for it trial only) but still thinking may be i have a good option ? -- Best Regards

aglia aglia (in italian meat ahi ahi !) Hello i have a serius questions . I have installe din my school samba as PDC for 8 subnet with client win98 Me 95 and i use it like profile server . Last month the school without call me buyed 10 cp with xp home ! It never possible that xp home do not support domain logons, and other nice features . I have samba 2.2.7a with mandrake 9.0 Thank for help me

Hi, I buyed a wildcard e100p and I installed that but when I load asterisk I have this error: Apr 29 18:59:25 WARNING[8192]: chan_zap.c:690 zt_open: Unable to open '/dev/zap/channel': No such device Apr 29 18:59:25 ERROR[8192]: chan_zap.c:5403 mkintf: Unable to open channel 1: No such device here = 0, tmp->channel = 1, channel = 1 Apr 29 18:59:25 ERROR[8192]: chan_zap.c:7526

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