similar to: Simple example for call transfer.

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 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

https://bugzilla.netfilter.org/show_bug.cgi?id=1069 Bug ID: 1069 Summary: fix incorrect checksum on big endian platform Product: libnetfilter_queue Version: unspecified Hardware: mips OS: All Status: NEW Severity: enhancement Priority: P5 Component: libnetfilter_queue

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.

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,

ok when i go to play half life 1 off steam, it gives me a weird error, heres a video of what happens.... http://www.youtube.com/watch?v=O-N9YV12C3A help would be nice... i kinda would like to play this game

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 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'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 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 a very strange trouble with samba 3.0.33 when I integrate a Linux server in my Windows 2003 AD. I do : - kinit administartor, it's work. - klist, it's work too. - net join ads -U administrator, it's work. I hev the message that my computer has join the domain and I see the Linux in my Domain. - wbinfo -t give me "checking the trust secret via RPC calls

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

Hello *'s, First Of all Marry Christmas, I want to setup asterisk at large means "my main asterisk server placed in my office(in Pakistan), and some offices outside Pakistan and i want to connect these locations to my main * server (in Pakistan) on remote locations i'll used asterisk can i do this or may be i changed my plans kindly guides me. Thanks In Advance. Adnan Ahmed.

Hello, I have a strange situation with an SPA3102 FXO/FXS device. I'm in situation that when i receive a call from PBX line I must forward the calls to 2 VoIP numbers. Right now i have the following settings: (S0<:1010 at GW1>). I want to forward at 1020 too. I tested (S0<:1010|1020 at GW1>) and doesn't work. Did you have any other ideea? Thank you.

Few weeks back local telco introduced option of custom ring tones. I am not talking about your phone ring tones but about ring tones you hear in your headset while phone is ringing on the other end. If I understand correctly, ringing tone is generated localy on asterisk if you are connected to phone network with E1/T1 connection. Which means that instead of regular beep-beep tone we could send

Ok. It's a Firefox Add-on: https://www.eff.org/https-everywhere Questions: 1) But: Why can't i find it on the offical Firefox Add-ons site?: https://addons.mozilla.org/en-US/firefox/ 2) Did anyone audited the "HTTPS Everywhere" code? 3) Can someone trust this Add-on? Is it safe to install/use? 4) If it's so great why isn't it more prevalent? What's youre