similar to: call-limit problem

Hi All, i have 3 asterisk server in a cluster using a cluster of mysql server via realtime, users can register via DNS SRV. I send/receive calls to an AS5400 via a SIP trunk defined on the realtime sip table, the trunk has call-limit=5. Problem i encountered is each of the 3 asterisk servers will 5 channels each to them instead of 5 for all 3 servers. Is there any solution to this?

Hi, I'm running asterisk realtime, i had prob when a user does not unregister properly. I tested with SPA942 and a PAP2, when phone is registered, i call using the SPA using x-lite no problem, but when i unplugged the power, it does not unregister properly, so asterisk think SPA942 is still registered, when i call using x-lite, asterisk tries to call it.so it gets stuck at [Aug 11

Hi, Is it possible to use realtime for Music On Hold? Is it also possible to store the music/audio files on the database, same way a voicemail can be stored on the database? Thank You Regards, Nhadie

I am using realtime mysql for extensions, sip, and voicemail. Outbound call routing does not really perform well in realtime extensions due to the high number of rows in the database (300k), so I can not use it. It appears with my limited knowledge that the query method is not robust enough for large databases. Given the fact that I already have realtime and mysql configured, what are my options

Hi, What does retransmitting NAT means? I have a client that uses SPA 942, and his phone sometimes cannot be called. i did a sip sebug and i keep on seeing retransmitting NAT. on the realtime it shows that it is registered, so when i try to call it , asterisk thinks it is still online so it tries to reach it instead of saying it's unavailable, [Oct 9 11:10:33] -- Called 103100 it

Hi All: I'd like to use Realtime Static in terms of the performance concern about dynamic realtime. Assume that I create a table: as following: CREATE TABLE `extensions_table` ( `id` int(11) NOT NULL auto_increment, `context` varchar(20) NOT NULL default '', `exten` varchar(20) NOT NULL default '', `priority` tinyint(4) NOT NULL default '0', `app` varchar(20)

Does anybody know the sql type for the "call-limit" field under sip peers? Everything on voip-info is missing that entry.

Hello, I am running version 1.4 with realtime support. I've set (for Snom phones 300/320/360) a call limit of 1 (incominglimit and outgoinglimit fields in the database). - When I used 1.4 SIP SHOW PEER show that it has a call limit of 1. The problem was that when such a phone received a call and did attended transfer it was left "in use" and could not receive new calls. -

The instructions on the wiki for asterisk Realtime give the extensions schema with the priority field set to be tinyint(4). This of course cannot hold the value 'hint' The question I have, is the solution simply to set the field to varchar(n) as that will then hold 'hint' or any integer value (if you make it big enough) ? Or is there any known places in the realtime

Hi, Is it possible to detect where the call came from and route it out to different sip trunks. e.g. i have user 100300 when that user calls outbound i will make him use of [sip-trunk-100] another user, 101300 when that users calls outbound i will make him use of [sip-trunk-101] actually the 100 and 101 at the beginning of the username is the accountcode i used for cdr. hope my question

Hello list, asterisk 1.4.30 2 situations in which call-limit should work, but it does not : [Jul 8 09:15:49] WARNING[11132]: app_queue.c:3272 try_calling: The device state of this queue member, test12, is still 'Not in Use' when it probably should not be! Please check UPGRADE.txt for correct configuration settings. In sip.conf I have : limitonpeer = yes In my realtime sip_buddies

Good morning, I'm having suddenly cut-offs and I don`t know why. It's been hapenning since I enabled cdr_odbc/func_odbc in my system. We use func_odbc to register some queue member's events (login, pause, etc.) at an external DB ('remoto' connector) and to uptade local tables at a local DB ('local' connector). Currently we are usind cdr_odbc to Postgresql and cdr_addon

hi all, just wondering what's happening here: i have a pap2 and an spa941. everytime i call my spa from my pap2 i can see it being added dynamically on the regcontext: [Oct 7 11:59:08] -- Saved useragent "Linksys/SPA942-5.2.8" for peer 100100 [Oct 7 11:59:08] -- Added extension '100100' priority 1 to sipregcontext but from spa to pap2 i dont see it, i looked

Hi, I'm using opensips as the registrar server for my users. I am redirecting calls going out to pstn to my asterisk server. call flow is basically: ua --> opensips server --> * server --> sip gateway provider if (uri=~"sip:00[0-9]*@sip\.myserver\.com") { xlog("L_INFO", "Call to PSTN\n"); #strip(2); #prefix("011");

Hi, I'm using heartbeat as a failover for my asterisk server. on the active server 1 i have 10.10.10.1 eth0 10.10.10.3 secondary eth0 asterisk listens to the secondary ip, so that if server 1 fails, server 2 will then get that IP. so if server 1 fails, server 2 will have the IP 10.10.10.2 eth0 10.10.10.3 secondary eth0 problem is i have to bind asterisk to the secondary IP if dont, i

hi, hs anyone able to make video to work on asterisk? i tried following this: http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+xten+eyeBeam i can see that eyebeam is trying to broadcast a video but the other eyebeam is not receiving it. i tested the same setup but this time using ser with rtpproxy and eyebeam video works fine. any ideas? where do you think should i start

>From voicemail.conf: ; Maximum number of messages per folder. If not specified, a default value ; (100) is used. Maximum value for this option is 9999. ;maxmsg=100 ; Maximum length of a voicemail message in seconds ;maxmessage=180 ; Maximum length of greetings in seconds ;maxgreet=60 I would like to configure these parameters on a per mailbox basis using Realtime voicemail. I

Hi I've just had a request from a customer who wants to use Busy Lamp Feed. I've had a look around and it would appear that you have top use the 'hint' priority. We are using asterisk 1.4.17 with realtime and the priority column in the extensions table is a tinyint so obviously I can't put hint in there. Has anyone any experience of working round this problem? Thanks

Hi, Whenever one of my trunks becomes unreachable or reachable again.. On logs i got the msg as follows: Jul 31 15:15:51] NOTICE[15112] chan_sip.c: Peer 'voiptrunk' is now Reachable. (12ms / 2000ms) [Jul 31 15:15:51] WARNING[15112] res_config_mysql.c: MySQL RealTime: Failed to query database. Check debug for more info. I dont wanna turn on the debug function because theres a lot of

Hi, Would just like to inquire if anyone here has a setup of asterisk to send traffic to AS5400 connected to an SS7-PRI.? this is more of a AS54 question, as i've been reading and i always stumble upon PGW2200 as a requirement to handle SS7 signaling on the AS54. Has anyone able to send calls from asterisk to an as 54 with SS7-PRI without PGW2200? TIA Regards, Nhadie --------------