similar to: Bug with Realtime?

Hi, I think I already know the answer to this question, but is there any way to do the following using realtime? Or do I have to create a full dialplan for each client without using includes? [client1_phones] include => client1_internal include => client1_outgoing_calls include => test_calls include => parkedcalls [client2_phones] include => client2_internal include =>

Hi, Please take a look at the following extensions.conf:- exten => _11XXXX,1,NoCDR() exten => _11XXXX,2,Dial(SIP/${EXTEN},10) exten => _11XXXX,3,VoiceMail() I'm already using realtime for some extensions/users/voicemail. Is there any way to do the following at point 3?:- Lookup the realtime users db and read the MailBox column for that buddy. If the mailbox column is empty, play

Hi, I'd like to be able to create contexts in real-time when I add new clients to my asterisk box. Currently, I have to create a blank context in extensions.conf and add:- switch => Realtime/@ Is there any way to avoid the step of creating the blank context and simply include all the entries from the database? Thanks Dan -------------- next part -------------- An HTML attachment was

Is there a guide anywhere which runs through how to set up asterisk with mysql? I've looked and almost all the document misses out relevant information. Thanks Dan Journo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050920/1ce45adb/attachment.htm

Hi, I have what I think is a relatively advanced question. Any help is appreciated, even if it's not a complete answer. I am using Asterisk in mostly realtime fashion, specifically SIP registrations are in a MySQL table. This works fine (mostly). I also set a few variables in the setvar column, like this: callerid_internal=test <710>;did=5555551234 Again, this works

hi, all i try to confiture realtime queue, but not work, details as below: Insert into queue_table(name)value('95040654321'); INSERT INTO queue_member_table VALUES ('', 'Zhang Shukun', '95040654321', 'SIP/1001', 2, 1); INSERT INTO queue_member_table VALUES ('', 'Li Aiwei', '95040654321', 'SIP/1002', 2, 1); INSERT INTO

Hi, When a VOIP user dials an external number, the calls are routed through our SIP provider. Is there a simple way to check whether the DDI exists locally before dialling out to the sip provider? Something like GotoIfExists(5551234 at incoming_calls) Currently, I'm paying for all calls, regardless of whether they exist locally. All DDIs exist in the incoming_calls context. Thanks Dan

I am using realtime on two boxes, one running 1.4.10.1 and one running 1.4.11. Everything works fine except for when I make a database change, such as a phones password. I change the DB, I prune the peer, I see it is gone and then I see it show up again in "sip show peer xxxx", but everything is not being updated. The phone will not register even though the DB and the phone have

Does anybody have realtime queue members working? Not the queues themselves, just the members. I have realtime working for voicemail and sippeers, but I can't get queue members to work. Here is what I have: res_mysql.conf: [general] dbhost = 127.0.0.1 dbname = ASTERISK dbuser = myuser dbpass = mypass dbport = 3306 dbsock = /tmp/mysql.sock queues.conf: [general]

I am in the process of setting up several * servers using realtime and connecting to mysql. I am trying to figure out if I should just use one database and one set of tables for all of the user data. Or if I should have separate databases for each * box. The boxes are independent of each other in that customerA only connects to box A. They will never fail over to box B or anything like

Hi! I am proving Asterisk 1.2.24 in realtime with MySQL 5.0.27 using Idefisk softphones. I followed the steps of "how to" of voip-org but always have this error: Sep 25 20:29:07 WARNING[12000]: res_config_mysql.c:360 update_mysql: MySQL RealTime: Failed to query database. Check debug for more info. Sep 25 20:29:07 WARNING[12000]: res_config_mysql.c:360 update_mysql: MySQL RealTime:

Hi everyone, I have a question concerning DNS SRV lookups. The situation is like this: - one central Asterisk server - many domains with SRV records, let's say we have bar.com and doe.com Now the question is: if the SRV lookup is done for foo@bar.com the call is mapped to foo@myasterisk.mydomain.net. Is that correct? If so, I have a problem: if somebody calls foo@bar.com, Asterisk

Hello, I have Asterisk set up with 6 extensions. When a call comes in, I use a Dial command to call all the extensions together until someone picks up. The problem is, when there is an incoming call and an extension is in use, if the extension puts down the phone while the incoming call is still ringing, that extension doesn't ring. This is because when the Dial command was executed,

Is it possible to use the "include" command to include other contexts if you are using realtime for extensions? I've searched voip-info and some people were asking about it, but I didn't find a real answer anywhere.

The Asterisk Development Team has announced the release of Asterisk 13.8.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.8.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: New

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

Hey Guys! We had migrate asterisk 1.2 to 1.8 now big issue is queue system. Before we had 3 queues and we were using AgentCallbackLogin but now its quite difficult to use AddQueueMember. Is there any easy way to logged into multiple queue using AddQueueMember ? and restrict agent for specific queue ? -S -------------- next part -------------- An HTML attachment was scrubbed...

Hi Guys, Basically, i'd like to be able to force a call to drop by using one of the following methods. Does anyone know if it is possible? It has to be more or less realtime. a) Issuing a command to asterisk via telnet. b) Altering a field in the realtime database c) Any other suggestion or possible method. Thanks in advance. Dan -------------- next part -------------- An HTML attachment

Is anybody using 1.2 (or 1.2.1) in a production network using Realtime (voicemail, sip or extensions) with 100+ SIP phones? If so, what are your experiences? We've been running 1.0.3 for about a year and it's been rock-solid. We'd like to upgrade to Realtime and 1.2, but I'm afraid of killing our stability. Obviously, we'd do it in stages (upgrade to 1.2, then realtime

Hi, I have Ubuntu server 10.04 64bit, and Asterisk 1.4.34, compiled from sources. I installed freetds-common,freetds-dev, libct4, libsybdb5, freetds-bin, but, when I run configure and then make menuconfig in section "Call Detail Recording" -> "cdr_tds" it's "disabled". It only writes that "Depends on: freetds(E)". On another server (same