similar to: Using Realtime queues and queue members

Juan, I am running a Calling Card application on a Dell PowerEdge 2850 with Asterisk 1.0.7. Recording conversations I have seen on my server causes the processors to burn more than necessary so I would recommend what William from Signate recommended: " Consider saving recorded calls in a database on a separate server. It will be simpler to build a retrieval interface that does not

Hey all, I know you all saw the topic and let out a groan. However, I understand how to get an overhead paging system to work with respect *, however I am now looking for a small(?) paging amp, that I could hook 3 or 4 horns to. I would like to just have the * extention be routed to a soundcard and out an output, so I would like an amp that is voice signal activated. Has anyone found anything

Hi, Asterisk 1.4.31 here. I have a queue with members defined, and those member have member names "member 1", "member 2", etc. They are in a realtime DB. When I modify those member names (column membername) the changes aren't reflected in the queue status ("show queues" from cli. They aren't reflected when a new call comes in, or when I reload the

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]

Hello, I'm using MySQL realtime Call Queues (table /queues/ and table /queue_members/). I would like to ring the members of the call queue in a certain order. Therefore I use ring strategy /lineair /and I put the members into the table /queue_members/ in the order in which they have to be rang. So I have the queue : | name | musicclass | announce | context | timeout |

Hi there, As described on voip-info here http://www.voip-info.org/wiki/view/Asterisk+RealTime+Queue, if I use realtime queues, alterations to the list of members don't alter until a new call joins the queue. Is there anything I can do about this? I've tried looking for a bug number, but to no avail. I use realtime queue members for logging in and out, and I'd like it to take

Hi, So I'm using Asterisk Realtime Queues and Queue members on 1.4.28. I've noticed if there are no people in the queue when a call enters, even after a queue member enters, the call is never rang to him. From the debug, it seems that Asterisk is only grabbing the queue member list upon entering the queue. And not again until another call enters the queue. As a result, a caller will

Is anyone out there using SIP Witch in conjunction with Asterisk? It claims to be able to "enhance existing IP-PBX solutions such as Asterisk", so maybe it can be used as a simple means to provide secure/encrypted calls. GNU SIP Witch - Summary <http://savannah.gnu.org/projects/sipwitch> GNU SIP Witch - GNU Telephony <http://www.gnutelephony.org/index.php/GNU_SIP_Witch>

Hi I am setting up a phone system for a small office. The office will have 5-8 phones and a fax line. There are 4 hunt lines coming into the office. We have made no hardware purchase yet. Being an asterisk newbie, before I suscribed to this list I just assumed that I would buy voip phones and connect all the phones to a private ethernet network. However, I see many people inquiring about FXS

Can anyone comment or share experences with using Dell 2850's with Asterisk. Proposed config is 2850, 2 x 3.6g procs, 2 g's of ram, 4 x 36g 15k rpm drives raid 10, Digium TE411P ( the echo cancelling cards ). Expected load is 1 or 2 pri's (most likely 1 ) 100 Polycom phones on the local network, 15 phone on a remote T1. 6 phone remote via the internet using IAX, Voicemail for

Hi all I've been doing some testing on realtime using mysql, an have a little question that could not find the answer to or maybe its not posible at this time. Is there a way use "register=>......" on a DB using realtime. For the moment I use it in sip.conf. It will help me a lot if this could be store on a DB somehow. commets or sugestions .... ? thanks Billy

Hi, is geoRSS overlay supported with openstreetmap ? I can''t get it to work. Google maps georss http://blogs.intermedia.uib.no/motel/experiments/bergenGeoRSS.html Openstreetmap georss http://blogs.intermedia.uib.no/motel/experiments/bergenOSMGeoRSS.html As you can see, nothing shows in the openstreetmap example. Regards -- Rune Baggetun

Hi all, first post. My company's office in the UK is soon going to get a Cisco VoIP solution system. What I am interested in, and couldn't find googling, is if it is possible to connect an Asterisk solution to the Cisco system and have all the nice advantages of it (mainly calling the extensions and directly reach the other office). Thanks, have a nice day Simone

Our company, Signate, is going to be offering a three day introductory Asterisk training course, the first of a series. The first class will be in San Francisco the week of June 28. It will be a beta test to get the kinks out and we will not charge for the class or the materials. Anyone who attends is responsible for their own travel and lodging, if necessary. I doubt anyone reading the lsit

Hi, I'm using queue configuration as follows: - queues from* queues.conf* - queue_members from *external Database thru ODBC*, using* Local channels * as interface - sip extensions from *external Database thru ODBC* When a call is sent from queue to an interface (local channel), it is answered but a message appears at the CLI: *[Nov 4 16:56:04] WARNING[13951]: app_queue.c:3014

Hello, using asterisk 1.8.11.1 using realtime queues When trying to remove a queue member, I get the following : -- Executing [122 at from-TESTCORP:2] RemoveQueueMember("SIP/testcorp5-0000000c", "testcorpq1,SIP/testcorp7") in new stack WARNING[18788]: app_queue.c:5653 rqm_exec: Unable to remove interface from queue 'testcorpq1': 'SIP/testcorp7' is not a

We use agents and queues, with CVS HEAD. I've read up on realtime queues and queue members, (and actually understand it!) but there is no reference to agents. Is it possible to have realtime agents as well ? Julian.

Hi all. What is the best hardware configuration to handle this following scenario? - 4 IVR menu with conference applications for each option; - Only SIP/g711 user access - 3500 simultaneous users(800 at the beginning) - No ZAP channels Where is the most important point of failure? CPU? Ethernet? RAM? Im planning to separate in three servers: Server01: 01 Xeon 3Ghz getting the 1st level of

Hello friends, I have a script that uses the sftp client to transfer a file to another server using PK authentication. It was working until I upgraded from OpenSSH 2.5.1p1 to 2.5.2p2-1 yesterday (on a RH Linux 6.2 system). The sftp command is: sftp -o "IdentityFile ~/.ssh/id_dsa" \ $PUSERNAME@$PSERVER >$TEMPFILE 2>&1 <<-! cd $PDIR put $DOC_ARCHIVE ls quit !

It is amazing to me at this point that there is not an official Digium list of supported servers (including 1u models!). Clearly the number 1 issue with the Digium PRI cards is the server that they are used in. The new cards even go as far as listing server that DO NOT work on the Digium site! The wiki references are old and do not have any testing parameters. C'mon guys! Certify a