similar to: Asterisk: Agent logs into queue, and there are calls in the queue, but calls don't go to agent.

Anyone know how to get app_queue to send logs to MySQL or any other sql server. I found info for cdr's and even configs but nothing on queue_log. If sql is not supported in the current app_queue I will be willing to pay someone to add it. John Bittner Simlab.net

Seems to be all morning. I have not been able to access for several hours now. W -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Marcel van Kaam, Fonetica Sent: Tuesday, June 14, 2005 7:18 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] VOIP-INFO down? Hi

Here is some dialog from the Console: -- Starting simple switch on 'Zap/13-1' Feb 10 07:22:36 NOTICE[21105]: chan_zap.c:6063 ss_thread: Got event 18 (Ring Begin)... -- Executing Goto("Zap/13-1", "mainmenu|s|1") in new stack -- Goto (mainmenu,s,1) -- Executing BackGround("Zap/13-1", "thank-you-for-calling-poker -support") in new stack

[root@asterisk zaptel]# modprobe ztdummy WARNING: Error inserting zaptel (/lib/modules/2.6.5-1.315/misc/zaptel.ko): Invalid module format WARNING: Error inserting zaptel (/lib/modules/2.6.5-1.315/misc/zaptel.ko): Invalid module format FATAL: Error inserting ztdummy (/lib/modules/2.6.5-1.315/misc/ztdummy.ko): Invalid module format FATAL: Error running install command for ztdummy [root@asterisk

Anyone get the Nortel 1140E phones working with Asterisk ? These look like great phones and I would like to start using them on our deployments. I know these will work with Asterisk but the sample config files are hard to find. My next step, if I cant find anything on this list is to purchase a Nortel Communication Server for testing. If anyone has a used NCS that works with these phone via SIP

Anyone know why zaptel would ignore a facility message from an ISDN PRI. I am trying to get Callerid name to work. The carrier says it on and I see it in the pri debug but asterisk never gets it. Any help would be appreciated. Thanks John Bittner Simlab.net Protocol Discriminator: Q.931 (8) len=9 > Call Ref: len= 2 (reference 572/0x23C) (Terminator) > Message type: ALERTING (1) >

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040331/c2abf19f/attachment.htm -------------- next part -------------- Hello, Is there any work in progress for supporting SMDI in Asterisk ? if Not, could anyone tell how to get started implementing it for Asterisk. Regards, Tony

Anyone know how to let asterisk deal with the progress codes coming from the carrier? The problem I am having is when a customer calls an invalid number the carrier tells me the call is invalid via a progress code but doesn't route me to a recording (this number is invalid). Instead they hang up on me causing a fast busy or sometimes hold up the call with dead air for 15 to 30 seconds then a

Hi folks, I'm running asterisk 1.2.10 and I need to use agent dynamic login. I read some doc and follow some tutorials but the agents can't login into the queue. Asterisk ask to me to dial the password agent and after this, it doesn't do nothing ( it doesn't tell login ok or login incorrect..). In the * console if I do show agents, any agent are logged. Any help will be

Given: Queue(foo|tHnr||bar) where queue foo includes something like IAX2/gw/18005551212 should # transfer work on the remote phone? A read of app_queue.c looks like it ought to work, but all I get is dtmf sent to the caller. (Incidently, I'd really prefer to be able to hit eg * during the announcement to have app_queue continue on as if there were a timeout. Has anyone looked into doing

See comments below. Henry Devito wrote: > Here's the problem. When I call 555 to login, it asks for the agent ID > which I enter as 501, it asks for the password which I enter as 1234, > then it asks for the extension I dial 501 It then says that extension is > not valid. What am I missing? Of course 501 is valid I can make and > take calls from it now. > > >

Hi, I am on my next venture now, Need to set up 3 queues. I would like these setup using the agentcallbacklogin. Does anyone have an example of this? I have looked through the wiki , but all that did was confuse me. One of the problems I'm having is how do I configure my extensions.conf to dial the agentcallbacklogin -------------- next part -------------- An HTML attachment was

Hi all! My problem is that calls being placed in the queue, and are waiting while the agents are busy, when an agents is then free they gets connected to the agent but there is silence (no voice). If a caller has not to wait in the queue, there is no problem. My agents have an iax2 client, and imcoming calls are over SIP. queue.conf: persistentmembers=yes autofill=yes ringinuse=no

Anyone having any problems with CVS-HEAD-12/24/04-15:59:15 and ALERT_INFO I have a system setup with polycom phones configured to auto answer on internal calls. When we upgraded to the latest CVS the auto answer stopped working. My dialplan has not changed. I did a sip debug and I dont see the alert-info tag in any of the sip traces. Any help would be appreciated. Thanks John Bittner Simlab.net

When agents are on the phone, and the CLI queue show command shows their status as busy, the queue still tries to send them calls. Running Asterisk 1.6.0.17 and using AddQueueMember to dynamically add agents. ringinuse is set to no for queue. Agents are using Polycom 430s. dialplan: exten => s,n,Queue(orders,itT,,,80) queue definition in queues.conf: [orders] maxlen=20 queue-thankyou=

Is anybody else out there using VoicePulse Connect and having problems this morning? I just noticed that they have absolutely no contact information in their website.. just want to make sure I didn't break something in my asterisk configs. -fedl

Is anyone successfully using call queues and call groups? If so do you have an example configuration? The wicki and mailing list information I found is pretty old. Thanks! Paul pmahler@signate.com

hello list, i have problem when i am useing wrapuptime with agents.conf my agents.conf looks like this [agents] autologoff=15 musiconhold => default wrapuptime=50000 group=1 agent => 1001,4321,Mark Spencer recordagentcalls=yes my aim is every call needs have wrapuptime of 5000 ms but when ever a call comes its directly connecting not wating any more. your views will be highly

I have queue set up in realtime on Asterisk 1.4.2. Below is the senario that is happenening :: I have created a test queue with only one agent. Once I call the test queue the agents phone rings if the aagent is logged on. everything till here is fine. Now if the agent does not pick up the call, the call automaticaly disconnects after 15 secs as set for the queue, till here also it is fine. But

First off this is not my work for extensions.conf it is modified from http://leifmadsen.wordpress.com/2009/07/15/migrating-from-agentcallbackl ogin-to-standard-dialplan-methods-part-1/ So credit to Leif Madsen <http://www.leifmadsen.com> But as to my question [AgentLogin] ;A replaced version of AgentCallbackLogin() using a GoSub() ; exten =>