similar to: transfering incoming message from app_queue

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

Hello, I'm looking for a similar feature... Dial a number via ZAP/g1 after the line gets answered wait 10 seconds send DTMF Regards, Marc -- Network Manager Marc Storck LuxAdmin.Org mstorck@luxadmin.org Internet Service Provider http://www.luxadmin.org 15, route d'Esch Phone: +352 2727 3030 L-4544 Belvaux Fax: +352

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

When an agent logs into a queue using AgentCallBackLogin, he should be ready to take calls until he logs out right? For some reason the first time a customer calls the queue, it rings the agent just fine but after the agent hangs up the phone and the next caller calls the queue, no more calls will be transferred to the agent. He shows as logged in, but the calls wait in the queue forever and

[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

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

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

I've been doing a lot of reading over the last few weeks on Asterisk, and will be implementing a test system this week to play with. I've got two questions in regards to the ideal implementation for our company. First, has anyone written any drivers to interface with proprietary phones? Specifically we have a comdial system and if we could use our existing 35 phones instead of having to

Hi! I've got a Comdial PBX that I would dearly love to replace with an Asterisk box. However, for various reasons, it appears not to be in the cards. Regardless of what management does, or does not, want, our current VM solution -- some Dialogic card with a "KeyVoice" application -- is dying. I'm 90% sure it's hardware. I'd rather shoot myself than replace the

I don't know if there is some fault in compiling or a bug of the new R-1.7.0 version: cl.pam.2 <- pam(as.dist(1-cor(mel.data)),2) plot(cl.pam.2) perform a right partitioning and silhouette plot on the old R-1.6.2 instead "Error in clusplot.default(x$diss,...... ; x is not numeric" is the output on the new R-1.7.0. Same platform: RH8.0 i686. Some suggestions? A.S.

Hello: I have successfully install spandsp and patch asterisk with it. But when I received a Fax is garble or shrink. Does any one know why???... Am using a PRI T100P card to receive the fax and save it to a tiff file... Any help will be greatly appreciated. Here are the versions. Latest csv from asterisk, spandsp-0.0.1k.tar.gz redhat 7.3 T100P has its own IRQ. Any help will be greatly

I'm having (I think) timing issues in relation to bridged T1-T1 calls via dynamic spans. Fax calls are intermittently working, but voice is fine. My box has a Sangoma A400 inside it as the primary Zaptel timing source. My T1 PRIs that are hooked to the box come in via a foneBRIDGE2 (dynamic TDMoE spans). PRI #1 is the telco and PRI #2 is an existing Comdial FX-II. For some reason, bridged TDM

One of the connections my asterisk PBX has is an analog extension from a Comdial hybrid. On the Comdial system, message waiting is turned on by dialing *3 and then the station number. It is turned off by dialing #3 and the station number. I was wanting to have Asterisk (or Comedian mail) set the message lamp in the Comdial system when a new message arrives for a user, and extinguish the lamp

Thanks everyone for the input. After pricing everything we need out, it's not worth trying to get our old system to work, so I've pitched ditching everything and starting over. I'm very excited and hoping they'll go for it. Regardless, I'm going to throw a box together for my house, we have no home phone (just cell phones) so this'll be a great way of testing. All

Anyone see an issue on asterisk 1.2 that it will not accept the invite from a Cisco gateway. If I turn off voice service voip signaling forward unconditional then Asterisk accepts the call but without cname. Below is a trace. Any help is appreciated. Thanks John Bittner Simlab.net voippbx01*CLI> <-- SIP read from 216.86.35.24:63549: INVITE sip:9734333001 at 69.60.198.130:5060 SIP/2.0

>But currently, I only have one ethernet jack per office. Routing >another 60 or so ports would add a very substantial expense in both >cabling and backbone expansion (what category ethernet is required, >BTW?). Most decent phones have an ethernet passthrough (2 port) so you can plug in your PC. As long as your LAN is decent (Cat5 100baseT switched) the overhead using VoIP is