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Alright, folks. I just deployed * into full production at my office. We have around 50 7905's, 5 7940's, and a handful of soft clients. We run a call center with around 15 agents. I also have a queue set up for the receptionists so that they don't get bombarded with calls. Everything seems to be working with a very few minor glitches. I firmly believe that the few problems we are

-----BEGIN PGP SIGNED MESSAGE----- Does anyone have any working example GR-303 configurations for zaptel and zapata conf? The information available on the wiki as well as in the sample configurations just doesn't seem to be enough to bridge the gap for me. In Zaptel.conf, Do you set up a GR-303 circuit like a PRI with b and d channels or do you set fxo or fxs, ks signalling? How do you

I have setup iax2 between two servers without success. when I launch asterisk with the asterisk -vvvvvgcd command I see serveral wanings listed below. Is this why I cannot make connections?? My question is, how do I setup zaptel timming without any cards if possible? Does anyone have the steps? Thanks for any information. James [chan_iax2.so] => (Inter Asterisk eXchange (Ver 2)) Apr

Dear All, I have been running a successful and very stable call center PBX based on 0.7.1 release. I need to be on this release because of a number of features that I have complied from 3rd party patches, for the call center. I will not be able to upgrade to release 1 until the patches catch up and I have done the required testing. The system was very stable until two days ago. The changes made

I am trying to give users the option to query our accts. payable database by supplying their PO number. I able to write queries via perl->DBI->mysql but have no idea how to get * to do it from the IVR. Is this possible? Can anyone point me in the right direction for help or examples? Thanks, Richard --------------------------------- What are the most popular cars?

...> this point. I have yet to get this to work and I also don't think I am > receiving any SIMPLE messages ti show me that I have messages waiting. > > Do you get a message waiting indicator? > > W > > -----Original Message----- > From: Chris A. Icide [mailto:chris@netgeeks.net]=20 > Sent: Monday, July 19, 2004 3:03 PM > To: asterisk-users@lists.digium.com > Subject: RE: [Asterisk-Users] Polycom IP 500 Voicemail > > On 12:40 PM 7/19/2004, Wiley E. Siler wrote: >> My Polycom is on loan as a demo and I assume it is one of the first >> revisi...

...gt; will not be able to dial # into ivr's. > > Search on wiki for # transfer > > Regards > > > Jason > > --__--__-- > > Message: 4 > Date: Mon, 19 Jul 2004 08:26:32 -0700 > To: asterisk-users@lists.digium.com > From: "Chris A. Icide" <chris@netgeeks.net> > Subject: [Asterisk-Users] Channel banks, voicemail, and immediate=no > Reply-To: asterisk-users@lists.digium.com > > When using a channel bank for analog handsets, you have a couple options in > the way you handle transactions involving the analog handsets and origination....

Hello All, I have some Polycom IP 500 phones that I would like to have configured for direct dialing to our voice mail system. So far I have been unable to get the hard button labeled Voice Mail to connect to Asterisk without first passing through the message center prompts. I have followed all the Admin Guide instructions regarding the phones .cfg files and using

Before I go off and post a feature request on the bug tracker, I want to make sure I've not misgoogled or miswikkid and not found an existing capability. What I'm looking for is the ability to determine whether or not a queue has any queue handlers (active agents), and if it does not, bypass sending the caller to the queue and pass them on to a message or IVR system. Is there a

Can anyone describe the asterisk implementation of this any better than the sample config files do? from zapata.conf ; Trunk groups are used for NFAS or GR-303 connections. ; ; Group: Defines a trunk group. ; group => <trunkgroup>,<dchannel>[,<backup1>...] ; ; trunkgroup is the numerical trunk group to create ; dchannel is the zap channel which will

I have a system running CVS HEAD 6/30/2004. We've only been using it for PSTN to channel bank handsets, but have decided to add sip phones into the mix. Now I have quite a few systems running sip phones just fine as well as some running both sip and analog via channel banks or tdm cards. When we tried to set up some sip extensions (they are behind nats, we are using xten light, and have

I'm having a small issue with the oh323 implementation when it comes to codec selection. Version info: CVS Head 6/30/2004 OH323 0.6.3 OpenPhone for windows version 1.8.1 Asterisk is configured as a h323 endpoint which either terminates to the PSTN locally through a PRI or terminates the h323 call to an IAX provider remotely. Asterisk also has G729 licences installed. in oh323.conf we

Is anyone familiar with the Trollphone's LCR package? There is a field in the egress table labeled substitue. Placing a $1 there results in the correct dial extension being passed. However how is this field used to substitute replacement dial extensions... in other words as an example, lets say my EXTEN as passed to and back from the rate_engine is 01144123456789 and I want to have a

I've got an interesting problem. I just recently upgraded an asterisk server from a may 9th cvs to a 9/29 cvs. When I did the upgrade I was unable to place calls through the PRI. The calls would process fine, but the provider would reject the call and send me a cause 69 or 77 failure code "Unallocated (unassigned) number". Upon contacting the provider, they told me I was

From the Wiki: Presentation indicator (octet 3a) Bits 7 6 Meaning 0 0 Presentation allowed 0 1 Presentation restricted 1 0 Number not available due to interworking 1 1 Reserved Screening indicator (octet 3a) Bits 2 1 Meaning 0 0 User-provided, not screened 0 1 User-provided, verified and passed 1 0 User-provided, verified and failed 1 1 Network provided How do these bits fit into

I've got a system with 5 pri circuits configured into a CFAS group with a primary and secondary d channel. There are three TE410P cards in the system. The 5 circuit span are located as follows: circuit 1 on span 5 circuit 2 on span 1 circuit 3 on span 6 circuit 4 on span 2 circuit 5 on span 9 primary d chan is on chan 24 of span 5 (chan 120) secondary d chan is on chan 24 of span 1 (chan

There is a patch on Mantis (http://bugs.digium.com/view.php?id=4371) Which includes several features. 1. Support for central voicemail server(s) with remote server notification via IAX In other words, this patch allows you to configure an Asterisk server as a central voicemail server and to send out voicemail notification to remote Asterisk servers who can then pass the notification on to

I've got an interesting scenario where transfers while getting an invite seem to break. Here is the scenario: You have a receptionist who has a 6 line phone (in this case, a polycom ip600 - also tested with a Cisco 7960) the receptionist has all six lines available for use (in the case of the cisco I tried registering all lines as one number as well as registering multiple lines and

from the zaptel sample config: # "dacs" : The zaptel driver cross connects the channels starting at # the channel number listed at the end, after a colon # "dacsrbs" : The zaptel driver cross connects the channels starting at # the channel number listed at the end, after a colon and # also performs the DACSing of RBS bits dacs=1-24:48

Am I reading the data below incorrectly, or does it appear that even though I have the directive canreinvite=no set for the two asterisk boxes, they are trying to do a reinvite (which fails) anyway? Is this expected behaviour in this situation? If so, how can I prevent this? ---- Lots of output ---- Using CVS Head from 2005-08-28, I have two asterisk boxen, one (box A) has a sip ua (2608)