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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

Group, Just want to share with the group my recent findings regarding CODECs/Vocoders and the effect it has had on voice quality and the intermittent noise and breakup problem I have which I mentioned in a previous emailing with the u-law CODEC. Calls again are placed through a SIP phone to a TDM400P to the PSTN. A good reference on the reasoning behind the selection of a CODEC was found in the

> -----Original Message----- > From: Dale [mailto:dale-list@lightwavetech.com] > Sent: Tuesday, January 18, 2005 1:45 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] QoS tagging - can Asterisk do this, > if not,what do you recommend? > > So my question is, does Asterisk offer the > ability to mark the voice data with the

I am just an Asterisk newbie doing a test install. I am using 2 X-Lite clients and have configured them according to the wiki on voip-info. A warning is still displayed on the Asterisk server console saying that I should disable RFC3389 on the client, even after I changed the Transmit Silence to yes. I am able to connect and call the other client, but when I do no audio is being transmitted by

Hello All, I have Polycom IP 500 phones which I would like to have message waiting indicators on. So far, I have my system running well but the problem I am seeing is that MWI doesn't seem to tell my phone that it should display a MWI state. The light does not show when you have message nor is there any indicator on the text lines of a message waiting. The wiki doesn't cover this

...t advertise= > d -=20 > and neither are the config parameters needed to adjust it / turn it off. > > I'll check with Uniden. > Thanks > Ran > > --__--__-- > > Message: 5 > Date: Mon, 19 Jul 2004 19:34:31 -0400 > From: "Robert Jackson" <RobertJ@promedicalinc.com> > To: <asterisk-users@lists.digium.com> > Subject: [Asterisk-Users] MWI - Config Stupidity or Notify Issues? > Reply-To: asterisk-users@lists.digium.com > > I am having a problem with the message waiting indicator. We are > currently using the ast_data modules for...

I am planning an * install at my business. It will be replacing an existing Toshiba system (I think it is a 424dk). I was wondering if anyone knows of a way for me to use my existing Toshiba phones to connect to *. I would rather not have to spend the $15,000 to replace all of my phones, but I can't find any other way to do it. Your help is greatly appreciated. Thanks, Robert Jackson

Hey all, I'm sorry to bother you with something so trivial, but I seem to be having an issue with the Asterisk::AGI module. I am a relative newbie with Perl so it could be a stupid syntax mistake that I missed. It seems when I try to execute either the stream_file or the get_data subs nothing is actually done. It doesn't seem to stream the files, but on the console it says it played the

I am having a problem with the message waiting indicator. We are currently using the ast_data modules for both our sip configuration and our voicemail configuration. In the mailbox field I have tried using both mailboxnumber@context and simply mailboxnumber. Yet so far I am still not getting a MWI on my 7905's or on my 7960's. My assumption would be that I am still missing something,

After just having update to the latest CVS I am getting the following message when I call VoicemailMain(): -- Executing VoiceMailMain("SIP/2302-1a12", "") in new stack -- Playing 'vm-login' (language 'en') -- Playing 'vm-password' (language 'en') -- Playing 'vm-youhave' (language 'en') -- Playing

I am not sure what is going on, but * is restarting itself every time a user hangs up after calling to check their voicemail. I am running CVS-HEAD-07/26/04-22:14:48, and this problem just started happening after I updated last night. I am rolling back to CVS-7/14/2004 so that we can keep working, but we need to address the voicemail issue. I will open a bug if this is not just something on my

> -----Original Message----- > From: Ryan Parlee [mailto:listbox@jesca.com] > Sent: Tuesday, August 10, 2004 6:10 PM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] CVS version tags > > > > Can someone please tell me which version tags I should be > using when checking out with cvs. Right now I have two > directory trees: > > 1)

> -----Original Message----- > From: Kevin [mailto:Asterisk@gtcus.com] > Sent: Saturday, August 21, 2004 5:16 PM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Queue Monitor > > > I understand that putting monitor-format in the queues.conf > file will start monitor recording of an active queue call. > Is there a way to automatically do the

I have heard about the ICD project to make a better Call center solution for asterisk, and I just noticed it on cvs. Is it a viable alternative or does it need work? Either way I am definitely interested as we are not quite happy with the current setup. Thanks, Robert Jackson

> -----Original Message----- > From: list@ipmotel.net [mailto:list@ipmotel.net] > Sent: Friday, October 01, 2004 12:35 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] OT: Toll Free > > > Can anyone recommend a good Toll Free provider? I'm looking > for 1 T-1, 2 > Toll Free numbers via VoIP. > NuFone provides us

I am looking at writing a script to basically perform the same as the iaxping.exe that has been referenced on the wiki, but I would rather it not be in vb6. is the source code for the existing vb6 app GPL'd so that I could base a perl script off of it? Or do I need to start from scratch? Also, if anyone has some example code or has already undertaken this project please let me know. I

> -----Original Message----- > From: Adam Fineberg [mailto:fineberg@levanta.com] > Sent: Saturday, January 08, 2005 2:38 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] New 'n' priority > > > Christopher L. Wade wrote: > > > 'n' and 's' as well as labels only work in CVS HEAD, its >

> -----Original Message----- > From: Robert Rozman [mailto:rozman@fri.uni-lj.si] > Sent: Monday, January 17, 2005 7:45 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] Can I start recording channel in > the middle ofconversation ? > > > Hi, > > I'd kindly ask for simple example if this is possible ? > > Is

I have a couple of real quick questions for ya'll. I am trying to setup * in a test scenario to emulate my company's 20 agent ACD. 1) First, I am curious if anyone has any configs that they can throw up on the wiki or just send via e-mail. I have a bit of it pieced together and am able to make some very basic queued calls, but I am sure that there are much smarter people out there

I am trying to setup a couple of virtual pbx's off of my one may asterisk box. So far I have been able to segment most everything via the Dial plan. My only question/problem has to do with the # Transfer function. I had set up # Transfers prior to segmenting the dial plan, and I cannot remember how I was able to specify which context to use when the user presses #. I haven't been able