Re: [asterisk-users] Asterisk 1.2.18 problem

asterisk (does /var/run/asterisk.ctl exist?)". This connects to an already running asterisk system. You first have to start asterisk, type this: asterisk -vc Or I put the script in /etc/init.d and tell the system to start it on boot. Then you can do asterisk -r __

Re: [asterisk-users] Conference room as Music on Hold

See: http://www.voip-info.org/wiki/index.php?page=Asterisk+config+musiconhold.conf Look for: Using a sound card as the source. You can then plug in a radio or what ever. shadowym wrote: Here is what I am trying to do. I have a SIP soft phone running on a PC that is streaming a local radi

Re: [asterisk-users] Voicemails randomly not deleting in 1.2.9.1 ??

Anthony Davis wrote: We updated our systems to 1.2.9.1 (from 1.2.4) about 3 weeks ago. About every other day since then, a user will complain that they deleted a voicemail, but that Asterisk continues to tell their phone that they have new messages. They also can no longer delete these m

[asterisk-users] Asterisk and NFS

Any one have an idea why asterisk would ignore a callout file that was dropped in via nfs The permossions are the same as a file dropped in locally that works fine. Kyle -- CONFIDENTIALITY NOTICE: This message, including any attachments, is for the sole use of the intended recipient(s) and ma

Re: [Asterisk-Users] RRMEMORY / Queues Not Working Right

Shayne wrote: I'm currently having the same issue. If anyone has a solution to this it would be greatly appreciated. I have a room with 70+ agents logged and and am not seeing this.. One thing to keep in mind is if you do a reload it will reset the queue and start over Kyle -- CONFIDENTIALIT

[Asterisk-Users] Asterish Guru needed in Phoenix ASAP

Need 2 full-time Asterisk Guru's needed in Phoenix area right away. Knowledge in some of following also needed: Php Perl MySQL PostGres C++ Visual Basic HTML Photo Shop Email resume off list - Will interview this week. Kyle -- CONFIDENTIALITY NOTICE: This message, inc

Re: [Asterisk-Users] Good VoIP providers that support Asterisk PBX's

www.interast.com Kyle andrew matthews wrote: sorry its really http://connect.voicepulse.com/ On 2/14/06, *andrew matthews* <[EMAIL PROTECTED] > wrote: http://connect.voicepulse.net They support astrisk, with iax2 :) On 2/14/06, *Jim Robinson* < [EM

Re: [Asterisk-Users] Solution for 1 time blast of 200, 000 recorded calls

Here in Arizona I have gotten recorded calls from politicians. There are apps to allow for this feature. If you want to know more contact me offline. Kyle ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUB

Re: [Asterisk-Users] Softphone and 911

Softphone or hard phone doesnt matter if the service provider has the right connections to provide E911 service. We are setting up E911 compliance right now with our service. Its not as easy as just updating the address, it take time, its not instant. Kyle Matt wrote: Greetings to all, Can a

Re: [Asterisk-Users] MOH Server

I setup Slimserver to stream online radio stations to asterisk. Not all online stations work, but once you find a couple good ones you just stick with it. Kyle Douglas Garstang wrote: Has anyone managed to set up a moh server for Asterisk? Reason would be to offload processing off the aster

[Asterisk-Users] Quadra software - Changing to Opensource

For anybody interested: I have updated the site stating that all our software will be changing to open source effective Feb 1st 2006. If anyone has any questions please feel free to email me directly. All the apps have been writting in Visual Basic and will be available for anyone to download

Re: [Asterisk-Users] Asterisk Crashing (high load issues)

Why would you have all those modules loaded on an asterisk server?? Do you *REALLY* even have a PCMCIA slot on your server? Do you need USB? Or parallel port? Do you use IPv6 with asterisk (not supported AFAIK)?? even bluetooth and stuff is running! I know I need to remove alot of things, an

Re: [Asterisk-Users] Asterisk Crashing (high load issues)

redundancy and less total downtime if one server goes down. We noticed the same kind of limitations you are and now do a max of 40 agents per server, and when we need more capacity we just add another server. MATT--- On 11/10/05, Kyle Hagan <[EMAIL PROTECTED]> wrote: Kyle Hagan wrote:

Re: [Asterisk-Users] Asterisk Crashing (high load issues)

Kyle Hagan wrote: We purchased a new Dual Xeon 3ghz, 2gb ram to upgrade our 3ghz Pentium 1gb ram, that has been having load issues due to our growing company. We are having problems... We use a predictive dialer that we custom programmed in perl. It basically drops, moves, files into the

[Asterisk-Users] Asterisk Crashing (high load issues)

We purchased a new Dual Xeon 3ghz, 2gb ram to upgrade our 3ghz Pentium 1gb ram, that has been having load issues due to our growing company. We are having problems... We use a predictive dialer that we custom programmed in perl. It basically drops, moves, files into the callout directory and u

Re: [Asterisk-Users] Time based call direction

Adam Moffett wrote: include => atlunchcontext|11:00-11:59|mon-fri|* include => notatlunchcontext|09:00-10:59|mon-fri|* include => notatlunchcontext|12:00-18:00|mon-fri|* include => afterhourscontext|18:01--8:59|mon-fri|* I wasn't aware that include allowed a time qualifier. Does that mean

Re: [Asterisk-Users] Time based call direction

Rene Nelson wrote: I would like to manipulate phone call direction to voicemail for lunch, after hours etc, but am unsure how to do this. Could someone point me to a howto or quickly explain the concept? include => atlunchcontext|11:00-11:59|mon-fri|* include => notatlunchcontext|09:00-10:59

Re: [Asterisk-Users] Queue Login Out Question

ded for your situation. --johann Kyle Hagan wrote: We have 60+ members loged into the queue and talking to 5-10k people a day. I need a better way to track them loggin in and out. The queue_log gets really big fast. And has data we dont need. Is there anyother way to track when someone loge

[Asterisk-Users] Zapbarge feature available?

We would like to beable to listen in and interact with the person in a queue, talk to our agent and NOT have the other person be able to hear us. Is there a way to do this? Kyle ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Us

[Asterisk-Users] Queue Login Out Question

We have 60+ members loged into the queue and talking to 5-10k people a day. I need a better way to track them loggin in and out. The queue_log gets really big fast. And has data we dont need. Is there anyother way to track when someone loges in and out. I can write to a different file when th

Re: [Asterisk-Users] ZAP bchan and dchan HELP!!

ilto:[EMAIL PROTECTED] On Behalf Of Kyle Hagan Sent: Wednesday, August 10, 2005 4:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] ZAP bchan and dchan HELP!! They want to setup a 1 primary and 1 backup dchannel for 4 T1's No can you give me an exam

Re: [Asterisk-Users] ZAP bchan and dchan HELP!!

---Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kyle Hagan Sent: Wednesday, August 10, 2005 4:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] ZAP bchan and dchan HELP!! We have install a DS3 with 28 DS1's we

[Asterisk-Users] ZAP bchan and dchan HELP!!

We have install a DS3 with 28 DS1's we have an Adtran MUX breaking out the DS1's, we are trying to setup the system with 2 dchannels for each 4 DS1's. Everything looks fine when modprobe zaptel and wct4xxp and ztcfg -vv but when I asterisk asterisk it says: Aug 10 16:33:32 ERROR[8954]: cha

Re: [Asterisk-Users] what is the problem with gmail and the list.

I got 23 emails since friday. And im NOT using gmail. Kyle Time Bandit wrote: I have not been receiving mail from the list 29th July, what is the problem with gmail and the list. No problem here. Check you Spam folder, and if you find email there from this list, select them all and cli

[Asterisk-Users] Setting Callerid in callout file problem

Here is the perl that writes the callout file. And it works ok, mostly. But the callerid is NOT being set on the call out. If I set it to dial through the dial plan: Channel: Local/[EMAIL PROTECTED] i can set it there. But I would think it should be set already in the callout file shown. What

[Asterisk-Users] Errors Question

Today we have been having some problems with the dchannel of out T1's. I was wondering if there is a way for asterisk to send out an email or page whenan error occurs. Not I know errors happen quite offen for many reasons, but I would like an email sent when there is a TI problem, or specefic

Re: [Asterisk-Users] PHPAGI problems

Try exten => 8661231234,1,agi,/some/directory/dtmf.php Kyle Hariharan Gopalan wrote: Hi Here is part of my extensions.conf exten => 8661231234,1,agi,dtmf.php When I dial this number, this is what I see in my asterisk console: -- Accepting AUTHENTICATED call from 198.22.67.70: > reque

Re: [Asterisk-Users] VoipSupply.com

Have heard great things about IP600. I hope IP300 is also as great as IP600. Thanks for your replies. -Original Message- From: Kyle Hagan [mailto:[EMAIL PROTECTED] Sent: Tuesday, May 17, 2005 5:38 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-

Re: [Asterisk-Users] VoipSupply.com

Manjit Riat wrote: I am going to buy some IP phones from them but I sent them an email couple of weeks ago and got no reply. Has anyone ordered anything from them? Any other places that I can buy from? Sorry if it’s a wrong post. I have used them many times and had no problems. What are you look

Re: [Asterisk-Users] Agent Login/Logout

Question is: how do they log out? I haven't been able to find a Logout command for the dial plan. Did I miss something? Do the same as when you log in but when it asks for the extension press #. OR ---queues.conf member => Agent/9001 member => Agent/9002 ---agents.conf agent => 9001,,tsr9001 age

Re: [Asterisk-Users] DS3 (T3) Card for Asterisk?

They said availability maybe end of summer... I need on now. Anyone know of a compatable card? Kyle Dean Collins wrote: DS3 about to be released on the digium site. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Kyle Hagan Sent

Re: [Asterisk-Users] DS3 (T3) Card for Asterisk?

Dean Collins wrote: DS3 about to be released on the digium site. Is there a DS# (T3) card that will work with asterisk? OR a card that supports more that 4 T1's per card? Kyle Any idea when it will be ready? Im gonna need on in less than a month... Kyle _

[Asterisk-Users] DS3 (T3) Card for Asterisk?

Is there a DS# (T3) card that will work with asterisk? OR a card that supports more that 4 T1's per card? Kyle ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or upda

[Asterisk-Users] Asterisk PRI problems (Crashing when full)

We have been running into problems here, we have 2 PRI's when they fillup, All channels in use, and we dial more calls asterisk becomes unstable and crashes alot. We are currently on: Asterisk CVS-v1-0-10/26/04-09:35:31 built by [EMAIL PROTECTED] on a i686 running Linux I know I need to upgrad

Re: [Asterisk-Users] Recording Queue agents

We use the AgentCallBackLogin option, that will work for you too. Kyle David John Walsh wrote: Hi all We have a psudo-switchboard in the offices, with receptionists who login into the switchboard queue, all is well. I have been asked to look into the possibilty of recording all the agents conversat

Re: [Asterisk-Users] Monitor with Asterisk@Home

You have to put the monitor after the person presses their selection. This is how ours is: exten => s,1,answer exten => s,2,SetCIDName('PMG') exten => s,3,SetVar(company=PMG) exten => s,4,Wait(1) exten => s,5,DigitTimeout,5 exten => s,6,ResponseTimeout,40 exten => s,7,Background(/var/lib/asterisk/s

Re: [Asterisk-Users] Call Center software opensource or commercial

Is that with channels recording ? ;) ___ We are running 40-50 simultanious calls at the call center here, and recording everycall in and out, with no problems On a Pentium 3ghz with 1gig ram. Kyle ___

Re: [Asterisk-Users] Options for Attendant Console.

www.quadrasoftware.com Kyle Will McCown wrote: We've been playing with Asterisk with an eye towards possibly replacing or augmenting our existing PBX serving about over 600 phones (and needing to expand). The one missing bit that I can't find any mention of is an Attendant Console. Are there any

Re: [Asterisk-Users] Linphone / Kphone

Darren Ellis wrote: Hi, I have * working with X-Lite and Sipura adapters, but I have one person who is linux based, and is trying to use Linphone and Kphone. His end works, but I get very bad echo on my end. Have any of you folks been able to get linux based soft phones working well with *? I

Re: [Asterisk-Users] Who makes these phones?

I have 3 of the Black ones. I think the are junk. They work, and I actually found a manual online for it. I ran into a weird problem last week. After I did a Reset to Factory. All the phones were getting th same IP address from the DHCP server, I found that the MAC address on the phones were th

Re: [Asterisk-Users] cisco 7960 image

It here: http://www.aarnet.edu.au/events/conferences/2004/apan-questnet/sipworkshop/uas/cisco7960/cisco7960.html Not that its LEGAL. But you would be able to test it. Kyle Jose Cruz (Branders IT) wrote: Thank you for that information. But I just wish to test it first, is there like a test firmware

[Asterisk-Users] VICI dialer help...

I've got the VICI predictive dialer runnning over IAXs to another asterisk server. It dials fine. I can make phone calls manually with no problem. When VICI dials a new number it rings the other end once and I get the error: Jan 26 13:53:10 NOTICE[10206]: Dropping incompatible voice frame on I

Re: [Asterisk-Users] Wildcard TE405P and TDM400 - TDM not working

add ;empty=49-72 ;empty=73-96 fxsls=49-50 change to fxsls=97-98 switchtype = 5ess signalling = pri_cpe group = 2 channel => 1-23 channel => 25-47 signalling = fxs_ls context= internal group = 3 channel => 49-50 change to channel => 97-98 That fixed the problem thanks. I ass

Re: [Asterisk-Users] 7960 SIP image

You can get the images here: http://www.aarnet.edu.au/events/conferences/2004/apan-questnet/sipworkshop/uas/cisco7960/cisco7960.html But its not the LEGAL way to get them. Since Cisco wants you to pay for it. Kyle Steve Beaumont wrote: You really need to get hold of the SIP image and load it in t

[Asterisk-Users] Wildcard TE405P and TDM400 - TDM not working

Im having a problem getting the 2 cards working together. I added the TDM with 3 FXO and 1 FXS board. The TE405p is working fine. THe TDM registers and identifies the FXO's and FXS but cant get them to go off hook. I hope someone can help. Asterisk seems to THINK its dialing it sends manager stat

[Asterisk-Users] Call Parking

I am using a manager app to do redirects, when I redirect to 700, for parking, the person on the other end hears the number its parked on. How do I stop this? Kyle ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/m

Re: [Asterisk-Users] Error during installation

Robert Kerry wrote: Hi, We're getting an error when installing Asterisk on a Trustix Linux server. I'm guessing that the error is related to SSL, although OpenSSL 0.9.7e is installed on the server. Any thought's on this matter would be appreciated. gcc -g -o asterisk -Wl,-E io.o sched.o logger.o

Re: [Asterisk-Users] Cisco SIP Firmware HERE!!!!

I never said it wasnt illegal, Someone asked about it and I found a site with it so I sent it. If someone is going to use it, it would be in their best interest to purchase it legally. If you know someone with an account with cisco (ie some Cisco certs) you can get them for free. Kyle Jason p w

[Asterisk-Users] Cisco SIP Firmware HERE!!!!

Alot of people have needed the SIP firmware for the Cisco 79xx phone. I found a link for them.. http://www.aarnet.edu.au/events/conferences/2004/apan-questnet/sipworkshop/uas/cisco7960/cisco7960.html Kyle ___ Asterisk-Users mailing list [EMAIL PROTECTED

[Asterisk-Users] Meetme and audio recording/playback

I am wondering if its possible to play an audio file in a meet me room? I am also recording the conversations (both sides) of the call. I dont have to record during the audio play back. Currently what we have to do is Call the person, start recording, talk with them, stop recording, transfer t

[Asterisk-Users] Auto dialout

Im trying to use the auto-dial out in asterisk and having problems Here is my 1.call file that I move into the outgoing directory: # # Create the call on group 2 dial lines and set up # some re-try timers # Channel: Zap/g2/4807480034 MaxRetries: 2 RetryTime: 60 WaitTime: 30 # # Assuming that y

Re: [Asterisk-Users] system errors

:18.27 asterisk 13838 root 9 0 7404 7332 3644 R 0.3 0.8 0:00.14 asterisk its showing 50% CPU x 2. when I strace -p 10744 or strace -p 23199 thats when I get the error. Kyle Steven Critchfield wrote: On Thu, 2004-11-04 at 10:11 -0700, kyle Hagan wrote: Im getting the following error

Re: [Asterisk-Users] system errors

Im not running strace constantly, I just run it to see whats going on. And that error it coming in on the Asterisk PID at a rate of about 10 per second, from what I can see. Kyle Steven Critchfield wrote: On Thu, 2004-11-04 at 10:11 -0700, kyle Hagan wrote: Im getting the following error

[Asterisk-Users] system errors

Im getting the following error when I do an strace -p # for asterisk . Its taking alot of resources. ioctl(0, SNDCTL_TMR_TIMEBASE or TCGETS, 0xb290) = -1 EIO (Input/output error) write(1, "voip1*CLI> ", 11) = -1 EIO (Input/output error) I checked the WIKI and the list and f

[Asterisk-Users] Parked Call Viewer Application - Freeware

I have written an application (win32) to view parked calls. The reason i wrote the code is because my GS doesnt support the system reading the parked extension back to me after I park it. This will allow you to view where the call is parked, the caller id and when it was parked there. You can

[Asterisk-Users] CREATIVE Critisim wanted for new application

We have been developing some new software for *. We are getting ready to finish some new applications (win32 for now) and would like your input on the design (Look) and feature set descriptions. And what you feel would be relevant additions. The applications we would like you to review are the A

Re: [Asterisk-Users] Zultys Zip 4x4

Mike Roberts wrote: Is anyone successfully using one of these with Asterisk? I cannot get the phone to register, this message keeps coming up on the Asterisk console: Jul 29 14:11:39 NOTICE[1125350192]: chan_sip.c:7323 handle_request: Registration from '"000BEA801CA6" ' failed for '204.194.36.138'

Re: [Asterisk-Users] Current echo status?

Scott Laird wrote: I've been following the list for months, and I have a working Asterisk setup, but it'd be *really* useful to me at this point if someone could summarize when Asterisk has echo problems and when it doesn't. For instance, I usually hear a far-end echo when talking on my 7940,

[Asterisk-Users] Call Queues help

I've got the call cuing all setup and working, but im trying to get the Callswaiting,you are caller #, etc, and its not working. I have the following inthere as stated: queue-youarenext = "queue-youarenext" ("You are now first in line.") queue-thereare = "queue-thereare" ("There are") queue-cal

[Asterisk-Users] Manager help

Can someone give me an example of what comes from the manager when a call comes in on a PRI./T1 when a call comes in? On the x100p i ger the following: Event: Newchannel Channel: Zap/1-1 - I need what would be populated in here Would it be same? ZAP/(card)-(1 to 100, for 100 block

Re: [Asterisk-Users] Zultys 4x4 or 4x5 ip phones?

Michael Graves wrote: Does anyone on-list use the Zultys 4x4 or 4x5 ip phones? I'd like to hear some opinion before I buy a few. I'm especially interested in the PSTN interface on the 4x5. Does it relay to * for VM when an incomming call is not answered by the phone? Thanks, Michael I have used

[Asterisk-Users] Caller ID software v1.1 available

I put v1.1 on the site today. Changes: # Fixed problem with skin, v1.1 # Added to Taskbar Icon with callerid, v1.1 # Added Capability to monitor incoming lines to asterisk server, v1.1 # Added status light, flashes red if not connected, v1.1 I know some of you are waiting for the receptionist progr

[Asterisk-Users] Asterisk info needed for new application development.

extension? (typically) 7) Do you connect to other service providers (FWD, NuFone, Etc..) if so whe do you use? I appreciate you taking the time to answer these questions. Please reply to: [EMAIL PROTECTED] Kyle Hagan www.quadrasoftware.com ___ Asterisk

[Asterisk-Users] Asterisk Caller ID Application (win32)

I have updated the Caller ID Application to v9.5 beta. Fixed Caller ID when monitoring Zap Channel. In config device to manage: ZAP/#-# Fixed Come to top on Call, was staing perm on top after 1st call Added Debug Screen visit: http://www.quadrasoftware.com to download the latest beta. Be sure to

Re: [Asterisk-Users] IAXTel Help

Kyle Hagan wrote: I've searched WIKI and Archives but nothing. Im getting: -- Called username:[EMAIL PROTECTED]/[EMAIL PROTECTED] Jun 21 17:04:12 WARNING[1158883520]: chan_iax2.c:5097 socket_read: Call rejected by 69.73.19.178: Unable to negotiate codec -- Hungup 'IAX2[Iaxtel]/8'

[Asterisk-Users] IAXTel Help

I've searched WIKI and Archives but nothing. Im getting: -- Called username:[EMAIL PROTECTED]/[EMAIL PROTECTED] Jun 21 17:04:12 WARNING[1158883520]: chan_iax2.c:5097 socket_read: Call rejected by 69.73.19.178: Unable to negotiate codec -- Hungup 'IAX2[Iaxtel]/8' == No one is available to answe

Re: [Asterisk-Users] Re: Grandstreams randomly go busy with Asterisk?

Brian Buhrow wrote: Hello. I've seen this behavior. What happens is that the Grandstreams forget to continue registering with Asterisk after a while. I bet when you find this happening, that sip show peers doesn't show ext/ext ip address for the one that isn't working. You can work around the

Re: [Asterisk-Users] Grandstreams randomly go busy with Asterisk?

I have one of 2 doing the same thing. Kyle Robert Withrow wrote: I've searched the lists but I didn't find anything exactly like this. I have two Grandstream BT101 phones connected to an Asterisk. Periodically, for reasons that I can't determine, one or the other (or both) of the BT101s decide(s)

[Asterisk-Users] Asterisk Receptionist - Lite - CallerID Source code

We have been getting email asking if your will be available. We are considering publishing source. But havnt made a decision either way. It can be swayed to publishing it if we can get donations on the web site to cover our time. If some one would like to donate a Wildcard TE405P and/or TDM400P w/

[Asterisk-Users] Asterisk CallerID app (win32)

I just uploaded a beta CallerID program. It talks through the Asterisk Manager . Pretty self expanatory for setup and configure. Please Let me know what you think. http://www.easyhomenetworks.com/AstRec/ If you feel our programs are useful please make a donation. We dont plan on SELLING these prog

[Asterisk-Users] Asterisk Receptionist Lite version

 We are reworking the Asterisk Receptionist at the moment expect an update in a few days.   We have a Lite version available for testing. Its a scaled down version, 20 extensions max.   Supports:   Call transfers Call Origionate Show busy extensions   Please download and let me know if any bug

Re: [Asterisk-Users] RE RE: Asterisk Receptionist manager program.

mail context [great_customer], and another 104 in voicemail context [really_great_customer]. Jeremy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kyle Hagan Sent: Thursday, June 03, 2004 3:46 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk Recept

Re: [Asterisk-Users] Asterisk Receptionist manager program.

so being defined. I don't know what to set in my prefs.conf file for extension... ?? -Brett Message: 2 Date: Thu, 03 Jun 2004 09:27:44 -0700 From: Kyle Hagan <[EMAIL PROTECTED]> Organization: Nuvo Technologies To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk Receptionist manag

Re: [Asterisk-Users] Asterisk Receptionist manager program.

ted ---stop--- ---start--- Response: Error ---stop--- ---start--- ActionID: 1 ---stop--- ---start--- Message: Permission denied ---stop--- --__--__-- Message: 2 Date: Thu, 03 Jun 2004 09:27:44 -0700 From: Kyle Hagan <[EMAIL PROTECTED]> O

[Asterisk-Users] Call Originate from Manager application.

When I to Call Originate from the manager app. It works fine except caller ID comes in as to both phones. Can I set this? Kyle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or u

Re: [Asterisk-Users] Asterisk Receptionist manager program.

-start--- Message: Permission denied ---stop--- Above shows there was a prblem checking the voice mail. To display New and Old Messages. --__--__-- Message: 2 Date: Thu, 03 Jun 2004 09:27:44 -0700 From: Kyle Hagan <[EMAIL PROTECTED]> Organization: Nuvo Technologies To: [EMAIL PR

Re: [Asterisk-Users] Asterisk Receptionist manager program.

I put a new version up last night. Caller ID shows up on the buttons. This time IAX is fixed. Works at home and at work through FWD. http://www.easyhomenetworks.com/AstRec/ Has anyone had anyother bugs popup other than the IAX problem? Some people are asking why the screen shot has more buttons t

Re: [Asterisk-Users] Asterisk Receptionist manager program.

I put an update on the website. It fixes the IAX calls crashing the program. ANd added Voice Mail checking. It will tell you how many new and old voice mails are in your box. http://www.easyhomenetworks.com/AstRec/ Kyle John Fraizer wrote: Kyle Hagan wrote: Ok I have a testing version

[Asterisk-Users] Asterisk Receptionist

 Updated asterisk Receptionist.   Shouldnt have problems with IAX calls causing an error.     http://www.easyhomenetworks.com/AstRec/   Kyle

Re: [Asterisk-Users] Asterisk Receptionist manager program.

We are planning to have this work with other OS's. But for now starting with windows. (gotta start somewhere.) Kyle Stefan de Konink wrote: First feedback... Bit (very) disappointed this is a Win32 program :( I had hoped to find some crossplatform sources, but probably many other people like the

Re: [Asterisk-Users] Asterisk Receptionist manager program.

for Asterisk Receptionist soon at www.easyhomenetworks.net/astrec with updates. Kyle Kyle Hagan wrote: To all that are interested Im hoping to have a beta (or alpha) available for download today. I will email the list as to where it can be downloaded. Kyle Brian D'Arcy wrote: Kyle, I also

Re: [Asterisk-Users] Asterisk Receptionist manager program.

uture. =) Thanks, Brian D'Arcy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kyle Hagan Sent: Friday, May 28, 2004 9:33 AM To: Asterisk Subject: [Asterisk-Users] Asterisk Receptionist manager program. We are writing a program using the manager for * for our re

[Asterisk-Users] Asterisk Receptionist manager program.

We are writing a program using the manager for * for our receptionist to use once the system go live. If anyone is interested in helping us with testing please let me know. We are designing it for a touch screen monitor for her to do transfers, see whose on the phone and a few other features. I

Re: [Asterisk-Users] Grandstream message light button

The flashing display IS the message waiting light, from what I can tell. Kyle Dean Collins wrote: I have finally set correctly the Message Wait Indicator feature on my Grandstream (I was using the wrong context – I thought the documentation referred to the [EMAIL PROTECTED] extension context not

Re: [Asterisk-Users] Problem with SIP softphone

Ok that fixed it. But why all of a sudden did it start doing this after I updated? Anyidea? It had been working fine for a few months. Kyle Philipp von Klitzing wrote: Hi! On my SIP softphone, when I stop speaking the audio stops. So if im not talking I cant hear the other person. FAQ! X

[Asterisk-Users] Problem with SIP softphone

Having a weird problem after I updated the other day. On my SIP softphone, when I stop speaking the audio stops. So if im not talking I cant hear the other person. Kyle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/li

Re: [Asterisk-Users] Softphone lag

I always get that with a softphone, but not with a hardphone. Grandstream BT100 is only $70 so Im gonna get those for most of the people. And a few higher end phones for the execs. Kyle Navnit Chachan wrote: Hi, IF i use a sip softphone or a iax softphone with asterisk, i get a lag of about 1 s

Re: [Asterisk-Users] Mystery SIP channels

I had the same thing come up on mine when I was having codec issues with one of my phones. Kyle Nik Martin wrote: What address is that? Is it a phone (or address of a PC with a softphone?) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Dollof

Re: [Asterisk-Users] Asterisk on Compact PCI platform

I had considered trying this but from what I have read flash drive have a 1million read write life expetancy? If you were to use one of these as your harddrive would it not wear out pretty quick? Or am I wrong? Kyle Jay Milk wrote: Since this is related... Does anyone have Asterisk working on a

Re: [Asterisk-Users] MPG123 errors

I had same issue until i used mpg123 version 0.59r Anon wrote: On Friday 07 May 2004 10:57 pm, Kyle Hagan wrote: When I put someone on hold audio doesnt play and i get mpg123: unknown option "mono", Any ideas. I searched wifi and archives. Are the music file you are playing mon

Re: [Asterisk-Users] call announce?

I use the Zultys 4x4 and it will allow me to announce before it transfers. But he GS BT-100 just transfers it right away. Kyle brian k. west wrote: No way to do that without writing your own custom application to do it. bkw - Original Message - From: "Gavin Hollinger" <[EMAIL PROTECTED]

Re: [Asterisk-Users] How to Echo extension number to caller?

Try: exten => *53,1,SayDigits(${CALLERIDNUM}) Kyle Paul Mahler wrote: I need to dial an extension that tells me what extension I'm dialing from. I'm running a bunch of analog phones off a channel bank to * over a T1. I have the following in extensions.conf. exten => 98,1,SayDigits(${EXTEN})

Re: [Asterisk-Users] What's in ${EXTEN} ? Why does voicemail prompt for an extension?

Here is how mine is: exten => *55,1,Voicemailmain(${CALLERIDNUM}) Mine works. Kyle Paul Mahler wrote: Why does voicemail prompt me for an extension instead of just asking my password? [voice-mail] exten => 99,1,VoicemailMain([EMAIL PROTECTED]) exten => 99,2,Hangup Paul Mahler [EMAIL PROTECTED

Re: [Asterisk-Users] IAX Freeworld

EMAIL PROTECTED] disallow=all allow=ulaw [iaxfwd] type=user context=fromiaxfwd deny=0.0.0.0/0.0.0.0 permit=65.39.205.0/255.255.255.0 my extentions.conf has the following for IAX: [home] exten => _6.,1,SetCallerId,"Kyle Hagan" < 410769 > exten => _6.,2,Dial(IAX2/410769:[EMAIL PROTECTED

Re: [Asterisk-Users] IAX Freeworld

James H. Cloos Jr. wrote: "Kyle" == Kyle Hagan <[EMAIL PROTECTED]> writes: Kyle> In coming works fine from FreeWorld via IAX. But when Kyle> Dialing out i get [an error] ... Does iax2.fwdnet.com even support iax2=>fwd? I thought it was just for registering

[Asterisk-Users] IAX Freeworld

I have looked all over the site(s) for help. But heres the problem. Im missing something. In coming works fine from FreeWorld via IAX. But when Dialing out i get: May 13 13:42:01 WARNING[1150495040]: chan_iax2.c:5256 socket_read: I don't know how to authenticate iaxtel to 65.39.205.121 my IAX.

Re: [Asterisk-Users] SNOM 200

Tim Sailer wrote: On Wed, May 12, 2004 at 01:12:10PM -0500, Greg Scasny wrote: Zultys 4x4 had been the best phone we have tested so far though, absolutely wonderful quality, speakerphone rocks, headset sounds great, MWI is perfect. Whom did you buy from? What codecs are supported? Tim

Re: [Asterisk-Users] Asterisk & Rhetorical Systems

So this integrates with asterisk? where do I find app_cepstral? Kyle - Original Message - From: "Ben Merrills" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Monday, May 10, 2004 8:45 AM Subject: RE: [Asterisk-Users] Asterisk & Rhetorical Systems I took a look at their site and pla

[Asterisk-Users] MPG123 errors

When I put someone on hold audio doesnt play and i get   mpg123: unknown option "mono",   Any ideas. I searched wifi and archives.   Kyle

Re: [Asterisk-Users] No Audio from Hard Phone to SIP

The manufacturer think its a bad phone. So Im getting another one today. They said thay have gotten the Zultys phones to work with asterisk with no problems. Will let everyone know. Zultys sait ULAW was the most common but did not state why most use it. Kyle - Original Message - From

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