[asterisk-users] Fwd: Legacy TDM400

attempts on Debian 10 and the Debian repository version of Asterisk didn't get me very far. Any pointers would be appreciated. -Roy -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new

[asterisk-users] Asterisk does not listed to port 5060

know what I miss in this configuration. Best Regards, Roy. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http

[asterisk-users] Passing parameters to voiceglue.conf

dynamically by matching number I want to make the line in voiceglue.conf as DEST_NUMBER http://127.0.0.1/vxml/FILE.vxml so that I can catch parameters and call the file dynamically. Please advice me. Best Regards, Roy

[asterisk-users] N Priority in Mysql

Why doesn't the n priority work in a mysql database?? This way I don't have to re-number everything when I insert a new line... -- Co-op Vacation Rentals www.coopvr.com 15218 Summit Ave Suite #300-354 Fontana, CA 92336 Phone/Fax (855) 760-COOP (2667) --

Re: [asterisk-users] Outoing Calls Motif Google Voice Calls Ring After Pick-up

What is the quickest way to apply this patch? Do I just overwrite the code in the res_xmpp.c with copy/replace and re-compile? Co-op Vacation Rentals www.coopvr.com 15218 Summit Ave Suite #300-354 Fontana, CA 92336 Phone/Fax (855) 760-COOP (2667) On 1/11/2013 3:34 PM, Kai-Uwe Jensen wrote:

Re: [asterisk-users] Outoing Calls Motif Google Voice Calls Ring After Pick-up

Could this be why incoming calls to voice do not ring asterisk extensions too? On Jan 10, 2013 4:52 AM, Joshua Colp jc...@digium.com wrote: Joshua Colp wrote: Kai-Uwe Jensen wrote: On Wed, Jan 9, 2013 at 5:38 PM, Roy Abshire r...@coopvr.com mailto:r...@coopvr.com wrote: I have

Re: [asterisk-users] DIDForSale spam

I just ignore spam if I'm not interested and flag them so they go right into my trash folder. I think its more exhausting debating the issue on this forum. I got the email too from DIDForSale but now I'm getting alot more from this thread. It really didn't bother me as much as reading all the

Re: [asterisk-users] Outoing Calls Motif Google Voice Calls Ring After Pick-up

...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Roy Abshire Sent: Monday, January 07, 2013 1:22 PM To: Asterisk Users Subject: [asterisk-users] Outoing

[asterisk-users] Outoing Calls Motif Google Voice Calls Ring After Pick-up

Outoing calls I make using Motif Google Voice Calls continue ringing even after the other end picks up. I have to restart Asterisk to resolve the issue. I don't see any errors. It's not recognizing that the other party picked up the phone and restarting Asterisk fixes it only for a day. --

Re: [asterisk-users] DECT phone for home: siemens A510 v. Grandstream DP715

I've been using the Gigaset A580 Base and A58H Phone for about 3 years now. Never gave me problems. The call Quality is excellent! I only have 1 handset connected to the Base but I want more. I bought a Linksys WIP330 as a 2nd phone to try out and that works just as good without a base unit.

Re: [asterisk-users] DECT phone for home: siemens A510 v. Grandstream DP715

, Roy Abshire r...@coopvr.com wrote: I've been using the Gigaset A580 Base and A58H Phone for about 3 years now. Never gave me problems. The call Quality is excellent! I only have 1 handset connected to the Base but I want more. I bought a Linksys WIP330 as a 2nd phone to try out and that works

Re: [asterisk-users] DECT phone for home: siemens A510 v. Grandstream DP715

. Just picking up your phone will not connect you to the call in progress on the other line. Co-op Vacation Rentals www.coopvr.com 15218 Summit Ave Suite #300-354 Fontana, CA 92336 Phone/Fax (855) 760-COOP (2667) On 12/11/2012 2:39 PM, sean darcy wrote: On 12/11/2012 04:37 PM, Roy Abshire wrote

Re: [asterisk-users] Intruder

] exten = s,1,Set(CALLERID(name)=Blacklisted) exten = s,n,Wait(3) exten = s,n,Playback(${playback}) exten = s,n,HangUp() If you want to add a KEY to your dialplan to add to blacklist or whitelist: [roy] exten = roy,*,Macro(blacklist-add,%,${DB(global/lastcallerid)}) exten = roy,#,Macro(whitelist

[asterisk-users] 3G Quality

Has anyone been able to configure Asterisk to work over 3G? I bought Nokia Cell Phones just for that purpose and they register fine over WiFi and 3G but the quality is just not good enough and sometimes the call just disconnects. I have Allow as: ilbc gsm ulaw alaw -- Co-op Vacation Rentals

Re: [asterisk-users] 3G Quality

Rentals www.coopvr.com 15218 Summit Ave Suite #300-354 Fontana, CA 92336 Phone/Fax (855) 760-COOP (2667) On 11/14/2012 2:32 PM, Carlos Alvarez wrote: On Wed, Nov 14, 2012 at 3:30 PM, Roy Abshire r...@coopvr.com mailto:r...@coopvr.com wrote: Ok, what about 4G, I tried it with my 4G ATT Hotspot

Re: [asterisk-users] 3G Quality

and was successful using 3G or 4G. Co-op Vacation Rentals www.coopvr.com 15218 Summit Ave Suite #300-354 Fontana, CA 92336 Phone/Fax (855) 760-COOP (2667) On 11/14/2012 4:02 PM, Carlos Alvarez wrote: On Wed, Nov 14, 2012 at 4:20 PM, Roy Abshire r...@coopvr.com mailto:r...@coopvr.com wrote: My goal

Re: [asterisk-users] Fax Configuration

I got it working so thought I'd follow up! Here is what I had to do... I'm using Actuate 11 and had res_fax support already I had to download, compile, and install res_fax_digium I had to order a Free Fax License from digium for 1 channel I had to change faxdetect=yes to faxdetect=cng in

[asterisk-users] Incoming Fax to Recipient using OCR

in Perl to search the output.txt file for the to: name or TO: NAME or To: Name Then I want to do something like: Switch($to) { Case: Roy - Email u...@gmail.com Case: Jeff - Email u...@yahoo.com Default: Email ad...@domain.com } -- Co-op Vacation Rentals www.coopvr.com 15218 Summit Ave Suite #300

[asterisk-users] Incoming Fax to Recipient using OCR

in Perl to search the output.txt file for the to: name or TO: NAME or To: Name Then I want to do something like: Switch($to) { Case: Roy - Email u...@gmail.com Case: Jeff - Email u...@yahoo.com Default: Email ad...@domain.com } -- Co-op Vacation Rentals www.coopvr.com 15218 Summit Ave Suite #300

Re: [asterisk-users] forwarding all calls to cells

I use Flowroute as my VOIP provider and this is exactly what I do with all my clients. No problems except it cost me an inbound/outbound connection each time which costs .02 = .01 inbound/.01 outbound per minute instead of .01. Co-op Vacation Rentals www.coopvr.com 15218 Summit Ave Suite

[asterisk-users] Fax Configuration

What is the best way for me to setup Fax Capability with VOIP only. I have a Asterisk Server hosted on the internet without a modem. I'm using Flowroute, which is working awesome, for VOIP calls. I only have a SIP Phone at home and two Printer/Scanner/Fax Printers. I'm not sure which Fax

[asterisk-users] Outgoing Google Motif Calls connect but continue ringing on outgoing side

I upgraded from Asterisk 10 to 11 and switched from gtalk.conf and jabber.conf to use motif.conf and xmpp.conf. I disabled gtalk and jabber from loading in modules.conf noload = res_jabber.so noload = chan_gtalk.so After copying my settings to the new conf files and restarting Asterisk I had

[asterisk-users] dial multiple extensions

continue to ring and the FXO and FXS ports are connected and passed into voicemail. I'm running on Debian Squeeze/6.0.1 and am running the stock Asterisk 1.6.2.9-2+squeeze2 package. If anyone has some suggestions, I'd be happy to hear them. Thanks! Roy

[asterisk-users] Asterisk modifies from header

Hi, I have configured few users in my system with '-' E.g. 1000-1001, 1000-1002 etc.. I am successfully able to registered as 1000-1001 and dials number 937575346. The other end sees 10001001 (missing '-') as a caller. I did verified in the packet capture surprisingly it is

[asterisk-users] b option in Directory

I'm running asterisk 1.4.21 and I see if I go on the wiki that there's a b option that let you enter the first name OR last name of a user. I see that to make this work I need a patch. I'm wondering how can I install this patch as it's an option one of my customer would like to have but I never

Re: [asterisk-users] Question about OSLEC or HPEC with AsteriskNow

AsteriskNow use CentOS 5 and it comes preinstalled with dahdi and asterisk with the freepbx GUI interface and it seems to be missing all the dev packages Martin On 2009-11-17, at 02:19, Olivier wrote: 2009/11/17 Martin Roy m...@mac.com I was previously using an old computer running

[asterisk-users] Question about OSLEC or HPEC with AsteriskNow

I was previously using an old computer running Asterisk 1.2 with zaptel. Once the CPU fried I switch to a new computer and I chose AsteriskNow 1.5 running in 64bits to simplify the installation process. I manage to find my way with configuring dahdi instead of zaptel and to switch all my

Re: [asterisk-users] PRI Splitter

Adtran Atlas 550. We were bring in a single pri into an atlas 550 and then splitting it up so that 6 channels went to a video system (h.320) and 17 channels to our PBX. You can also convert the signaling or send out on different type of connections like v.35. Pretty cool device and rock solid.

Re: [asterisk-users] Polycom LDAP Corporate Directory

it. I have quite a few polycoms and didnt even know polycom had this feature! :) This is obviously a separate peice of software that must be purchased and installed on the phones. Looks amazing though- any idea on pricing?. On Fri, 2008-04-18 at 14:53 -0400, Anciso, Roy wrote

[asterisk-users] Polycom LDAP Corporate Directory

Anyone use the LDAP feature yet on the polycom phones? If so how well does it work? How are you using it in your environment? http://polycom.com/usa/en/products/voice/desktop/soundpoint_ip/applicati ons/corporate_directory_access.html Roy Anciso Director of Technology Manistee Intermediate

Re: [asterisk-users] Hardphone SIP phone costs

I understand the maximizing pricing and branding aspect of phones but when you look at feature set it just doesn't make sense. And as far as purchasing the phone you can get it without a contract at the same price. When I starting thinking about it, can anyone else see a time when desk phones

[asterisk-users] Hardphone SIP phone costs

doesn't make sense. Am I the only one that thinks this? Roy Anciso Director of Technology Manistee Intermediate School District 772 East Parkdale Avenue Manistee, MI 49660 Ph: 231-723-4264 Fx: 231-398-3036 [EMAIL PROTECTED] ___ -- Bandwidth

Re: [asterisk-users] Druid Open Source Edition

Is there a limit on how many phones you can use? I couldn't find anything on the website about this. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joshua Wilson Sent: Wednesday, March 12, 2008 1:43 PM To: asterisk-users@lists.digium.com

[asterisk-users] Linksys SPA-942 Phones

using Asterisk? * How is the speaker phone quality? Thanks Roy Anciso Director of Technology Manistee Intermediate School District 772 East Parkdale Avenue Manistee, MI 49660 Ph: 231-723-4264 Fx: 231-398-3036 [EMAIL PROTECTED] ___ -- Bandwidth

[asterisk-users] Script for seeding polycom phones with an extension directory

directory\n; echo item_list\n; foreach ($directory as $v) { echo item\n; echo $v; echo /item\n; } echo /item_list\n; echo /directory\n; ? Roy Anciso Director of Technology Manistee Intermediate School District 1710 Merkey Road Manistee, MI 49660 Ph: 231-723-4264 Fx: 231-723-1690

[asterisk-users] Followme

. Before the followme app is initiated the caller is prompted to locate the person (by pressing 1 which initiates followme) or to continue onto voicemail. Thanks Roy Anciso Director of Technology Manistee Intermediate School District 1710 Merkey Road Manistee, MI 49660 Ph: 231-723-4264 Fx

Re: [asterisk-users] Followme

the followme app. On Jan 22, 2008 10:25 AM, Anciso, Roy [EMAIL PROTECTED] wrote: I've been reading up on followme app for asterisk 1.4 and I have it working but I was wondering if the following was possible: Based on followme.conf present the caller with the option to locate the person: Call

Re: [asterisk-users] Voicemail - is it possible to automatically usethe extension being dialed from?

Heres what I do for this: exten = *85,1,VoicemailMain(${CALLERID(NUM)}) From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of arkda Sent: Tuesday, January 22, 2008 5:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject:

Re: [asterisk-users] cisco ip phne 7911G with asterisk

Of Christian Pinedo Zamalloa Sent: Friday, January 18, 2008 8:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] cisco ip phne 7911G with asterisk On Wed, Jan 16, 2008 at 10:26:04AM -0500, Anciso, Roy wrote: Now that you have your 7911g phone up running, would

Re: [asterisk-users] cisco ip phne 7911G with asterisk

Now that you have your 7911g phone up running, would you mind checking the audio quality when leaving a voicemail for on another local asterisk user from this phone? I have a 7911g and I hear loud audio taps from the phone. The 7961g phone doesn't have this issue. I'm just trying to rule out the

[asterisk-users] Record calls then send them to users voicemail

Just wondering if this is possible: Make a call from a registered sip extension (Doesn't matter if it's internal or external) during the call press a key sequence let say *90 to start recording call. When the call ends the recording automagically goes to their voicemail. Thanks Roy Anciso

Re: [asterisk-users] Cisco 79xx XML services

Although it's not LDAP I used a script that I found on the voip wiki and changed it so it looked at only sip configuration files. It also alphabetizes the output so it can be displayed that way on the phone. Below are my notes on the subject. If someone is willing to post this to the wiki and

Re: [asterisk-users] Cisco 7941G-GE with Asterisk and CTPSEP odyssee

I've upgraded from SCCP to SIP 8.x.x branch on 7961g and 7911g without any problems. As far as the CTLSEP File (Straight from Cisco): http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/english/i pp7960/addprot/mgcp/frmwrup.htm#wp1047292 The CTLSEP MAC file is a certificate trust

Re: [asterisk-users] Cisco 7941G-GE with Asterisk and CTPSEP odyssee

NO CTL and CTL processing FAILED** ctl-err 12 (file is too small) NOT 09:29:03.227508 DHCP: Restart - delay = 1 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anciso, Roy Sent: Friday, January 04, 2008 8:43 AM To: Asterisk Users Mailing List - Non

Re: [asterisk-users] Cisco 7941G-GE with Asterisk and CTPSEP odyssee

and CTPSEP odyssee That is not the name the phone requests When uping my 7960, the empty file did the trick I so far am unable to go beyond 7.1 however, as Asterisk rejects anything I dial with 7.3 Anyone have SIMPLE sample config files? John Novack Anciso, Roy wrote: Try naming the empty

Re: [asterisk-users] Cisco 7941G-GE with Asterisk and CTPSEP odyssee

:02.696335 SECD: EROR:updateCTL: ** had NO CTL and CTL processing FAILED** ctl-err 12 (file is too small) NOT 09:29:03.227508 DHCP: Restart - delay = 1 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anciso, Roy Sent: Friday, January 04, 2008 8:43

Re: [asterisk-users] 7970 CTLFile.tlv?

I believe you can create a blank file to keep the phone from complaining. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Rubenstein Sent: Friday, December 21, 2007 10:16 AM To: Asterisk -Users Subject: [asterisk-users] 7970 CTLFile.tlv?

Re: [asterisk-users] Cisco 7961 new firmware stops readingconfiguration files

Chad, You might want to upgrade to the latest firmware. I have 7961g on 8-3-3SR2S and works very well. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chad Osmond Sent: Thursday, December 20, 2007 10:33 AM To: asterisk-users@lists.digium.com Subject:

[asterisk-users] Cisco 7911g Poor Audio Quality w/ Asterisk Voicemail and MOH

the music stops and then starts to play the beginning of the MOH file and continues until the voicemail recording takes over. The best way to describe it is like a dj scratching records. Does anyone else have issues like these? Thanks Roy Anciso Director of Technology Manistee Intermediate

[asterisk-users] Play Beep instead of MOH

Is there a way to tell asterisk to beep every few seconds rather than play MOH. Thanks Roy Anciso Director of Technology Manistee Intermediate School District 1710 Merkey Road Manistee, MI 49660 Ph: 231-723-4264 Fx: 231-723-1690 [EMAIL PROTECTED

[asterisk-users] Digium TE120P versus Sangoma A101D-X

was transferred to sales person they would not give me the keys stating that I have to have an analog card to obtain the license. Thanks Roy Anciso Director of Technology Manistee Intermediate School District 1710 Merkey Road Manistee, MI 49660 Ph: 231-723-4264 Fx: 231-723-1690 [EMAIL

Re: [asterisk-users] Music on Hold Problem w/ Transfers

I'm having this problem. Here is my output with verbosity on 10: -- Executing [EMAIL PROTECTED]:1] Dial(SIP/2524-099012b0, SIP/2523|15) in new stack -- Called 2523 -- SIP/2523-09905220 is ringing -- SIP/2523-09905220 answered SIP/2524-099012b0 -- Packet2Packet bridging

Re: [asterisk-users] Music on Hold Problem w/ Transfers

Asterisk version 1.4.13 Also when I listened in on a transfer it sounds like the moh is trying to start but then immediately stop and tries to start again. Below is my musiconhold.conf: [default] mode=files directory=/var/lib/asterisk/moh random=no -Original Message- From: [EMAIL

[asterisk-users] Cisco phones and 32 directory object limit

Roy Anciso Director of Technology Manistee Intermediate School District 1710 Merkey Road Manistee, MI 49660 Ph: 231-723-4264 Fx: 231-723-1690 [EMAIL PROTECTED] ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk

[asterisk-users] Page Command

Hello List, I'm looking at the page command. I was wondering if there was a way to set a wild card to dial all registered sip devices. For example page all 1XXX extensions. Thanks in advance Roy Anciso Director of Technology Manistee Intermediate School District 1710 Merkey Road

Re: [asterisk-users] Cisco 7911/7941/7970/7971 Softkey XML Files

Sorry forgot the images: http://picasaweb.google.com/ranciso/AsteriskImagesCiscoPhones From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anciso, Roy Sent: Thursday, November 15, 2007 3:47 PM To: Asterisk Users Mailing List - Non-Commercial

Re: [asterisk-users] Cisco 7911/7941/7970/7971 Softkey XML Files

7911/7941/7970/7971 Softkey XML Files 2007/11/15, Greg Oliver [EMAIL PROTECTED]: On Thu, 2007-11-15 at 07:31 +0100, Olivier wrote: 2007/11/14, Greg Oliver [EMAIL PROTECTED]: On Tue, 2007-11-13 at 08:44 -0500, Anciso, Roy wrote: Hello List, Does

Re: [asterisk-users] Cisco 7911/7941/7970/7971 Softkey XML Files

, 2007-11-13 at 08:44 -0500, Anciso, Roy wrote: Hello List, Does anyone have access to the soft key configuration files for the Cisco 7911/7941/7970/7971 phones? Checked up on the Cisco site and didn't find much up there. Thanks Softkeys running both SCCP and SIP firmware are both sent

[asterisk-users] Cisco 7911/7941/7970/7971 Softkey XML Files

Hello List, Does anyone have access to the soft key configuration files for the Cisco 7911/7941/7970/7971 phones? Checked up on the Cisco site and didn't find much up there. Thanks Roy Anciso Director of Technology Manistee Intermediate School District 1710 Merkey Road Manistee, MI

Re: [asterisk-users] Cisco 7911/7941/7970/7971 Softkey XML Files

To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7911/7941/7970/7971 Softkey XML Files Anciso, Roy wrote: Hello List, Does anyone have access to the soft key configuration files for the Cisco 7911/7941/7970/7971 phones? Checked up on the Cisco site

Re: [asterisk-users] Cisco IP Communicator with Asterisk

The version I have is 2.1.2.0. It makes for a really nice software sip phone:) The other thing I should note is that you only need the SEPXXX.cnf.xml file and dialplan.xml file in your tftp directory. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of

[asterisk-users] Cisco IP Communicator with Asterisk

CiscoIPCommunicatorSetup.msi /qb SIP=1. After installation configuration is just like configuring a Cisco 7970 hard phone. I used the configuration instructions outlined by Kerry Garrison at Asterisk Tutorials http://www.asterisktutorials.com/showproduct.php?ProductID=10. Roy Anciso Director

[asterisk-users] Selecting OSLEC for zaptel-1.4.6

Hello list, Can someone outline the steps for selecting OSLEC canceller in 1.4.6? I know there was a bug fix for this but I can't figure out how to select it. Thanks Roy Anciso Director of Technology Manistee Intermediate School District 1710 Merkey Road Manistee, MI 49660 Ph: 231

Re: [asterisk-users] Selecting OSLEC for zaptel-1.4.6

] Selecting OSLEC for zaptel-1.4.6 Dave Fullerton wrote: Anciso, Roy wrote: Hello list, Can someone outline the steps for selecting OSLEC canceller in 1.4.6? I know there was a bug fix for this but I can't figure out how to select it. snip / Roy Anciso You shouldn't need to. As long

Re: [asterisk-users] Outgoing PRI CID?

I do this to tie extensions to a particular number: exten = _9X./_2XXX,1,SET(CALLERID(all)=Manistee ISD2317231516) exten = _9X./_1XXX,1,SET(CALLERID(all)=MISD Tecnology2317234264) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Turbo Fredriksson Sent:

Re: [asterisk-users] Cisco Phones

: Roy, While there is a difference in the feature set provided by the SIP and Skinny images for the Cisco phones, the loss is not appreciable in my view. There are some differences in interface aesthetics as well. ___ --Bandwidth

[asterisk-users] Cisco Phones

For those of you running Cisco phones, did you start out with a Cisco CallManager and move to Asterisk? And if you did switch do you find that you or your users are missing features they once had? How have you handle the issue? Thanks Roy Anciso Director of Technology Manistee

[asterisk-users] Cisco phones with Asterisk

Hello List, For those of you using Cisco phones, did you have to purchase a 'SIP license' for each phone? Thanks Roy Anciso Director of Technology Manistee Intermediate School District 1710 Merkey Road Manistee, MI 49660 Ph: 231-723-4264 Fx: 231-723-1690 [EMAIL PROTECTED

[asterisk-users] What web GUI are people happy with?

Just wondering what web GUI people like for asterisk. I installed asterisk from source and I was looking at possibly installing web GUI for system management. So far freepbx.org looks promising anybody else have any suggestions. Thanks Roy Anciso Director of Technology Manistee

Re: [asterisk-users] What web GUI are people happy with?

: [asterisk-users] What web GUI are people happy with? Anciso, Roy wrote: Just wondering what web GUI people like for asterisk. I installed asterisk from source and I was looking at possibly installing web GUI for system management. So far freepbx.org looks promising anybody else have any

[asterisk-users] How to busy out zap channels

I know this topic came up many months back and some discussions were being had on how to do this within the Zaptel drivers. However, I'm looking for even a crude hack that someone has put together to get this done. We have PRI's and LD T1's that are load balanced on two boxes. The hunt order goes

Re: [asterisk-users] Asterisk on Fedora Core 4

Kempgen wrote: Martin Roy wrote: I just did a clean install of Fedora Core 4 on a PC with a TDM400 installed. I installed zaptel 1.2.20.1 and asterisk 1.2.24 correctly. I did make config for both to have zaptel and asterisk start when I boot the computer. My main problem right now is that zaptel

[asterisk-users] Asterisk on Fedora Core 4

I just did a clean install of Fedora Core 4 on a PC with a TDM400 installed. I installed zaptel 1.2.20.1 and asterisk 1.2.24 correctly. I did make config for both to have zaptel and asterisk start when I boot the computer. My main problem right now is that zaptel doesn't load at startup so

Re: [asterisk-users] Channel Bank Recommendations

On 8/29/07, Mark Bell [EMAIL PROTECTED] wrote: Need to add some fxs and fxo ports behind a fonebridge2 box any recommendations a channel bank We're using a Rhino here and haven't had one problem with it. It's connected to an analog fax server and lights up for hours at a time. Probably been

Re: [asterisk-users] where is 1.4.12?

On 8/29/07, Steve Totaro [EMAIL PROTECTED] wrote: Kind of harsh for am employee of Digium on a public Asterisk mailing list, don't you think? Enough The Digium/Aseterisk bashing seems to be at an all time high recently. You seem to be involved in a lot of it. Russell has given most of

[asterisk-users] asterisk-backports.org giveaway

, if someone wants to take over, be my guest :) roy --- Roy Sigurd Karlsbakk [EMAIL PROTECTED] Tlf: 98013356 --- Why is it drug addicts and computer afficionados are both called users? -- Clifford Stol ___ --Bandwidth and Colocation provided

Re: [asterisk-users] Channel Bank

On 5/6/07, Doug Lytle [EMAIL PROTECTED] wrote: The Adit 600 is a favorite of mine. Doug Would second the Adit too. We are running a Rhino now and have had no problems with it. -Brian ___ --Bandwidth and Colocation provided by Easynews.com --

Re: [asterisk-users] Cacti/Nagios monitoring, what do you want graphed.

On 4/12/07, Brandon Kruse [EMAIL PROTECTED] wrote: Hey guys, What are some of the numbers you guys want graphed? Curious how you are going to do this and will it be backwards portable. One of our engineers wrote an app that queries the manager interface to build RRD data. That's sent over

[asterisk-users] Dial out from AGI

like so sub dial($number){ print Dial(\Zap/1-1\, \Zap/g2/$number\)\n; } but I get the error handle_exec: Could not find application (Dial(Zap/1-1,Zap/g2/866555) Anyone have any suggestions? Thanks, Roy ___ --Bandwidth and Colocation provided

Re: [asterisk-users] Any quiet 24 port POE switches out there?

On 1/3/07, John French [EMAIL PROTECTED] wrote: I have an upcoming install which places the switch close to some employees in a quiet work environment. Can anyone recommend a quiet 24 port POE switch? The 8port Netgear switch on my desk doesn't have any fans. FS108p. Not sure if they make

Re: [asterisk-users] webvoicemail

On 12/13/06, Ed Nuñez [EMAIL PROTECTED] wrote: I've been trying to find where to download the Web Vmail application and instructions on how to install it for Asterisk BE. Any ideas? Is this any different than the vmail.cgi that comes with the open version? Otherwise, you will just need to

Re: [asterisk-users] Polycom MyStat

On 12/13/06, LST [EMAIL PROTECTED] wrote: I think that is strictly a Polycom to Polycom thing (Buddywatch). I do not believe it affects Asterisk (i.e. Busy = DND). With that being said, I don't think it works very well even with all Polycom phones. I can change my status to Busy and look

Re: [asterisk-users] extension launch into AGI

Time Bandit wrote: I've got a simple set up with 1 fxo port and 1 fxs port in a Digium card connected to a POTS line and a phone set (physical extension). I've got all incoming calls launching directly into an AGI script. I'd like to do the same for the physical extension. In other words, when

[asterisk-users] extension launch into AGI

set, the AGI is launched without dialing any digits. Anyone have any ideas for me to try? Thanks in advance, Roy ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http

Re: [asterisk-users] Re: Re: Voicemail Press '0'

On 10/10/06, LJ [EMAIL PROTECTED] wrote: In my Asterisk 1.2.9.1 installation I use the following:in voicemail.conf include the following:exitcontext=vmloginoperator=yes Sorry to revive a month old thread but here was the easy button solution for me. With debugging on I did a reload

Re: [asterisk-users] Cisco 7960 Skinny calling SIP phone

PROTECTED] Content-Type: text/plain; charset=utf-8Which asterisk release are you running chan_skinny under?- Original Message -From: Will Roy [EMAIL PROTECTED]To: asterisk-users@lists.digium.comSent: Monday, October 30, 2006 7:52:01 PM GMT-0600 US/Central Subject: [asterisk-users] Cisco 7960

Re: [asterisk-users] app_meetme not loading

On Sun, Oct 29, 2006 at 04:47:48PM +0800, Will Roy wrote: I originally built my Asterisk server without installing the Zaptel package as it was going to be a purely SIP based system. However when I went to setup conferencing using meetme I found out that app_meetme is dependant on the ztdummy

[asterisk-users] Cisco 7960 Skinny calling SIP phone

Before I got down the path of converting a Cisco 7960 I haveover to SIP I wanted to try and set it up using Skinny. Thephone registersok withAsterisk. When I call a SIP softphone extension on my network the call is made and I can answering it. However no voice is heard over the call. When I

[asterisk-users] app_meetme not loading

I originally built my Asterisk server without installing the Zaptel package as it was going to be a purely SIP based system. However when I went to setup conferencing using meetme I found out that app_meetme is dependant on the ztdummy for timing. I have now installed the zaptel package and I

Re: [asterisk-users] Digium vs. Sangoma

On 10/23/06, Unmetered Pipe [EMAIL PROTECTED] wrote: I don't mean to be a troll in any way shape or form. I was on IRC last night and I observed the following convo. below. What do you guys make of it ? That you are a troll? -Brian ___ --Bandwidth

Re: [asterisk-users] Rpath PoundKey 1.2

Is there a way to correct the problem or can the files be generated? Did you run the registration program? Asterisk won't start unless it's registered with Digium. -Brian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users

[asterisk-users] Re: [asterisk-announce] Digium G.729 codec binaries updated for Asterisk 1.4 beta

and playing guitars at people - Terry Pratchett --- Roy Sigurd Karlsbakk [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit

[asterisk-users] [asterisk-dev] Re: [asterisk-announce] Digium G.729 codec binariesupdated for Asterisk 1.4 beta

and playing guitars at people - Terry Pratchett --- Roy Sigurd Karlsbakk [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit

[asterisk-users] new in 1.4?

? thanks roy --- Humans mostly aren't particularly evil. They just get carried away by new ideas, like dressing up in jackboots and shooting people, or dressing up in white sheets and lynching people, or dressing up in tie-dye jeans and playing guitars at people - Terry Pratchett

Re: [asterisk-users] DSL router with integrated SIP proxy?

as outgoing proxy, works very well. -- Roy-Magne Mo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Call forward with CFU?

Hi all I've got the following message from the telco regarding call forward number presentation. Can someone please help me decipher this? I don't understand shit about this :P roy Roy , Below is an extract taken from a working scenario of the CFU (A B - B C) functionality problem

Re: [asterisk-users] When does Scalability requests Asterisk

-gateways talking SIP to a hub server talking to clients. Clients register with hub server. pstngw gets a call in, sends it to hub server, hub server sends reinvite to pstngw, pstngw sends invite to client whose NAT gateway does not know the pstngw's address and throws the packet away... roy

[asterisk-users] SHSU asterisk installation?

if someone know how they plan to do this, in detail. thanks roy --- Humans mostly aren't particularly evil. They just get carried away by new ideas, like dressing up in jackboots and shooting people, or dressing up in white sheets and lynching people, or dressing up in tie-dye jeans and playing

Re: [asterisk-users] SHSU asterisk installation?

know how they plan to do this, in detail. Yes, there have been several threads about this. Obviously I _have_ tried to google about this, so if you could point me to one of those threads, I'd be grateful roy --- Humans mostly aren't particularly evil. They just get carried away by new ideas

[asterisk-users] Problems with outgoing calls

ringsAll solutions I have read seem to be based on the assumption that a X100P card is being used with analogue lines.Any help on this would be much appreciated. Dr Roy GardnerDirectorwww.psycle.comTel: 01948 780120Mob: 07713 985657 ___ --Bandwidth

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