Re: [asterisk-users] PRI Setup problem

Thanks for all of the good suggestions. I've been able to get things working. I had been trying to use zaptel svn in order to get past error messages with compiling ztdummy.ko for the 2.6.22 kernel (http://bugs.digium.com/view.php?id=10426 which has been apparently been solved in svn). Too b

[asterisk-users] zaptel upgrade trouble (1.2.10 -> 1.2.20.1)

Hello! I have been trying upgrade zaptel from 1.2.10 to 1.2.20.1. I am using asterisk 1.2.10 with one TDM2400P (all 6 module in use) and one TE405P. When I upgrade to 1.2.20.1 the order of cards mess up and therefore zaptel.conf is unusable and gives error. Why is it happen and what do I need to c

Re: [asterisk-users] Anyone use the Linksys phones?

in the office we use 4 IP Phone SPA942 from linksys and i must say that they are great ;) easy to install and maintain ... some stuff is missing but it works great ... Greatings, Jan De Coster Robert Webb wrote: > Is anyone out there using any of the newer linksys phones since Cisco > took ov

Re: [asterisk-users] PRI Setup problem

Alvin Austin wrote: > Thanks for all of the good suggestions. I've been able to get things > working. > > I had been trying to use zaptel svn in order to get past error messages > with compiling ztdummy.ko for the 2.6.22 kernel > The newest kernel that I've been able to use with the current

Re: [asterisk-users] Anyone use the Linksys phones?

On 9/24/07, Erik Anderson <[EMAIL PROTECTED]> wrote: > On 9/24/07, Steve Davies <[EMAIL PROTECTED]> wrote: > > > > The phones can send a parameter to the provisioning server to indicate > > that they want an "Update" if they do this, and you send no network or > > other major config parameters, the

Re: [asterisk-users] zaptel upgrade trouble (1.2.10 -> 1.2.20.1)

On Tue, Oct 02, 2007 at 10:06:06AM +0200, Artifex Maximus wrote: > Hello! > > I have been trying upgrade zaptel from 1.2.10 to 1.2.20.1. I am using > asterisk 1.2.10 with one TDM2400P (all 6 module in use) and one > TE405P. When I upgrade to 1.2.20.1 the order of cards mess up and > therefore zapt

Re: [asterisk-users] How To Transfer Asterisk Installation to a Different Machine

This asrticle was meant as a backup, but I guess it's basically the same thing as what you are looking for: http://astrecipes.net/index.php?n=93 I hope this helps l. In data Mon, 01 Oct 2007 20:08:20 +0200, Robert DeVries <[EMAIL PROTECTED]> ha scritto: > I am having some hardware problems w

Re: [asterisk-users] zaptel upgrade trouble (1.2.10 -> 1.2.20.1)

On 10/2/07, Tzafrir Cohen <[EMAIL PROTECTED]> wrote: > On Tue, Oct 02, 2007 at 10:06:06AM +0200, Artifex Maximus wrote: > > Hello! > > > > I have been trying upgrade zaptel from 1.2.10 to 1.2.20.1. I am using > > asterisk 1.2.10 with one TDM2400P (all 6 module in use) and one > > TE405P. When I upg

Re: [asterisk-users] zaptel upgrade trouble (1.2.10 -> 1.2.20.1)

On Tue, Oct 02, 2007 at 12:47:55PM +0200, Artifex Maximus wrote: > On 10/2/07, Tzafrir Cohen <[EMAIL PROTECTED]> wrote: > > On Tue, Oct 02, 2007 at 10:06:06AM +0200, Artifex Maximus wrote: > > > Hello! > > > > > > I have been trying upgrade zaptel from 1.2.10 to 1.2.20.1. I am using > > > asterisk

[asterisk-users] Announcement file is unavailable?????

Folks, please, take a look at this asterisk log message: [Oct 2 08:55:13] WARNING[10290] app_queue.c: Announcement file 'atcert' is unavailable, continuing anyway... [Oct 2 08:55:13] WARNING[10290] app_queue.c: Agent on Agent/1001002 hungup on the customer. but: -bash-3.1$ whoami asterisk -bas

Re: [asterisk-users] Selecting a specific line from Zap/g And secondary dial tone

Dear List; Thanks alot for the help. But how can I let the second dial tone (after pressing the extension to select that FXO port) to be difference than normal dial tone? Regards Bilal Ghayad -- Correction, on FXO port not FXS, second, read his email first: "Also, how it

Re: [asterisk-users] Anyone use the Linksys phones?

Chris Bagnall wrote: > > The 4 line limitation has never been a problem for the vast majority of > people. > I can't imagine what an office worker would do with four line appearances. I use a 6 line Polycom but I register different line appearances to different customer PBX's that I am worki

Re: [asterisk-users] Digium acquires Switchvox

On Thu, 2007-09-27 at 10:25 -0500, Jared Smith wrote: > We're also planning an audio conference next Tuesday in which you'll > be able to dial in and ask any questions you may have concerning the > acquisition. (I'll post the exact time and details as soon as I have > them.) The dial-in number fo

[asterisk-users] Hey

Just a quick thanks for all being here. I started to type up a message and realized my problem, so instead I'm saying thanks for all the good information you all pass through my mailbox every day and giving me a place to realize my error before I even ask the question.

[asterisk-users] E&M Wink and T4xxP losing ability to dial

Hey folks, I'm pulling my hair out on this situation and would welcome some advice: I'm using the AMI Manager to Originate calls onto 2 E&M wink T1 circuits via a T4XXP card from Digium. Everything works fine for about 600(+/- 50) calls then the Manager is suddenly unable to launch calls. Using

Re: [asterisk-users] SIP Panel?

Matt - I'd like the sourcecode for the SIP panel. - Walt Matt Riddell wrote: > -BEGIN PGP SIGNED MESSAGE- > Hash: SHA1 > > Terry Giufre-Sweetser wrote: > >>Dear List, >> >>Has anyone found or written a status panel application, windows or >>linux, that uses SIP notifies and subscripti

Re: [asterisk-users] Announcement file is unavailable?????

Carlos Alberto Hastenreiter Assumpção wrote: > Folks, please, take a look at this asterisk log message: > > [Oct 2 08:55:13] WARNING[10290] app_queue.c: Announcement file > 'atcert' is unavailable, continuing anyway... > [Oct 2 08:55:13] WARNING[10290] app_queue.c: Agent on Agent/1001002 > hungup

Re: [asterisk-users] Selecting a specific line from Zap/g

No, ignorepat is for FXS ports (FXS ports use FXO signaling). Also, ignorepat does not apply to SIP phones, because SIP phones provide their own dialtone, not a dialtone provided by Asterisk. Al lists wrote: > Correction, on FXO port not FXS, > second, read his email first: > "Also, how it will

[asterisk-users] app_conference

Hi list, Has anyone use app_conference? I want to hear what your opinions are. Thnx. <>___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.co

[asterisk-users] issues with rtsavesysname

I am currently struggling to implement this feature in realtime asterisk. I have configured realtime asterisk and it works great, however I can not get regserver to update via the rtsavesysname in sip.conf. Heres my configs.. sip.conf [EMAIL PROTECTED] asterisk]# more sip.conf [general] displays

Re: [asterisk-users] PRI Setup problem

Doug Lytle wrote: > Alvin Austin wrote: >> Thanks for all of the good suggestions. I've been able to get things >> working. >> >> I had been trying to use zaptel svn in order to get past error messages >> with compiling ztdummy.ko for the 2.6.22 kernel >> > > The newest kernel that I've bee

[asterisk-users] Supervised call transfer problem

Hi all, I am running Asterisk in conjunction with a Sip proxy. Asterisk is registered to an external SIP carrier (sip.uni.it) If a call reachs Asterisk through the SIP carrier, then it is forwarded to the external SIP proxy extension ([EMAIL PROTECTED]), when the extension 530 that h

Re: [asterisk-users] PRI Setup problem

Stephen Bosch wrote: > He's using the beta wanpipe, which works with the newer kernels. > > So am I. wanpipe-3.1.4.tgz I guess I'll give it another go. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty

Re: [asterisk-users] zaptel upgrade trouble (1.2.10 -> 1.2.20.1)

On 10/2/07, Tzafrir Cohen <[EMAIL PROTECTED]> wrote: > On Tue, Oct 02, 2007 at 12:47:55PM +0200, Artifex Maximus wrote: > > On 10/2/07, Tzafrir Cohen <[EMAIL PROTECTED]> wrote: > > > On Tue, Oct 02, 2007 at 10:06:06AM +0200, Artifex Maximus wrote: > > > > Hello! > > > > > > > > I have been trying u

[asterisk-users] Queue members, URI.

Is there an advantage to having a Queue members URI in the form: SIP/User (or indeed IAX2/User) Over Local/@context ? I know that the latter will allow you to do things like set counting logic etc. through dialplan operations, but the former appears to be a more direct route to calling the part

[asterisk-users] Zaptel slow dial out - TDM400P

Below is a copy of my log, zapata.conf & extensions.conf that relate to the ZAP lines. Basically when we dial out it takes on 10-12 seconds before the ZAP line actaully picks up. I'm hoping to find out what the cause is for this as it's causing user grief with extremely long connect times, and I

Re: [asterisk-users] PRI Setup problem

Doug Lytle wrote: > Stephen Bosch wrote: >> He's using the beta wanpipe, which works with the newer kernels. >> >> > So am I. wanpipe-3.1.4.tgz > Hmn -- here's what his post said: > I've recompiled with the latest svn sources f

Re: [asterisk-users] PRI Setup problem

Stephen Bosch wrote: > Doug Lytle wrote: > > I don't see a 3.3.0.p4 on the wiki, but maybe it's on the ftp... > > That it is! ftp://ftp.sangoma.com/linux/custom/3.3 Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve n

[asterisk-users] Segmentation fault in app_voicemail (ODBC/PSQL problem)

I have been testing with asterisk 1.4.11 and have found a segmentation fault while using voicemail. It happens when I try to forward a voicemail. As soon as I press the option the server crashes. I ran asterisk up inside gdb and got the following stack trace

Re: [asterisk-users] What's the deal with ATAcomm?

> Dear Atacomm Customers, > We apologize, but as of 6:00pm CST Friday, September 21st, Atacomm > and its parent company Ataractic Corporation has ceased > operations. We appreciate the 7 years of loyalty and support from > our customers. We sincerely regret any adverse effects this may have cause

Re: [asterisk-users] What's the deal with ATAcomm?

Kenneth Padgett wrote: >> Dear Atacomm Customers, >> We apologize, but as of 6:00pm CST Friday, September 21st, Atacomm >> and its parent company Ataractic Corporation has ceased >> operations. We appreciate the 7 years of loyalty and support from >> our customers. We sincerely regret any adverse

Re: [asterisk-users] Queue members, URI.

Thomas Kenyon wrote: > Is there an advantage to having a Queue members URI in the form: > > SIP/User (or indeed IAX2/User) > Over > Local/@context > > ? > > I know that the latter will allow you to do things like set counting > logic etc. through dialplan operations, but the former appears to b

Re: [asterisk-users] Hey

Yes, I second that. I talk and talk and talk but haven't said thank you enough! Ken Williams wrote: > Just a quick thanks for all being here. I started to type up a > message and realized my problem, so instead I'm saying thanks for all > the good information you all pass through my mailbox e

[asterisk-users] WARNING[26913]: channel.c:786 channel_find_locked: Avoided deadlock for '0x82d9668', 10 retries!

Is anyone familiar with this error message? WARNING[26913]: channel.c:786 channel_find_locked: Avoided deadlock for '0x82d9668', 10 retries! Why does it happen, and how can I prevent from happening. ___ --Bandwidth and Colocation Provided by http://

[asterisk-users] Rhino RCB8FXX

Anyone know if Rhino is planning on supporting zaptel 1.4 anytime soon? This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of t

[asterisk-users] Rhino RCB8FXX

We are using it successfully with zaptel 1.4 -- just be sure and get the latest drivers which are now independent of the zaptel sources. on Tuesday 10/02/2007 Jeremy Mann([EMAIL PROTECTED]) wrote > Anyone know if Rhino is planning on supporting zaptel 1.4 anytime soon? > > ___

Re: [asterisk-users] Rhino RCB8FXX

Latest being 1.1.1 ? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John covici Sent: Tuesday, October 02, 2007 2:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Rhino RCB8FXX We are using it successfully with

Re: [asterisk-users] Supermicro PDSME+ and TE110P [ ref:00D36mPe.50033qy57:ref ] NEW CASE 22828

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Olivier wrote: > Hello Matt, > > Do you mean this comes from (lack of) e1000 network card driver support in > Linux ? >>From memory, Supermicro systems are sold as Linux compliant. > Then, providing Linux compliant drivers should be Supermicro's probl

[asterisk-users] Having problems posting to the list

Hi All I'm having problems posting to this list, no bounces the mails just dont show any advice how to get the postings through is there filtering? robb ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing

[asterisk-users] Asterisk 1.4.12 and Asterisk-addons 1.4.3 released

The Asterisk Development Team has announced the releases of Asterisk 1.4.12 and Asterisk-addons 1.4.3. The Asterisk-addons release contains just a few fixes for the modules in that package, but the Asterisk release contains a large number of bug fixes for all parts of Asterisk. There are many are

Re: [asterisk-users] Rhino RCB8FXX

I am actually using 1.1.0, I might take a look at .1, but this should work with 1.4. on Tuesday 10/02/2007 Jeremy Mann([EMAIL PROTECTED]) wrote > Latest being 1.1.1 ? > > > -Original Message- > From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John covici > Sent: Tuesd

Re: [asterisk-users] meetme conference using g729?

On 10/2/07, Brian West <[EMAIL PROTECTED]> wrote: > > Ok Let me chime in on this one. > If you can use ulaw/alaw because you'll end up with tandem encoding which > will make the conference sound worse to some people. > > All audio coming in will get transcoded to signed linear and pushed down > int

Re: [asterisk-users] meetme conference using g729?

You still do not understand. It doesn't matter if the call coming in is g729 you must transcode it to signed linear, mix the frames and then code it back into g729 you end up with quality loss doing that. /b On Oct 2, 2007, at 4:42 PM, Mark Quitoriano wrote: > > anyway still if there's a

Re: [asterisk-users] Anyone use the Linksys phones?

We use 10 IP Phone SPA941 with asterisk , they are easy to set up, upgrade and are working very well. Regards Rafael -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of jan de coster Sent: Tuesday, October 02, 2007 4:39 AM To: Asterisk Users Mailing List

Re: [asterisk-users] meetme conference using g729?

On Tuesday 02 October 2007 16:55:52 Brian West wrote: > On Oct 2, 2007, at 4:42 PM, Mark Quitoriano wrote: > > anyway still if there's a hack for meetme to work with g729 codec > > this won't be an issue. So is there a hack or patch that i can use > > any codec for meetme? tnx > > You still do not

Re: [asterisk-users] Asterisk 1.4.12 and Asterisk-addons 1.4.3 released

> the Asterisk release contains a large number of > bug fixes for all parts of Asterisk. I am thankful to see the amount of fixes that have gone into this release. However, seeing this many fixes does not give me a warm fuzzy feeling that we won't see a lot more fixes in the near future. So are b

Re: [asterisk-users] meetme conference using g729?

Thanks for making it clearer :) My mind is mush today! /b On Oct 2, 2007, at 5:39 PM, Tilghman Lesher wrote: Or, in other words, you cannot mix compressed data. You must first decompress the data for mixing, then recompress it for transmission. During both operations, there is a potential fo

Re: [asterisk-users] Asterisk 1.4.12 and Asterisk-addons 1.4.3 released

Don Pobanz wrote: >> the Asterisk release contains a large number of >> bug fixes for all parts of Asterisk. > > I am thankful to see the amount of fixes that have gone into this > release. However, seeing this many fixes does not give me a warm fuzzy > feeling that we won't see a lot more fixes

[asterisk-users] Problem making chanskype work in Asterisk 1.4.11 in Fedora 7

Hi, I installed chanskype in Asterisk 1.4.11 in Fedora 7 as follows: a) obtained chanskype tar ball from www.timhunt.net b) tar zxpvf chanskype-trixbox.tgz c) cd chanskype.trixbox d) ./install.sh e) ./makeaccount.sh 1 f) make vncserevr autostart using ntsysv g) install chanskype-1.2.11 h) Add in

Re: [asterisk-users] What's the deal with ATAcomm?

Send me an email off the list, i have em somewhere in my HDD. On 10/2/07, Eric ManxPower Wieling <[EMAIL PROTECTED]> wrote: > > Kenneth Padgett wrote: > >> Dear Atacomm Customers, > >> We apologize, but as of 6:00pm CST Friday, September 21st, Atacomm > >> and its parent company Ataractic Corporat

[asterisk-users] Asterisk like big Voicemail system

Hi friends. I am working in a TELCO, we have a trouble with our very old Alcatel Voicemail system (and now we dont have support and worst this system was forgotten for Alcatel) I've used Asterisk for just small jobs, but I've proposed use it and tomorrow begins with the tests :) ... so: we ha

Re: [asterisk-users] Asterisk like big Voicemail system

If you are asking for dial plan help. The easiest way to do this is to have it come into an extension that just goes to voice mail then use externotify in voicemail.conf to turn on mwi or stuter dial tone. for subscriber service just start in a different extension that calls voicemailmain. you can

[asterisk-users] extensions.conf vs. AEL

Hello, I see that most people are using the extensions.conf syntax (most of the examples and questions here use that syntax). recently I've translated all my dial plan to AEL syntax and I find it much easier, especially when you need IFs. Why most people don't use it? Am I missing something?

Re: [asterisk-users] extensions.conf vs. AEL

In my opinion the dialplan isn't where that logic belongs. /b On Oct 3, 2007, at 12:32 AM, Yehavi Bourvine +972-8-9489444 <[EMAIL PROTECTED] > wrote: > Hello, > > I see that most people are using the extensions.conf syntax (most > of the > examples and questions here use that syntax). recen

Re: [asterisk-users] Having problems posting to the list

but they do, apparently On 10/2/07, robert boardman <[EMAIL PROTECTED]> wrote: > Hi All > > I'm having problems posting to this list, no bounces the mails just > dont show > > any advice how to get the postings through is there filtering? > > robb > > _

Re: [asterisk-users] Zaptel slow dial out - TDM400P

On 10/2/07, Ken Williams <[EMAIL PROTECTED]> wrote: > Any suggestions would be greatly appreciated. Try removing all the echo cancel stuff just to see if that makes any difference at all. ___ --Bandwidth and Colocation Provided by http://www.api-digital