Re: [Asterisk-Users] Wits end with echo

Richard Scobie wrote: Jon Reynolds wrote: I have updated the phones to 1.0.12 firmware, I have echotraining=800, echocancel=yes, echowhenbridged=yes, in my sip.conf file. I am using Mark2 as the echo suppresion and still I have echo. Is this correct? I do not believe having these echo par

Re: [Asterisk-Users] CentOS vs. Vanilla Kernel

Thanks to all for their advice on this thread. I'll stick with the default CentOS kernel. Julian. Ryan Amos wrote: The default CentOS kernel has worked fine for me. Just an FYI; CentOS uses the RedHat EL kernel source to build... It's pretty heavily patched so if you want to use the latest st

Re: [Asterisk-Users] Wits end with echo

Jon Reynolds wrote: Hello, I have an AAH-1.5 with a TMD400P with four lines, 8 Grandstream GXP-2000 phones, I am having echo issues on the GXP-2000 side. Here is what I have tried so far: The server has everything in the bios turned off except what is needed, USB, LPT, Serial etc,etc. I

[Asterisk-Users] chan_iax2: ast_sched_runq

Today I updated a couple of TDM400 based asterisks to the latest CVS head and started seeing the following messages. The update prior to today was a couple of weeks ago. -- Starting simple switch on 'Zap/6-1' -- Executing Dial("Zap/6-1", "IAX2/[EMAIL PROTECTED]/0") in new stack -- C

[Asterisk-Users] Test environment (Windows Softphone)

Hi! I want to test asterisk with about 10 Softphones (on windows) with just one windows machine (in the best case). I'm thinking of a softphone, that I can run in multiple instances on one computer and that can be configured to play a file on an incoming call or to make a call after some time...

Re: [Asterisk-Users] Initial release of AMPortal Debian/Xorcom-Rapidpackages

On Wed, Nov 09, 2005 at 02:14:18PM -0700, Claudio Canseco wrote: > Hi Tzafrir, > I tried to install amportal from deb packages, but I'm getting the same > errors as Jose. I used the new source links for apt, as told on the rapid > mail list: > > deb http://rapid.dotsrc.org/ unstable/ > deb-src ht

Re: [Asterisk-Users] CentOS vs. Vanilla Kernel

Not a problem that I've had :) Linux foxtrot.tessera.co.uk 2.6.9-22.0.1.EL #1 Thu Oct 27 12:26:11 CDT 2005 i686 i686 i386 GNU/Linux Opened pseudo zap interface, measuring accuracy... 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100.00% 100

[Asterisk-Users] Can't create iax channel

Hi all Could somebody please give me an idea as to whats wrong here. I'm trying to connect 2 servers using IAX, I'm not trunking them because I read that you need zaptel hardware installed at both sides to do the trunking. Theregistration seems to have worked as the output of iax show peers o

Re: [Asterisk-Users] DTMF method AVT

Thanks for confirmation. I changed the DTMF to ATV without even knowing that it is FRC2833 and it start working perfectly. -- #Joseph On Thu, 2005-11-10 at 01:35 -0500, BJ Weschke wrote: > AVT is RFC2833. > > On 11/9/05, Joseph <[EMAIL PROTECTED]> wrote: > > What kind of DTMF method signaling

Re: [Asterisk-Users] groupware + unified messagerie +Asterisk

Hi, I guess you know this project, but just in case: http://jivesoftware.org/asterisk-im/ IMHO, Egroupware would be best groupware solution to start on, but they have little interest in doing that (searching their mailing list for "voip" returned 2 hits...). We will gradually start working

Re: [Asterisk-Users] MusicOnHold does not play

Matt Riddell wrote: > Stephen Bosch wrote: > >>Hello: >> >>Forgive me if this is documented somewhere. I've been on >>www.voip-info.org and the Asterisk website and done no end of searches. >> >>Music on hold stops without playing any sound whatsoever. Here is >>Asterisk output: > > > Do you hav

Re: [Asterisk-Users] DTMF method AVT

AVT is RFC2833. On 11/9/05, Joseph <[EMAIL PROTECTED]> wrote: > What kind of DTMF method signaling is "AVT" ? > > My Sippura seems to support only InBand, AVT, INFO, InBand+Info, Auto > INFO does not work with Asterisks voicemail system so it is useless for > me. > > InBand - I have a problem wit

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 call

[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] Re: sipphone for freebsd

On 11/10/05 08:52 Pablo Allietti said the following: yes but both of them have problem with voice. some skype too anybody can have this problems in freebsd? i hear cutted conversations`: perhaps there's contention for your sound/mic devices. what does the hw.snd.pcm0.vchans say, also what's

RE: [Asterisk-Users] Intel Desktop MotherBoards *NOT* Unsuitable for Digium Boards

Forrest: Any secondary effects you can see from running SP on an SMP kernel, any bitching from dmesg at boot? Cool hack. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists

Re: [Asterisk-Users] Help with dialplan to allow breakout to DISA

On Tue, Nov 08, 2005 at 07:38:20AM -0500, Frank Tarczynski exclaimed: >Since this is my DID, I want the line to ring as normal but allow a user >to breakout and ultimately get an outgoing line. > >In this way the DID line would function as a normal telephone line. A >point lost on many responde

Re: [Asterisk-Users] Intel Desktop MotherBoards *NOT* Unsuitable for Digium Boards

George Pajari wrote: To make a long story short, according to Intel Dealer Technical Support (we became Intel dealers in order to get answers to our questions) there is no Intel motherboard that permits the IRQs to be configured uniquely. They are all hardwired and shared. This information ap

Re: [Asterisk-Users] Changes from 1.2beta2 to 1.2RC-1

Rod Bacon wrote: > I've spent some time clicking my way around the digium website, but > can't seem to locate a list of changes from * 1.2beta2 to 1.2RC-1. > > Can anyone point me in the right direction? http://www.sineapps.com/news.php?rssid=1093 -- Cheers, Matt Riddell __

Re: [Asterisk-Users] groupware + unified messagerie +Asterisk

harry gaillac wrote: > it's no what i expect the easier solution you provide > the more customers you get ! Indeed. However, I tend to be of the opinion that you should have enough money in the bank for a full year of wages for someone if you take on extra staff. While this may make my growth s

Re: [Asterisk-Users] Asterisk Fax support using T.38

Lilantha Karunaratne wrote: > Thanks for the URL pointer. Apparently this is ‘live’ project and we do > not think we could do any testing on that but will do these tests > internally I suppose. > > > > Anyone using * with T.38 on a commercial platform? Only in testing. If I get it working ton

Re: [Asterisk-Users] DTMF detection in TE406P ??

We disabled it in the wct4xxp.c driver as well and everything has worked great since then. I have heard from support to try raising the threshold value while leaving hardware DTMF detection on and it didn't help raising it by 100. In the big picture, doing hardware DTMF detection gives you an incr

Re: [Asterisk-Users] MusicOnHold does not play

Stephen Bosch wrote: > Hello: > > Forgive me if this is documented somewhere. I've been on > www.voip-info.org and the Asterisk website and done no end of searches. > > Music on hold stops without playing any sound whatsoever. Here is > Asterisk output: Do you have mpg123 installed? Is it versi

Re: [Asterisk-Users] DTMF detection in TE406P ??

Kevin P. Fleming wrote: There are other steps that can be taken if necessary first. Can you please elaborate on this? It may just save a lot of calls to Digium support about the same issue. (I have noticed this sporadically). ___ --Bandwidth and C

[Asterisk-Users] Changes from 1.2beta2 to 1.2RC-1

I've spent some time clicking my way around the digium website, but can't seem to locate a list of changes from * 1.2beta2 to 1.2RC-1. Can anyone point me in the right direction? -- == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Me

[Asterisk-Users] long calls on same channel

A customer had a question concerning long calls on his L.D. bill. The two calls in question on the L.D. bill match the mysql output. The calls are going out over a regular T1 outgoing only . Any ideas ? Asterisk box w/ TE405P spand1 ami d4 ksfxs > channel bank > single line phones

Re: [Asterisk-Users] Intel Desktop MotherBoards Unsuitable for Digium Boards

On Wednesday 09 November 2005 14:38, Mojo with Horan & Company, LLC wrote: > This is exactly what I did on a mobo that shared irqs without recourse, > and it caused me to find out that if you disable the audio device, > ztmonitor fails as it requires /dev/dsp. So get your gains and echo > problems

Re: [Asterisk-Users] DTMF detection in TE406P ??

On Wednesday 09 November 2005 21:46, Boris Bakchiev wrote: > I noticed that the latest wct4xxp sources allow disabling DTMF support > in VPM modules. It just disables the VPM's capability to detect DTMF; DTMF detection still works within Asterisk as it'll be done on the host CPU (i.e. the old way

Re: [Asterisk-Users] Asterisk Fax support using T.38

Since this is for a commercian project, you should not promise your employer/client any faxing with FoIP using Asterisk, since this will come back to haunt you, instead use POTS/TDM for your faxing. The best solution will be to use a channel bank for the faxing, and make sure that it doesn't use an

Re: [Asterisk-Users] DTMF detection in TE406P ??

Boris Bakchiev wrote: PRI<->TE406P SPAN1<->TE406P SPAN3<->PABX Please contact Digium technical support. I noticed that the latest wct4xxp sources allow disabling DTMF support in VPM modules. There are other steps that can be taken if necessary first. ___

Re: [Asterisk-Users] Problems with HINT

Upgrade to asterisk-1.2.0-rc1 and ensure that your sip file contains subscribecontext=sip-text. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listin

[Asterisk-Users] DTMF detection in TE406P ??

Hi,     I’m getting a lot of false DTMF detections on my system. Following is a diagram of my system: PRI<->TE406P SPAN1<->TE406P SPAN3<->PABX   Basically anyone talking to me with a higher pitch voice (Ladies) I get “beeps” all over the place.   If I unplug PRI from Asterisk and p

RE: [Asterisk-Users] Asterisk Fax support using T.38

Thanks for the URL pointer. Apparently this is ‘live’ project and we do not think we could do any testing on that but will do these tests internally I suppose.   Anyone using * with T.38 on a commercial platform?     Cheers!     Lilantha   From: [EMAIL PROTECTED]

RE: [Asterisk-Users] Asterisk Fax support using T.38

Yes while we agree on your point, if the requirement is to relay faxes over a VoIP network by an operator, then we need to use FoIP methods via T.37 / 38. Do you know of anyone using * with T.38 on a commercial implementation?     Cheers!     Lilantha   From: [EMAIL

Re: [Asterisk-Users] CentOS vs. Vanilla Kernel

[EMAIL PROTECTED] wrote on 11/07/2005 01:17:31 PM: > HW: HP DL360 1GB Ram Single 3GHz Pentium (with Hyper-threading turned off). > OS: CentOS 4.2 > Dual Embedded NIC enabled > USB disabled > serial disabled > printer disabled > 2x73GB SCSI in HW Raid 1 > > What is the opinion of this fine list  

Re: [Asterisk-Users] Re: SNOM360 & Monitoring Extension States

Peter Dean wrote: if you have the following; exten => _226,hint,SIP/226&SIP/101 There is no need for a _ prefix here, by the way, you are not using a pattern match. Still not able to pick the call up off of a flashing light though. Did someone tell you that has been implemented? It hasn'

Re: [Asterisk-Users] Re: SNOM360 & Monitoring Extension States

Just as an additional update to the hints for the SNOM360. if you have the following; exten => _226,hint,SIP/226&SIP/101 then if the monitoring phone dials out, all the lights associated with a monitored phone will light up. the following appears to work with * v1.2.0-rc1; exten => _226,hint,SIP/

Re: [Asterisk-Users] upgrade to 1.2 beta 2 issue

[EMAIL PROTECTED] wrote: I'm also using about 16 of the IAXy devices on this box, so I'm getting quite a few of these messages. It's been fixed in CVS HEAD and will be so in the 1.2 release. ___ --Bandwidth and Colocation sponsored by Easynews.com

RE: [Asterisk-Users] Cisco DHCP and Polycom boot server

When you set up the DHCP pool in Cisco you need to use syntax like: --> option 66 ascii "a.b.c.d" Peter > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Noah Miller > Sent: Thursday, 10 November 2005 9:08 AM > To: Asterisk Users Mailing List - N

Re: [Asterisk-Users] Asterisk 1.2.0-RC1 Crashing with g729 codec and ATA 186

Asterisk Users Mailing List - Non-Commercial Discussion on November 9, 2005 at 9:32 PM -0400 wrote: Do you know if I can get it to work with both my Cisco 12 SP+ and my ATA-186? Oops. I think I see my answer in the wiki.  http://www.voip-info.org/wiki/view/chan

[Asterisk-Users] Siemens optiPoint 420 Advance and Economy

Does anyone knows where I can find the SIP firmware for these phones? Can these phones support the HINT feature from asterisk? Thanks and regards, Stephen ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asteris

[Asterisk-Users] DTMF method AVT

What kind of DTMF method signaling is "AVT" ? My Sippura seems to support only InBand, AVT, INFO, InBand+Info, Auto INFO does not work with Asterisks voicemail system so it is useless for me. InBand - I have a problem with this one when I try to connect to a bank automated systems (some of them d

Re: [Asterisk-Users] upgrade to 1.2 beta 2 issue

On Nov 9, 2005, at 2:32 PM, Kevin P. Fleming wrote: Paul Dugas wrote: I run CVS a the house and have been getting these for quite some time now. I have an old-model IAXy that has been misbehaving in this manner for months. I've become desensitized ;) It may be a bug in chan_iax2 or the

Re: [Asterisk-Users] iax2 config sanity check

> > "Brian" == Brian Capouch <[EMAIL PROTECTED]> writes: > > Brian> The username and the peer name aren't the same thing. > Brian> There is some ambiguity floating around as to just how the > Brian> syntax parses out fully. > > Brian> Use the username, microcomaustralia (ugh.

Re: [Asterisk-Users] Asterisk 1.2.0-RC1 Crashing with g729 codec and ATA 186

Sergio Chersovani <[EMAIL PROTECTED]> on November 9, 2005 at 8:56 PM -0400 wrote: Gervais de Montbrun ha scritto: > Asterisk has been crashing like crazy since trying to run the latest > RC-1 version and it seems to crash every time I try to use my Cordless > phone. I have set the ATA to use

[Asterisk-Users] MusicOnHold does not play

Hello: Forgive me if this is documented somewhere. I've been on www.voip-info.org and the Asterisk website and done no end of searches. Music on hold stops without playing any sound whatsoever. Here is Asterisk output: > Asterisk Ready. > *CLI> -- Starting simple switch on 'Zap/4-1' > Nov 10

Re: [Asterisk-Users] iax2 config sanity check

> "Brian" == Brian Capouch <[EMAIL PROTECTED]> writes: Brian> The username and the peer name aren't the same thing. Brian> There is some ambiguity floating around as to just how the Brian> syntax parses out fully. Brian> Use the username, microcomaustralia (ugh. that name is

Re: [Asterisk-Users] Realtime Voice Changer Patch

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Matt Riddell wrote: > LOL!!!: > > Trend SMEX Content Filter has detected sensitive content. > > Sender = Matt Riddell > Subject = Re: [Asterisk-Users] Realtime Voice Changer Patch > Policy = Sexual Discrimination That's the silliest thing I've ever

Re: [Asterisk-Users] how to setup Agent dialing in multiple asterisk servers

Of course we use IAX to dial agents logged in another asterisk server.   Ex. I have three asterisk servers pbx1, IPaddress xxx.xxx.xxx.2 pbx2, IPaddress xxx.xxx.xxx.4 pbx3, IPaddress xxx.xxx.xxx.6   Here's a portion of our dialplan when I dial agent/1XXX from pbx1:   ; attempt to dial agent from wi

RE: [Asterisk-Users] Avaya 4612 IP phones with Asterisk?

According to the latest SIP release notes from Avaya, the 46xx series work with the SIP images here: http://support.avaya.com/japple/css/japple?PAGE=ProductArea&temp.productID=107755&temp.bucketID=108025 As previously mentioned you need to upgrade to SIP. I'm using 4610sw's. It doesn't seem tha

Re: [Asterisk-Users] Cisco 7940 - TFTP

Kris Edwards wrote: Hey all. I recently got the above mentioned phone and am having trouble upgrading the firmware. I have the sip firmware (availabe but not loaded), but wouldn't mind using sccp it's just that I can't even get the config file to load via tftp. The phone loops requesting CTLSEP.

Re: [Asterisk-Users] Asterisk 1.2.0-RC1 Crashing with g729 codec and ATA 186

Gervais de Montbrun ha scritto: Asterisk has been crashing like crazy since trying to run the latest RC-1 version and it seems to crash every time I try to use my Cordless phone. I have set the ATA to use the g729 codec that I purchased from Digium. Below is an example of a debug output from m

Re: [Asterisk-Users] force to expire a sip registration

Jason Pyeron wrote: On Thu, 10 Nov 2005, Matt Riddell wrote: Jason Pyeron wrote: without restarting * how can I purge only 400b? I.E. remove it, reload Asterisk, re add it, reload again. That really falls under not restarting Asterisk, but thanks. Um, a reload is not a restart. A re

Re: [Asterisk-Users] Intel Desktop MotherBoards Unsuitable for DigiumBoards

George Pajari wrote: As to those who continue to doubt the stupidity of Intel's mobo designers, here are the salient excepts of an lspci -vb from an "Intel Desktop Board D915GVWB" machine with a TDM04B board: And the saga continues... 'lspci -vb' does not understand IO-APIC mode, as best I c

[Asterisk-Users] Getting SpeedDial buttons to work on a Cisco 12 SP+

Hi all, Does anyone know how I can configure the SpeedDial buttons on my Cisco 12 SP+? I've searched through the list archives and do not see anyone talking about it. I'm running the RC1 of Asterisk and I see Asterisk pass a template to my phone, but where can I configure this template? Any h

Re: [Asterisk-Users] Realtime Voice Changer Patch

On Thu, 2005-11-10 at 12:49 +1300, Matt Riddell wrote: > > The frequencies are generally between 8-14Hz. While these frequencies > > by themselves generally cannot pass a telephone network (300-3300Hz US > > 300-3400Hz EU typically) when mixed with other frequencies they can be > > passed. This i

Re: [Asterisk-Users] force to expire a sip registration

On Thu, 10 Nov 2005, Matt Riddell wrote: Jason Pyeron wrote: without restarting * how can I purge only 400b? I.E. remove it, reload Asterisk, re add it, reload again. That really falls under not restarting Asterisk, but thanks. -- -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=

Re: [Asterisk-Users] Realtime Voice Changer Patch

On Thu, 2005-11-10 at 12:22 +1300, Matt Riddell wrote: > LOL!!!: > > Trend SMEX Content Filter has detected sensitive content. > > Sender = Matt Riddell > Subject = Re: [Asterisk-Users] Realtime Voice Changer Patch > Policy = Sexual Discrimination > Action on this mail = Delete message > Um yea

Re: [Asterisk-Users] Unsuccessful Native Bridge Between Zap Channels

Andrew Kohlsmith wrote: being unable to native bridge between two different technologies makes sense... but why wouldn't you be able to native bridge between two Zap channels? If DTMF is required for call features, usually. If one of them is already in a three-way call that can also be a r

Re: [Asterisk-Users] Wits end with echo

This pretty much helped me with the rxgains and txgains: http://lists.digium.com/pipermail/asterisk-users/2004-November/071301.html you don't use ztmonitor directly to modify the rx and txgains, you just use it as a meter to make modifications to the zapata.conf file. Moj Jon Reynolds wrote:

[Asterisk-Users] Asterisk 1.2.0-RC1 Crashing with g729 codec and ATA 186

I have an Asterisk setup for my SOHO office. It only has two phones on it right now. Although it used to have a few... Right now it has one old Cisco 12 SP+ using the the skinny protocol and a cordless phone attached to a Cisco ATA-186. Asterisk has been crashing like crazy since trying to run

Re: [Asterisk-Users] Asterisk Fax support using T.38

Andy Kuo wrote: > Lilantha > > I've been looking for fax solutions with Asterisk too. Unfortunately, > it seems like there's no T.38 support for Asterisk so far. In fact, I > think there's only fax-to-email solution for * now. > > I'm getting some SIP ATA's with T.38 support next week, but I a

Re: [Asterisk-Users] Realtime Voice Changer Patch

trixter aka Bret McDanel wrote: > I will give you one of my super secret someday projects. A voice stress > analyzer built into asterisk (would be done in a similar way in terms of > interfacing, perhaps just play a beep or if non voice only circuit send > text). :) Nice! > VSAs work by detecti

Re: [Asterisk-Users] R2-Digital (Q.421)

Hi Jesus, The Cisco kit, and one or two other products, offer an R2 digital using DTMF mode, but this is the first time I have heard of it being used. The spec for this is definitely not Q.421. That spec does not mention DTMF at all. R2 using DTMF doesn't appear to be in the ITU specs, as far

[Asterisk-Users] Re: sipphone for freebsd

On Wed, Nov 09, 2005 at 01:20:47PM +0800, Dinesh Nair wrote: > > > On 11/09/05 07:17 Pablo Allietti said the following: > >Hi all > >anybody can tell me what sipphone are available for Freebsd? > > /usr/ports/net/kphone > /usr/ports/net/linphone yes but both of them have problem with voice. so

[Asterisk-Users] Error compiling app_rxfax on 1.2-rc1

I was trying to install the new RC1 version of 1.2 and I get the following error: In file included from app_rxfax.c:15: ../include/asterisk/file.h:27:2: error: #error You must include stdio.h before file.h! In file included from app_rxfax.c:15: ../include/asterisk/file.h:56: error: syntax err

Re: [Asterisk-Users] Realtime Voice Changer Patch

LOL!!!: Trend SMEX Content Filter has detected sensitive content. Sender = Matt Riddell Subject = Re: [Asterisk-Users] Realtime Voice Changer Patch Policy = Sexual Discrimination Action on this mail = Delete message -- Cheers, Matt Riddell ___ http:

Re: [Asterisk-Users] dell and digium hardware

I bought a PCI-e Areca 1210 SATA II raid controller. Who knows what Dell were thinking when they decided to stick a PCI-e slot in the system. http://www.areca.com.tw/products/html/pciE-sata.htm Craig - Original Message - From: "Brian Roy" <[EMAIL PROTECTED]> To: "Asterisk Users Maili

Re: [Asterisk-Users] force to expire a sip registration

Jason Pyeron wrote: > On Wed, 9 Nov 2005, Eric "ManxPower" Wieling wrote: > >> Olle E. Johansson wrote: >> >>> Jason Pyeron wrote: >>> take for example a phantom SIP/400b from a previos phone config, without restarting * how can I purge only 400b? testserver*CLI> sip show

Re: [Asterisk-Users] Intel Desktop MotherBoards Unsuitable for DigiumBoards

As the originator of this thread I would like to respond to some of the posts so far. 1. To the person who asked if I was referring to Intel boards or chipsets. I said boards and I meant boards. I have done no research to determine if the limitation extends to boards based on Intel chipsets m

Re: [Asterisk-Users] Intel Desktop MotherBoards Unsuitable for DigiumBoards

Justin Tunney wrote: > A little off-topic but I actually got 6 TDM400P cards in one system to > all be on their own interrupts using this "P4 Photon V1.01P" 865chip > board. The last card shared interrupts with some other stuff that > wasn't interrupt intensive. The system was stable ringing 12 p

Re: [Asterisk-Users] Zap/TDM400p with old phone.

On Nov 8, 2005, at 11:52 PM, Eric "ManxPower" Wieling wrote: Tom Rymes wrote: Folks, I am trying to use and **OLD** Automatic electric phone with my Asterisk server by plugging it into an FXS port on my TDM400p. The phone works great when plugged into a POTS line from Verizon. However,

Re: [Asterisk-Users] Realtime Voice Changer Patch

On Thu, 2005-11-10 at 11:32 +1300, Matt Riddell wrote: > Justin Tunney wrote: > > I have just written a patch for Asterisk 1.2.0-rc1 that allows you to > > install a voice changer on a channel. > > > > http://www.lobstertech.com/voicechanger/ > > > > If you are a developer, please feel free to he

Re: [Asterisk-Users] R2-Digital (Q.421)

Jesus Mogollon wrote: Hi Steve: Thanks for your help. I really appreciate it.. My provider is CANTV in Venezuela. There's a venezuelan variant in the code and I'm using that. Incoming works perfectly, outgoing is not working. I'm being told that incoming is MFCR2 but outgoing is R2-Digital with

[Asterisk-Users] Wits end with echo

Hello, I have an AAH-1.5 with a TMD400P with four lines, 8 Grandstream GXP-2000 phones, I am having echo issues on the GXP-2000 side. Here is what I have tried so far: The server has everything in the bios turned off except what is needed, USB, LPT, Serial etc,etc. I have uncommented Echo

Re: [Asterisk-Users] Asterisk Fax support using T.38

T.37 actually isnt that bad. When its 'fax' (ie analog data passed to represent data) its TDM, however to cross the internet and reap cost savings its effectively mime encoded and sent via SMTP. This gives you the TDM capabilities (no jitter, low latency, all the things faxes like). The combinat

[Asterisk-Users] system command vs mailfax and quotes?

hello, been playing with spandsp, and rxfax.  seems to work well. i have not been able to run the mailfax command successfully from within asterisk. [jeff-fax-in] exten => s,1,SetVar(FAXFILE=/var/spool/asterisk-fax/${UNIQUEID}.tif) exten => s,2,rxfax(${FAXFILE}) exten => h,1,system(/usr/bin/mail

Re: [Asterisk-Users] Asterisk Fax support using T.38

Lilantha I've been looking for fax solutions with Asterisk too.  Unfortunately, it seems like there's no T.38 support for Asterisk so far.  In fact, I think there's only fax-to-email solution for * now.   I'm getting some SIP ATA's with T.38 support next week, but I am not sure if I can somehow g

Re: [Asterisk-Users] Receptionist phones

With CVS, you also get a flashing led when the phone is ringing.   PaulH - Original Message - From: Colin Anderson To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Thursday, November 10, 2005 4:24 AM Subject: RE: [Asterisk-Users] Receptionist

RE: [Asterisk-Users] Intel Desktop MotherBoards Unsuitable for Di gium Boards

I've seen it happen in slimline Compaqs, where the IRQ on the PCI slot was shared with the onboard LAN. That'd be stupid stupid design for a 1U server box, but, hey, I wouldn't put it past Dell (shudder) -Original Message- From: Matt Riddell [mailto:[EMAIL PROTECTED] Sent: Wednesday, Novem

Re: [Asterisk-Users] what is the role of trunk=yes

From http://www.asteriskguru.com/tutorials/iax_conf.html trunk yes | no If set to yes,it will be used IAX2 trunking for this context. IAX2 trunking basically saves bandwidth by taking the frames from multiple simultaneous calls and merging them into the same outbound packet. Both sides must su

Re: [Asterisk-Users] what is the role of trunk=yes

On Wed, 2005-11-09 at 22:18, jonny hashem wrote: > Hi: > Can any body tell me what is the role of the trunk=yes > property that exists in iax.conf? http://www.voip-info.org/wiki/index.php?page=Asterisk+bandwidth+iax2 Rgds Pete ___ --Bandwidth and Colo

Re: [Asterisk-Users] Asterisk Fax support using T.38

Sure, why not. :) On 11/9/05, Cory Andrews <[EMAIL PROTECTED]> wrote: > Nice analogy, can I borrow that one? > > Cory J Andrews > Partner / Purchasing > +++ > VOIPSupply.com - Everything you need for VOIP > 454 Sonwil Drive > Buffalo, NY 14225 > +++ > tf voice - 800-398-VOI

Re: [Asterisk-Users] Problems with HINT

Pls update to rc1 and if you're still having the same problem, open a bug on bugs.digium.com so we can get it fixed before release. On 11/9/05, Alvaro Parres <[EMAIL PROTECTED]> wrote: > Hi list: > >I'm configuring in a CVS-HEAD asterisk ( about 2 weeks ago), some hint > extension. But they a

Re: [Asterisk-Users] Intel Desktop MotherBoards Unsuitable for Digium Boards

Yup. I've been corrected already. :) I guess it's more dependent on the chipset than the proc. On 11/9/05, Leo Ann Boon <[EMAIL PROTECTED]> wrote: > BJ Weschke wrote: > > > No. APIC was in 2.4 as well, but you need an Intel CPU in there (I > >think) in order to be able to take advantage of it.

Re: [Asterisk-Users] Realtime Voice Changer Patch

Justin Tunney wrote: > I have just written a patch for Asterisk 1.2.0-rc1 that allows you to > install a voice changer on a channel. > > http://www.lobstertech.com/voicechanger/ > > If you are a developer, please feel free to help add features and > clean up the patch so that we can hopefully get

Re: [Asterisk-Users] Kapanga SoftPhone HOWTO

Ben Higley wrote: > HI All: > > I have downloaded the Kapanga Softphone http://www.kapanga.net > > and it works great for audio. > > Im trying to enable the video portion... > > I enabled h263;h261;h263p in the allow section of my sip_buddies > > and the phone registers.. > > does someone hav

Re: [Asterisk-Users] Intel Desktop MotherBoards Unsuitable for Di gium Boards

Andrew Kohlsmith wrote: > On Wednesday 09 November 2005 03:29, Matt Riddell wrote: > >>Colin Anderson wrote: >> >>>Onboard LAN with an un-movable IRQ would mess that up good >> >>Only if you had just one pci slot. > > With 1U systems that is often all you get. It'd be pretty sad if they only had

[Asterisk-Users] what is the role of trunk=yes

Hi: Can any body tell me what is the role of the trunk=yes property that exists in iax.conf? Regards; jonny __ Yahoo! FareChase: Search multiple travel sites in one click. http://farechase.yahoo.com _

[Asterisk-Users] Script for load testing

Guys. Do any have some already made scripts for load testing or creating lots of calls for load testing an asterisk install? Wanted to check with you first, since probably somebody has done this before. ___ --Bandwidth and Colocation sponsored by Easyn

[Asterisk-Users] Cisco DHCP and Polycom boot server

Hi - I've been trying to set up my Polycom phones to get the boot server info (tftp-server-address) from DHCP on a Cisco router. I've previously just specified it manually on the phone, and that works well enough, but I need to change now (because of the number and geographic locations of the ph

Re: [Asterisk-Users] Asterisk Fax support using T.38

Nice analogy, can I borrow that one? Cory J Andrews Partner / Purchasing +++ VOIPSupply.com - Everything you need for VOIP 454 Sonwil Drive Buffalo, NY 14225 +++ tf voice - 800-398-VOIP X22 l voice - 716.630.1555 X22 f - 716.630.1548 e - [EMAIL PROTECTED] AIM - b2Cory C

RE: RE : [Asterisk-Users] asterisk-1.2-bêta2 | presence/subscription support in the SIP channe l driver

Olivier, Oui !!! pour asterisk ou openpbx. SER est un excellent proxy sip ! Il est evident qu' SER n'offre pas les fonctionalités d'un ipbx. je ne pense pas que toneec soit viable , combien d'opérateurs offrent ces services (Skype)... Votre pojet stagne! Vous avez fait le choix de beacoups d'I

Re: [Asterisk-Users] Intel Desktop MotherBoards Unsuitable for DigiumBoards

A little off-topic but I actually got 6 TDM400P cards in one system to all be on their own interrupts using this "P4 Photon V1.01P" 865chip board. The last card shared interrupts with some other stuff that wasn't interrupt intensive. The system was stable ringing 12 phones 24/7 in 100 degree heat

RE: [Asterisk-Users] (Resolved) A2Billing PIN does not get registered -keeps getting prompted

Hi all     Was able to resolve issue by using another softphone (sjphone). Was using adoresoft phone  - will continue testing with adoresoftphone to try to resolve dtmf issue.   Thanks again areski   From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Omar M

[Asterisk-Users] Problems with HINT

Hi list:    I'm configuring in a CVS-HEAD asterisk ( about 2 weeks ago), some hint extension. But they are not working:    /etc/asterisk/extensions.conf       [sip-test]    exten => 116,1,dial(SIP/116)    exten => 112,hint,SIP/112    exten => 112,1,dial(SIP/112)       /etc

Re: [Asterisk-Users] Intel Desktop MotherBoards Unsuitable for Digium Boards

BJ Weschke wrote: No. APIC was in 2.4 as well, but you need an Intel CPU in there (I think) in order to be able to take advantage of it. AMD's don't have this option available. No necessary true. The nVidia nForce chipset for AMD Athlon supports IOAPIC, and does work with 2.4. Dual processo

RE: [Asterisk-Users] strange tone is droping calls

Ok Lets hope I do have polarity reversals.. |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Matt Riddell |Sent: Wednesday, November 09, 2005 2:34 AM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] str

Re: [Asterisk-Users] Test environment for a Predictive Dialer

Hello, Yes, we have this kind of test setup in our office using a quad T1 card in each of two servers and cross-connecting each T1 to the other to test VICIDIAL for performance and compatibility. A few notes, if you are going to use a PRI between the two servers make sure you set one to pri_cpe a

Re: [Asterisk-Users] groupware + unified messagerie +Asterisk

it's no what i expect the easier solution you provide the more customers you get ! --- Matt Riddell <[EMAIL PROTECTED]> a écrit : > harry gaillac wrote: > > What about egroupware ! > > We use it, although there is no simple click to > install installation package > for Asterisk integration. > >

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