Re: [asterisk-users] remove

Check the footer at the bottom of this message for instructions on how to unsubscribe :-) Matthew Fredrickson On Fri, Jun 1, 2018 at 12:11 PM, David Mutterer wrote: > > -- > _ > -- Bandwidth and Colocation Provided by

[asterisk-users] Testing...

Test from non-digium email. -- Matthew Fredrickson Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out

[asterisk-users] Test from Digium address

Testing again. -- Matthew Fredrickson Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new

[asterisk-users] One more test

I need to send one more test. Here it is! -- Matthew Fredrickson Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com

[asterisk-users] More testing

More testing. Test test test. :-) -- Matthew Fredrickson Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check

[asterisk-users] Test

Testing again :-) -- Matthew Fredrickson Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new

[asterisk-users] AstriCon Approaching, Super Earlybird Pricing Expires In 3 Days

Hey All, So one of the jobs that I get to do as head of the Asterisk project is to help inform people about the yearly conference we have about Asterisk named Astricon. For those who are not familiar with it, AstriCon is a fantastic event for anyone that is serious about Asterisk. This year,

Re: [asterisk-users] Wanted: WebRTC tutorial

On Tue, Apr 24, 2018 at 10:54 AM, Bruce Ferrell wrote: > A while back (last year maybe?), there was a Digium blog post on setting up > WebRTC. > > I was never able to get that working. > > I was working with Asterisk 15 on a RHEL derived distro and had no idea of > where to

[asterisk-users] Asterisk Community Services Outages

Dear Asterisk Community, For the past 24 hours or so, Digium’s upstream provider has had a few outages that have affected Asterisk community services, including Asterisk.org, the mailing lists, and potentially other services. We apologize for any inconvenience that it has caused. Hopefully

Re: [asterisk-users] Strange problem with PRI on 64-bit?

On Tue, Apr 3, 2018 at 4:38 PM, Tony Mountifield wrote: > In article >

Re: [asterisk-users] Strange problem with PRI on 64-bit?

On Tue, Apr 3, 2018 at 5:44 AM, Tony Mountifield wrote: > I have some more investigation to do on this, but I wanted to see if anyone > here had any insight into the issue I've run into. > > The hardware is a HP DL360 G6 with a TE420 gen 5 4-port T1 PRI card. It is one > of

Re: [asterisk-users] More testing - sorry guys

Thanks :-) On Wed, Mar 28, 2018 at 3:52 PM, Markus Weiler <markus_wei...@mailworks.org> wrote: > I received it :-) > > > Am 28.03.2018 um 22:44 schrieb Matt Fredrickson: >> >> Just a test. >> > > > -- >

[asterisk-users] Sorry for interruption of service

Hey All, Just as a public service announcement, we had a 12-16 hour window with mailing list service interruption. Last night we scheduled a time to update the mailing list server but today found some problems impacting mailing service after the updates. Due to this discovery, we quickly

[asterisk-users] More testing - sorry guys

Just a test. -- Matthew Fredrickson Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk

Re: [asterisk-users] AMI potential memory leak

On Wed, Mar 21, 2018 at 4:03 PM, Dan Cropp wrote: > We are communicating with Asterisk via AMI. Running Asterisk version > 13.18.5 on an Ubuntu box. > > > > If you look at the event response, the Result field is filled with random > characters. I’m not sure what to do because

[asterisk-users] Test

Testing, 1, 2, 3. -- Matthew Fredrickson Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new

[asterisk-users] Test

This is a test. -- Matthew Fredrickson Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new

[asterisk-users] FOSDEM

Hey All, For any interested in potentially meeting up to talk about Asterisk and other fun things, Ben Ford from Digium's Asterisk development team and myself will be in Brussels for FOSDEM Feb 3-4. I hope to see many of you there! -- Matthew Fredrickson Digium, Inc. | Engineering Manager 445

[asterisk-users] Recent Video Interview and Upcoming Webinar about Asterisk 15

Hey Everybody, As a project, we would like to do a better job of getting additional information about new developments in Asterisk to the community. I think this is something I have struggled with in the past (to some extent) and would like to improve upon in the future. For anybody interested,

[asterisk-users] Asterisk EOL Announcement

Dearly Beloved, We have gathered here today to mourn the passing of a deeply regarded branch of Asterisk - Asterisk 11. As of today, it has officially reached its end of life. It was a good branch, having served 5 years faithfully in the service of its users. As far as history goes, 11.0.0 was

[asterisk-users] Asterisk 14 Security Fix Only Mode

Hey all, For those who may not be aware Asterisk 14 transitioned from bug fix mode to security-fix-only mode a few weeks ago (Sept 26th). For those of you that are still on this release, it's a good time to consider building an upgrade plan for moving to 15.x.x. I sincerely apologize for the

[asterisk-users] Asterisk 15 Beta Released

It is with great pleasure I wish to inform you of the first beta release of the new Asterisk 15 branch. It's a very exciting time to be a user of Asterisk! Asterisk 15 is arguably the biggest release of Asterisk that has happened in the last 10 or so years. There has been a lot of work done in the

Re: [asterisk-users] Moving call DAHDI from channel X to Y.

On Sun, Jul 30, 2017 at 8:34 PM, Daniel Harper wrote: > I am seeing the in the asterisk logs that channels (PRI ISDN) are > being moved .. > > [Jul 29 16:31:48] VERBOSE[16125] logger.c: -- Moving call > (DAHDI/57-1) from channel 57 to 58. > > I then see the moved

[asterisk-users] Asterisk 11 EOL 6 Month Notice

Greetings, This is your friendly 6 month warning that Asterisk 11 will be reaching an official end of life state on October 25, 2017. As many of you know, for the past 6 months Asterisk 11 has been in security fix only mode. This means it currently does not receive bug fixes, but it does

Re: [asterisk-users] Asterisk 13.15.0, webrtc, Google chrome 58 beta and "bad media description"

On Sat, Apr 8, 2017 at 7:23 AM, Dan Jenkins wrote: > > On Fri, Apr 7, 2017 at 9:44 PM, Teijo wrote: > >> Hello, >> >> I've been using webrtc (Jscommunicator) with Asterisk occasionally. Only >> problem until now which remained was that if dtls_rekey

Re: [asterisk-users] 100% CPU after upgrade.

On Mon, Apr 3, 2017 at 4:45 PM, Mike Diehl wrote: > Those are all rational questions, so here we go: > > We upgraded from 11.x, though the system was a backup server, so it was never > actually used. > > The system is a 2.4Gh quad-core Xenon with 4G of RAM, so it should

Re: [asterisk-users] 100% CPU after upgrade.

One thing you didn't mention was what version you previously upgraded from... Also, more information about the system in general would help. (Endpoints, is it realtime or flat file configured, if realtime, what type of database, what channel drivers (SIP or PJSIP, and others). Matthew

Re: [asterisk-users] Bounty on Google Voice

On Wed, Mar 29, 2017 at 11:45 AM, Saint Michael wrote: > The channel motif and res_xmpp do not work. But there is one company that > does make it work and charges $US 6 for a lifetime connection to your own > free Google Voice number, from SIP. I wonder if anybody would be able

Re: [asterisk-users] UniMRCP and Asterisk 14

On Thu, Mar 23, 2017 at 8:27 PM, Richard Kenner wrote: > When I look at the lastest UniMRCP manual, they only mention as high as > Asterisk 13. Does anybody know if I need to do anything to allow it > to work on Asterisk 14 and, if so, what that is? I can't speak for the MRCP

Re: [asterisk-users] Manager events showing in CLI

Try doing a `core set debug 0` at the Asterisk CLI. That should disable it. Or remove debug from your console output in logger.conf. Best wishes, Matthew Fredrickson On Sun, Mar 26, 2017 at 5:35 PM, Telium Technical Support wrote: > I did that too – no debug related

Re: [asterisk-users] Issue with handling of 480 DND

Response inline. On Fri, Jan 6, 2017 at 12:47 PM, Markus Weiler wrote: > Nobody any idea? > > It would be really helpful, > > Markus > > > > > Am 06.01.2017 um 12:07 schrieb Markus Weiler: > >> Hi List, >> >> we're calling a sip phone from our Asterisk Server, and

[asterisk-users] Asterisk 14 web broadcast

Hey All, Slight interlude from your regularly scheduled programming. For any interested, I will be giving a web broadcast today about Asterisk 14 and what's new with Asterisk since the 13 release. For those of you that aren't aware, I'm responsible for day to day management of the Asterisk

Re: [asterisk-users] Force hangup not working on stuck channel

Also, it looks like in https://issues.asterisk.org/jira/browse/ASTERISK-21762 there might be a workaround (see the last comment at the bottom). Matthew Fredrickson On Fri, Nov 4, 2016 at 2:01 PM, Matt Fredrickson <cres...@digium.com> wrote: > On Thu, Nov 3, 2016 at 11:16 AM, Carlos Ch

Re: [asterisk-users] Force hangup not working on stuck channel

On Thu, Nov 3, 2016 at 11:16 AM, Carlos Chavez wrote: > I am unable to force a hangup on a channel that has been stuck for over two > days: > > IAX2/from-CD-11006 oficina 27701 Up Dial > IAX2/to-CD/2883 3467130007

Re: [asterisk-users] What's the smallest, lightest Asterisk you can build? Does size even matter?

On Tue, Nov 1, 2016 at 6:00 PM, Jonathan H wrote: > All I need is PJSIP, ulaw, alaw, wav, astdb and all the dialplan functions. > > I don't need any other DB layer, I have no hardware, and I was wondering > what the smallest build possible was. > > I experimented, but

Re: [asterisk-users] RS485 Audio device

On Fri, Oct 28, 2016 at 7:09 PM, Jerry Geis wrote: > Hi All, > > Is there any devices or pair of devices that do audio over RS485 > and then convert to SIP for us in asterisk? > Of course a speaker and push button at the other end. > > Is there anything like that out there?

[asterisk-users] Asterisk 11 - Security Fix Only Notice

Hey All, This is a friendly notice that as of today Asterisk 11 has entered security fix only mode. From this point onward additional releases of Asterisk 11 will no longer be made unless there is a security fix being applied to the branch. Users of Asterisk 11 are encouraged to move to one of

Re: [asterisk-users] Got bitten by the 255 char variable limit - how best to work around it?

On Sat, Oct 22, 2016 at 8:05 PM, Jonathan H wrote: > I loop through a list in Asterisk which is generated by a Python AGI > and I've just been bitten by a variable limit I didn't realise existed > before. > > The only way I can think of working around this is to get Python

Re: [asterisk-users] Audio when card is in condition yellow

Usually a card is supposed to send yellow alarm (so it's transmitted) when it detects LOS (loss of signal) on the T1, or essentially a red alarm condition is detected. So if yellow is being sent, it means that at least one end is not able to sync up the line, which means you'll have junk/garbled

Re: [asterisk-users] queue_log/cel sqlite

On Thu, Oct 20, 2016 at 9:45 AM, marek cervenka <cerva...@gmail.com> wrote: > > Dne 20/10/2016 v 16:32 Matt Fredrickson napsal(a): >> >> On Thu, Oct 20, 2016 at 4:50 AM, marek cervenka <cerva...@gmail.com> >> wrote: >>> >>> i tested this &g

Re: [asterisk-users] queue_log/cel sqlite

On Thu, Oct 20, 2016 at 4:50 AM, marek cervenka wrote: > i tested this > > # cat /etc/asterisk/extconfig.conf > [settings] > queue_log => sqlite3,cdrDb > > # cat /etc/asterisk/res_config_sqlite3.conf > [cdrDb] > dbfile = /var/lib/asterisk/realtime.sqlite3 > > sqlite3

[asterisk-users] Asterisk 11 - Security Fix Mode

Hello Everyone, As many of you are already aware, we are rapidly approaching the time when the Asterisk 11 branch will go into what is known as security fix only mode. Up to this point, bug fixes have been included and merged into the 11 branch. For Asterisk 11, this new phase of life shall

Re: [asterisk-users] Farewell

Best of wishes to you in your retirement! It's been a great 10 years, and I'm personally looking forward to the great things coming in the next 10. Matthew Fredrickson On Wed, Aug 17, 2016 at 8:37 AM, Vincent Medina wrote: > I just wanted to wish all of you good luck I'm

Re: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ?

urrent versions of Chrome or Firefox. > That said, LetsEncrypt certs work fine for this, so no need to spend out on > one. > > Switch to Asterisk 13.10 and save yourself a whole lotta headache. > > On 11 August 2016 at 15:09, Jonas Kellens <jonas.kell...@telenet.be> wr

Re: [asterisk-users] Asterisk & Vitelity Invite issues

PM, Tammy Firefly <tammy-li...@wiztech.biz> wrote: > > > On 8/9/16 12:40 PM, Matt Fredrickson wrote: >> On Mon, Aug 8, 2016 at 9:25 AM, Tammy Firefly <tammy-li...@wiztech.biz> >> wrote: >>> Hi All, >>> >>> We have asterisk 11.23 run

Re: [asterisk-users] Original Callerid on transfer in asterisk 13

How are you attempting to view the original CallerId? Matthew Fredrickson On Wed, Aug 10, 2016 at 2:59 PM, Israel Gottlieb wrote: > Hi > Is there any configuration change in asterisk 13.9.1 to show original > callerid on a transfer > In asterisk 11.21 it works as expected > >

Re: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ?

now > after my personal experience with Asterisk 11 and webRTC. > > You also say Asterisk 13. How about Asterisk 12 then ?? > > > > Kind regards. > > > > On 10-08-16 21:53, Matt Fredrickson wrote: > > I don't see an ice-ufrag or ice-pwd line in the response from &

Re: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ?

I don't see an ice-ufrag or ice-pwd line in the response from Asterisk, correlating with your suspicion that there is no ICE. Are you sure that the stun server you're using (the google one) still works? I haven't tried that server in a while, but I distantly seem to recall that maybe they shut

Re: [asterisk-users] EAGI script with missing audio on /dev/fd/3

On Tue, Aug 2, 2016 at 11:42 AM, nik600 wrote: > Dear all > > i'm trying to access to the input audio raw stream with a very basic EAGI > script: > > > #!/bin/sh > echo "EXEC Queue 2001" > cat /dev/fd/3 > /tmp/pippo > > This is my dialplan: > > exten => 001,NoOp(test) > exten

Re: [asterisk-users] Asterisk & Vitelity Invite issues

On Mon, Aug 8, 2016 at 9:25 AM, Tammy Firefly wrote: > Hi All, > > We have asterisk 11.23 running sip to vitelity and from there IAX trunks > split off to where they need to go. We are having a problem getting > chan_sip to quit ignoring re-invites from Vitelity. Our

Re: [asterisk-users] AstriCon 2016 - XMPP and Asterisk

Looking forward to seeing you there, and hopefully to seeing your talk! Matthew Fredrickson On Tue, Aug 2, 2016 at 11:02 AM, Marcelo Terres wrote: > Going to AstriCon 2016 ? > > Don't miss my talk about how to use XMPP and Asterisk to improve the > user experience. > >

Re: [asterisk-users] Force out-bond call to specific CIC

Yes, as far as I remember, in your dial string, simply use a Dial(DAHDI/X/1234567) where X is the dahdi device channel number. Hope that helps. Matthew Fredrickson On Wed, Jul 13, 2016 at 5:22 AM, Mehdi Shirazi wrote: > Hi > > How is it possible to use Dial application

Re: [asterisk-users] Unable to create channel DAHDI

Looks like the hookstate is listed as offhook. I don't think chan_dahdi will attempt to make a call out a device that is offhook. Hope that helps, Matthew Fredrickson On Tue, Jun 7, 2016 at 1:36 PM, Brent Davidson wrote: > In trying to troubleshoot the Delay after

Re: [asterisk-users] Advices on how to evaluate voice quality in a mixed Dahdi/SIP environment ?

On Wed, May 18, 2016 at 9:44 AM, Olivier wrote: > I've got the following setup: > > PSTN ITSP SDSL Modem-Router Gateway - > Asterisk with B410P --- SIP Phones Wow. > Both SDSL Modem-Router and Gateway are managed by my ITSP. > > Some calls coming from PSTN

Re: [asterisk-users] Avaya Phones and Asterisk

On Fri, May 20, 2016 at 8:54 AM, Diogo Cosito wrote: > Dear gentlemen, how are you? > I wonder if anyone has experience with Avaya devices, 9608G and 9641GS > models, running on SIP and using TCP transport. > The calls work well, but the callerid only "pass" number of

Re: [asterisk-users] pjsip segfault problem

Have you tried updating to pjproject version 2.5.x? It should have the patch that you listed in your other email, which I believe should be included in that branch. Hope that helps, and best of luck. Matthew Fredrickson On Thu, May 26, 2016 at 4:11 AM, Marek Červenka

Re: [asterisk-users] T.38 with Audiocodes gateway

On Fri, Apr 29, 2016 at 1:34 AM, Olivier wrote: > Hello, > > I'm helping a colleague (*) which has the following setup: > > ITSP --- --- Asterisk 13 --- -- > Audiocodes MP-112 --- --- Fax machine > > My issue is the following : > Audiocodes gateway reject INVITEs with 488

Re: [asterisk-users] PRI error "ROSE REJECT"

Fredrickson On Fri, Mar 25, 2016 at 9:15 PM, Carlos Chavez <cur...@telecomabmex.com> wrote: > On 2016-03-25 16:02, Matt Fredrickson wrote: >> >> PRI debug of the entire call would be great, also, switchtype would be >> awesome as well. >> >> Thanks! >> >

Re: [asterisk-users] PRI error "ROSE REJECT"

PRI debug of the entire call would be great, also, switchtype would be awesome as well. Thanks! Matthew Fredrickson On Thu, Mar 24, 2016 at 4:07 PM, Carlos Rojas wrote: > Hi > > Did you activate the pri debug on the cli asterisk? > > On Thu, Mar 24, 2016 at 12:59 PM,

Re: [asterisk-users] OPUS support in Asterisk 13

On Thu, Mar 24, 2016 at 9:09 AM, Chirag Desai wrote: > Hi all, > > Sorry if this has been asked before. I searched a lot, but found conflicting > answers, so hoping for some clarification. > > My question is does Asterisk 13 support OPUS? If so which version exactly? Sort

Re: [asterisk-users] accept DMTF tone during ringing

For what channel driver, and what use case? It's my understanding that in the traditional telephone network (ISDN/SS7/analog), prior to a call being answered, you were not necessarily guaranteed a two way media path. Sometimes it was available (there are few stories of large companies who

Re: [asterisk-users] Fax and Asterisk

- Lee Howard [EMAIL PROTECTED] wrote: Andrew Nowrot wrote: I am trying to build reliable fax solution with asterisk, iaxmodem and hylafax. I am attempting to do this on Compaq DL-360 with 2 pentium 3 1.2 GHz (512 cache) and 2GB of RAM. I am using a Sangoma A101. After

Re: [Asterisk-Users] AstriCon 2006 Location

On Sun, Sep 18, 2005 at 11:32:00AM -0500, Brian Capouch wrote: Senad J wrote: If you are looking for the maximum number of cheap flights from around the world, and plenty of convention and room space, the answer is Las Vegas :-) I would definitively agree! Yes, but what would one do

Re: [Asterisk-Users] Seperate Incoming calls on TDM02?

On Thu, Sep 15, 2005 at 12:04:41PM -0400, C. Hatton Humphrey wrote: I have a TDM02B to bring in two POTS lines for my incoming calls; I need to point each line to a different IVR... is there somewhere that can I can look to get this setup working? Basically, each line is for a different

Re: [Asterisk-Users] PRI echo

On Mon, Sep 12, 2005 at 08:48:31PM -0600, Gabriel Gunderson wrote: Upgrade! Upgrade! Upgrade! At the very least, upgrade to the last 1.0.x. Even better would be to use the CVS-HEAD or the latest 1.2beta release of libpri, zaptel, and asterisk. There are new features that help

Re: [Asterisk-Users] PRI echo

On Sat, Sep 10, 2005 at 12:51:21AM -0700, Jason Kim wrote: My configuration is pri*(te405p)---iaxclient. My * version is 1.0.7 running on tyan dual opteron board. I have several problems. Upgrade! Upgrade! Upgrade! At the very least, upgrade to the last 1.0.x. Even better would be to

Re: [Asterisk-Users] -- PROGRESS with cause code 34 received?

On Wed, Sep 07, 2005 at 01:47:49PM +0200, Roy Sigurd Karlsbakk wrote: hi i get these messages every now and then -- PROGRESS with cause code 34 received wtf is this? Nothing to see here, move along :-) Seriously though, it's basically just and interesting message to see. The cause

Re: [Asterisk-Users] One way echo canceling?

On Thu, Sep 01, 2005 at 12:40:16PM -0400, Doug Lytle wrote: When there is a call on zap 1, from a sip phone on the remote office I have not seen it myself, but I have heard that some people have ahd trouble with overlapdial and echo cancellation. I have not been able to confirm

Re: [Asterisk-Users] One way echo canceling?

On Wed, Aug 31, 2005 at 09:46:37PM -0400, Doug Lytle wrote: When there is a call on zap 1, from a sip phone on the remote office side and typing 'zap show channel 1' shows echo cancel is on, doing the same thing from the Definity to a SIP phone shows echo cancel off. Shouldn't it be on

Re: [Asterisk-Users] TDM04b and echo

On Wed, Aug 31, 2005 at 08:45:34PM -0500, chris gamble wrote: the echo isnt horrible most of the time, and seems extremely random in that i can call a number once without echo, then dial the same number a second time and get echo. things i am currently considering (and would like to know if

Re: [Asterisk-Users] Will Echo problems EVER be solved, I'm scared

On Sun, Aug 28, 2005 at 02:52:20PM -0400, Andrew Kohlsmith wrote: On Sunday 28 August 2005 11:59, Steve Underwood wrote: I don't follow why knowing that impedance mismatch is the problem has stopped you making fxotune fix it. :-\ Where you the one who asked me how to make fxotune work well

Re: [Asterisk-Users] Will Echo problems EVER be solved, I'm scared

On Mon, Aug 29, 2005 at 03:24:01PM -0500, Ric Moseley wrote: Are changes to the zapata.conf file read on the fly or do you have to restart the asterisk process? It doesn't make any changes to the zapata.conf file. It has it's own config file that you have to set it up to load from before you

Re: [Asterisk-Users] Will Echo problems EVER be solved, I'm scared

On Fri, Aug 26, 2005 at 02:00:54PM -0600, Rich Adamson wrote: Relative to the fxotune app, it would appear the app is specific to the v2.4 kernels (/dev/zap*), which the v2.6 kernels don't use It should with 2.4 and 2.6. 2.6 kernels with properly configured udev rules should create the

Re: [Asterisk-Users] Will Echo problems EVER be solved, I'm scared

Ok, fxotune is a work in progress so to speak. I fixed something in it about a week ago that may help it adjust to the line better (whereas before I'm not sure that it was at all). Try the latest CVS-HEAD version of fxotune as your first step. (oh, after you use fxotune you should turn off your

Re: [Asterisk-Users] HDLC/Zaptel/Kernel 2.6.11(.9)

On Tue, Aug 23, 2005 at 11:47:19AM -0500, Matt Schulte wrote: I'm having a heck of a time getting hdlc to work on kernel 2.6.11.9 .. I compiled hdlc, hdlc_gen, hdlc_cisco, hdlc_raw, into the kernel (note into, and not 'modules'). Kevin can correct me if I got this wrong, but IIRC, he

Re: [Asterisk-Users] Asterisk 1.0.9: TE411P replacement for TE410P 1stgen causes crashes

On Tue, Aug 23, 2005 at 08:03:42PM +0200, [EMAIL PROTECTED] wrote: I replaced a TE410P Rev C 1st Generation Firmware with a TE411P without any cfg changes (zaptel/zapata). As a result Asterisk crashes on outbound from PRI4 going to PRI1 calls: Aug 23 18:22:00 WARNING[4693]: chan_zap.c:7545

Re: [Asterisk-Users] any ISDN/PRI signaling experts out there?

On Fri, Aug 19, 2005 at 07:01:00AM -0600, Damon Estep wrote: On the same setup, if I connect another PRI device to it that emulates switch side signaling and includes the CNAM as a Display IE in the setup, the SIP invite is properly formatted and * receives the calling party name. Does

Re: [Asterisk-Users] TE411P problem

On Fri, Aug 12, 2005 at 09:03:42PM -0500, Tim Connolly wrote: You might start by running /usr/src/zaptel/zttest. See if you stay at 100%. That's going to be the first thing digium checks. You might also run the autosupport script and take a look at it for anything obvious. I'm having lots of

Re: [Asterisk-Users] TE411P problem

On Fri, Aug 12, 2005 at 09:03:42PM -0500, Tim Connolly wrote: I'm having lots of stability problems with my 411's. I'm not blaming the 411 yet, just seems odd that I ran for months on a TE110P with a peak of 10-15 calls, and now my box kernel panics each time it hits the same load. Granted,

Re: [Asterisk-Users] PRI dropped calls w/ asterisk dropped between pstn norstar

On Wed, Aug 10, 2005 at 08:50:50PM -0400, Gary Reuter wrote: I dropped an asterisk server with a TE405P between a Norstar Meridian PBX and it's PRI PSTN connection. Everything seemed to work fine using a pass-thru-type dialplan configuration... except now we've realised that outbound calls to

Re: [Asterisk-Users] ISDN DID

On Tue, Aug 09, 2005 at 05:20:15PM -0500, Panitaxx wrote: thanks for your response. here is the log of one call: Enabled debugging on span 1 Asterisk*CLI Protocol Discriminator: Q.931 (8) len=33 Call Ref: len= 2 (reference 72/0x48) (Originator) Message type: SETUP (5) [a1]

Re: [Asterisk-Users] PRI problem

On Tue, Jul 12, 2005 at 10:59:42PM +0800, matt001 wrote: currently we are able to use our USA sip phone to conenct into the E1 box, but still unable to dial out to chinese phone numbers. They said from their ISDN switch console, it shows D channel not connected to the voip server yet. here

Re: [Asterisk-Users] ISDN PRI No Audio

On Wed, Jul 06, 2005 at 05:24:06PM -0500, Andy Brezinsky wrote: [Span 3 D-Channel 0] Channel ID (len= 6) [ Ext: 1 IntID: Explicit, PRI Spare: 0, Exclusive Dchan: 0 [Span 3 D-Channel 0]ChanSel: Reserved [Span 3 D-Channel 0] Ext: 1 DS1 Identifier:

Re: [Asterisk-Users] ISDN PRI No Audio

On Wed, Jul 06, 2005 at 12:42:52PM -0500, Andy Brezinsky wrote: Console Output: -- Accepting call from '414944' to '80094042XX' on channel 2/24, span 4 -- Executing Wait(Zap/48-1, 3) in new stack -- Executing Answer(Zap/48-1, ) in new stack -- Executing Playback(Zap/48-1,

Re: [Asterisk-Users] Gizmo: Skype done right?

On Thu, Jun 30, 2005 at 01:31:55PM -0700, Jerry Glomph Black wrote: I've just submitted this as a Slashdot story, too. I have absolutely no connection with any of the principals, I just think they are doing the right thing. This could have a major impact on the Asterisk community, and VoIP

Re: [Asterisk-Users] Setting Caller ID after Dial

On Wed, Jun 29, 2005 at 12:37:34PM -0500, Eric Wieling aka ManxPower wrote: Bryce Chidester wrote: The CallerID that is seen by others on calls originating from your PRI is set by your PRI provider; you have no control from Asterisk about this as it gets overridden by the provider. You

Re: [Asterisk-Users] INBAND DTMF G729 ASTERISK

On Fri, Jun 24, 2005 at 11:59:51AM -0400, Julio Arruda wrote: Andrew Kohlsmith wrote: On Friday 24 June 2005 10:58, [EMAIL PROTECTED] wrote: But there are some products that supports DTMF inband on G729. Ok, it will not work in most cases(like everyone told) but why Asterisk dont support

Re: [Asterisk-Users] OT: MAX TNT and PRI calling name (CNAM) facility message

On Thu, Jun 23, 2005 at 12:20:34AM -0600, Kevin Blackham wrote: Does anyone have a MAX/APX with working ingress PRI calling name? I recently acquired a MAX TNT on the cheap and it's integrating fine except for one thing. In the 11.0.0 release notes, it is stated that ISDN calling name will,

Re: [Asterisk-Users] INBAND DTMF G729 ASTERISK

On Thu, Jun 23, 2005 at 03:19:06PM -0400, [EMAIL PROTECTED] wrote: Why don't Asterisk support inband DTMF with G729? Is there a way to do that!? I think the answer to that is quite obvious; G.729 is a lossy voice oriented compression technique. Any inband DTMF data will be essentially

Re: [Asterisk-Users] New Asterisk Implementation

On Wed, Jun 22, 2005 at 05:32:22PM +0200, harry gaillac wrote: look at ser projects: asterisk is limited to 250 channels What kind of crack are you smoking? There are people that have set up more than 250 channel systems. Matthew Fredrickson ___

Re: [Asterisk-Users] New Asterisk Implementation

the device files directly for channels above the limit of /dev/zap/ device entries. On 6/22/05, Matt Fredrickson [EMAIL PROTECTED] wrote: On Wed, Jun 22, 2005 at 05:32:22PM +0200, harry gaillac wrote: look at ser projects: asterisk is limited to 250 channels What kind of crack are you

Re: [Asterisk-Users] Clicks in audio with TE100P PRI

On Thu, Jun 09, 2005 at 12:51:30AM -0300, Alejandro G wrote: I should tell you that the TE100P is connected to another E1 board (not a live E1) from Natural Microsystems which acts as a gateway to PSTN. This board works as a PRI master but I don't think that this could be the problem as long

Re: [Asterisk-Users] SS7

On Tue, Jun 07, 2005 at 11:30:27AM -0400, Matt wrote: Hi, Has anyone used the SS7 link from Digium? If so, how did it work for you? Any issues? Anything to be aware of? Do I just need a T1 card like the PRI card I have now from Digium? FYI, I don't think that Digium has an SS7 link :-)

Re: [Asterisk-Users] TE410P

On Tue, Jun 07, 2005 at 07:18:32PM +0100, Tony Hoyle wrote: Juan Pablo Abuyeres wrote: lspci -v says: 02:08.0 Communication controller: Unknown device d161:0410 (rev 02) Flags: bus master, medium devsel, latency 32, IRQ 52 Memory at dd20 (32-bit, non-prefetchable)

Re: [Asterisk-Users] Philips - [QSIG] - Alcatel - [H323] - Asterisk -[SIP] - Users.

On Wed, May 04, 2005 at 04:26:01PM +0200, Andreas Sikkema wrote: DT wrote: Firstly we have to connect our Asterisk system to a Philips PBX throught QSIG protocol (interfaces S0), but we doesn't find any documentation about the support of QSIG and S0 interfaces by Asterisk.

Re: [Asterisk-Users] Zap/PRI: received AOC-E charging

On Tue, Apr 26, 2005 at 09:04:48AM -0500, Matthew Boehm wrote: Trying to make a call via our PRI: (CVS everything, CVS-NHEAD-04/23/05-16:08:12) -- Executing Dial(IAX2/[EMAIL PROTECTED], Zap/R2/2815699900|30) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called

Re: [Asterisk-Users] Local Echo

On Tue, Apr 12, 2005 at 05:43:18PM -0700, Noah Silverman wrote: Great suggestion. I'll try it ASAP. Where do I get fxotune? It's in CVS-HEAD zaptel. You'lll need to use the CVS-HEAD zaptel drivers as well, since there is a new IOCTL for doing echo tuning. Matthew Fredrickson

Re: [Asterisk-Users] Local Echo

On Tue, Apr 12, 2005 at 07:03:26PM -0700, Bashir Ullah - www.Lamsre.Com wrote: hi i did not find fxotune under zapte-1.0.6 , please let me know is it different module , need to install seperate, please show me the way , i am having same echo problem and finding its solution for mt tdm fxo.

Re: [Asterisk-Users] TE110P - NT-Mode ?

On Tue, Apr 12, 2005 at 12:15:02PM +0200, Henry Jensen wrote: Hello, I still try to connect a TE110P card to a TMS2 card in a Siemens HiPath 3750. The TMS2 card can be used to connect to an NT (Amtsanschluss) or to connect to another S2M-Line (PRI). When connecting to another PRI, I can

Re: [Asterisk-Users] Local Echo

On Tue, Apr 12, 2005 at 10:16:16AM -0700, Noah Silverman wrote: I have a strange echo problem. When speaking on the phone with someone, I hear MY OWN voice with a sever echo. The other party sounds perfect, and they can hear me perfectly. It is as if only the sidetone has an echo. I'm

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