Re: [asterisk-users] Fwd: Your message to asterisk-users awaits moderator approval

On Mon, Jan 25, 2021 at 3:04 PM Jeff LaCoursiere wrote: > > A 40Kb limit seems a bit draconian these days. I simply attached a small > pic to illustrate a point. May I vote to up the limit? 100K? > I've increased it and we'll see how it goes. -- Joshua C. Colp Asterisk Technical Lead

[asterisk-users] Fwd: Your message to asterisk-users awaits moderator approval

A 40Kb limit seems a bit draconian these days.  I simply attached a small pic to illustrate a point.  May I vote to up the limit?  100K? Cheers, Jeff LaCoursiere StratusTalk, Inc. Forwarded Message Delivered-To: j...@stratustalk.com Received: by

[asterisk-users] Fwd: [CFP] FOSDEM 2021, RTC devroom, speakers, volunteers neeeded

FOSDEM - Real Time Communications devroom CfP = NOTE: we have extended the deadline but we will give preference to people who submitted earlier when we set the time for each talk. Please submit your proposal ASAP so that you can get your preferred time

[asterisk-users] Fwd: Asterisk 16.14.0 pjsip transport-tls cert parsing error

Hi, I'm experimenting with Asterisk-16.14.0 on a CentOS7 box, and run into problems loading the SSL certificate to establish transport-tls. Tried self-signed certificate generated with ast_tls_cert under contrib/scripts and the one issued by Letsencrypt, both would bomb out with a parsing

Re: [asterisk-users] Fwd: Legacy TDM400

AFAIK it requires DAHDI version 2 For unknown reasons, this and many other card drivers were removed in DAHDI version 3 Suggest you compile from source, rather than any repository, selecting the last DAHDI version 2 and at least Asterisk 13, though it is EOL or nearly, it still is a good

[asterisk-users] Fwd: Legacy TDM400

Hello all, It's been quite some number of years since I played around with Asterisk and I'm just now getting back into it. I think the last version I worked with was 1.8. I have a legacy Digium TDM400 PCI card and am wondering if that will still work on newer versions of Asterisk. My initial

Re: [asterisk-users] Fwd: call-id on cdr's

On Wed, Oct 14, 2020 at 5:09 AM Rui Mota wrote: > Hi. > I'd like to get the call-id from a call (the *C-061c* part) to set in > on a field on cdr's, like > > [Oct 8 16:09:37] VERBOSE[25701][*C-061c*] app_queue.c: > SIP/12234-6b18 is busy > [Oct 8 16:09:37]

[asterisk-users] Fwd: call-id on cdr's

Hi. I'd like to get the call-id from a call (the *C-061c* part) to set in on a field on cdr's, like [Oct 8 16:09:37] VERBOSE[25701][*C-061c*] app_queue.c: SIP/12234-6b18 is busy [Oct 8 16:09:37] VERBOSE[25701][*C-061c*] app_queue.c: Nobody picked up in 0 ms [Oct 8 16:09:38]

Re: [asterisk-users] Fwd: blf problems after dialplan reload

On Wed, Jul 22, 2020 at 6:24 PM Israel Gottlieb wrote: > > > > Hi Guys > we have a system that uses a lot of custom hints based on the extension > the extensions use the format of ext-system for example 200-pbx01 > when starting asterisk the "core show hints" show the correct hints and > blf

[asterisk-users] Fwd: blf problems after dialplan reload

Hi Guys we have a system that uses a lot of custom hints based on the extension the extensions use the format of ext-system for example 200-pbx01 when starting asterisk the "core show hints" show the correct hints and blf works as expected in the extensions.conf we have _.,hint,Custom:${exten}

[asterisk-users] Fwd: [CFP] FOSDEM 2020, RTC devroom, speakers, volunteers neeeded

FOSDEM - Real Time Communications devroom CfP = Overview [FOSDEM](https://fosdem.org) is one of the world's premier meetings of free software developers, with over five thousand people attending each year. FOSDEM 2020 takes place 1-2 February

[asterisk-users] Fwd: [CFP] FOSDEM 2020, RTC devroom, speakers, volunteers neeeded

FOSDEM - Real Time Communications devroom CfP = Overview [FOSDEM](https://fosdem.org) is one of the world's premier meetings of free software developers, with over five thousand people attending each year. FOSDEM 2020 takes place 1-2 February

Re: [asterisk-users] Fwd: Ihnen steht ein Kupon bei PIOSPARTSLAP IT Remarketing e.K zur Verfügung

On 2/5/19 3:55 PM, hw wrote: Derzeit interessante Angebote habe ich dort nicht entdeckt. Sorry, this wasn't supposed to go to the list! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out

[asterisk-users] Fwd: Ihnen steht ein Kupon bei PIOSPARTSLAP IT Remarketing e.K zur Verfügung

Derzeit interessante Angebote habe ich dort nicht entdeckt. Forwarded Message Return-Path: Received: from mail.g4yg.de (localhost [127.0.0.1]) by sunflo-mx (Cyrus v2.4.17) with LMTPA; Mon, 04 Feb 2019 13:06:54 +0100 X-Sieve:CMU Sieve 2.4 Envelope-to:

Re: [asterisk-users] Fwd: Backport fix

Saint Michael wrote: There is a big bug https://issues.asterisk.org/jira/browse/ASTERISK-24768 It affects version 11, but it was fixed only from 13.20 onwards. However, millions of people still use version 11. This bug makes Asterisk

[asterisk-users] Fwd: Backport fix

There is a big bug https://issues.asterisk.org/jira/browse/ASTERISK-24768 It affects version 11, but it was fixed only from 13.20 onwards. However, millions of people still use version 11. This bug makes Asterisk crash every few hours under any load that has RTP going through Asterisk. For

Re: [asterisk-users] Fwd: PJSIP subscribe

I have seen the problem is with: Jitsi, XLite, Bria 3.X Grandstream Wave work correctly. I have to test with hardphone. Regards El 09/06/2016 a las 01:11, George Joseph escribió: On Wed, Jun 8, 2016 at 8:48 AM, Annus Fictus > wrote:

Re: [asterisk-users] Fwd: PJSIP subscribe

On Wed, Jun 8, 2016 at 8:48 AM, Annus Fictus wrote: > Hello, > > How can I know if is a BUG and report on Asterisk-Jira? > Asterisk is sending the correct "ep:busy" status but the clients appear to not be using it. I'm not sure what else we can do on the Asterisk side.

[asterisk-users] Fwd: PJSIP subscribe

Hello, How can I know if is a BUG and report on Asterisk-Jira? Thank you Regards Mensaje reenviado Asunto: PJSIP subscribe Fecha: Mon, 6 Jun 2016 19:13:35 +0200 De: Annus Fictus Para: Asterisk Users Mailing List - Non-Commercial

Re: [asterisk-users] Fwd: Unable to place outbound calls

Could potentially be a NAT problem. Could you do a sip debug and see whether there is any traffic to / from the address? Sent from my iPhone > On 15 Mar 2016, at 9:52 PM, Administrator TOOTAI wrote: > > Le 15/03/2016 11:20, Feroz Ahmed a écrit : >> Hi I need help > > Hello

Re: [asterisk-users] Fwd: Unable to place outbound calls

Le 15/03/2016 11:20, Feroz Ahmed a écrit : Hi I need help Hello Ahmed This is the error: [...] [Mar 14 19:55:15] WARNING[20595][C-000b]: app_dial.c:2437 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent) == Everyone is busy/congested at this

[asterisk-users] Fwd: Unable to place outbound calls

Hi I need help This is the error: Really destroying SIP dialog 'NDMxOWRmYTRhMWVkMGFhMjllMzU4YmNmNjQwN2NlM2Y.' Method: SUBSCRIBE -- Executing [00919885497796@internal:1] Set("SIP/1001-000b", "CALLERID(num)=8790771141") in new stack -- Executing [00919885497796@internal:2]

Re: [asterisk-users] Fwd: Sublime Text License Key

Yeah... I emailed them and let them know I screwed up, but no reply. I humbly request you don't use my key. Sublimetext is pretty great, and you can download and use it for free, fully functional, without pretending to be me. The nag screen is even polite. I'm so embarrassed that I did this... On

Re: [asterisk-users] Fwd: Sublime Text License Key

Wonder what happens when an entire mailing list tries to use that key?... On 05/10/15 15:28, Optical Phoenix wrote: -- Forwarded message -- From: *Sublime HQ Pty Ltd* > Date: Wednesday, July 25, 2012 Subject: Sublime Text

[asterisk-users] Fwd: Sublime Text License Key

-- Forwarded message -- From: *Sublime HQ Pty Ltd* Date: Wednesday, July 25, 2012 Subject: Sublime Text License Key To: "opticalphoe...@gmail.com" Hello, Thanks for purchasing a copy of Sublime Text! Your license key is: -

Re: [asterisk-users] Fwd: Sublime Text License Key

The company making sublime text gets few thousands of dollars of notional loss :) On Oct 5, 2015 8:45 PM, "Steve Howes" wrote: > Wonder what happens when an entire mailing list tries to use that key?... > > On 05/10/15 15:28, Optical Phoenix wrote: > > --

Re: [asterisk-users] Fwd: Sublime Text License Key

On 05/10/15 16:18, Mitul Limbani wrote: The company making sublime text gets few thousands of dollars of notional loss :) I was thinking more about if they'd built in software activation type stuff. But yea, stealing bad etc too. Steve --

Re: [asterisk-users] Fwd: ferie estive

On Mon, Aug 24, 2015 at 9:40 PM, Pete Mundy p...@fiberphone.co.nz wrote: Any chance the list admins could unsubscribe Mr Anzaldi until he gets his broken auto-responder fixed? He has been unsubscribed and alerted to the issue. Thanks! -- Rusty Newton Digium, Inc. | Community Support Manager

[asterisk-users] Fwd: ferie estive

Any chance the list admins could unsubscribe Mr Anzaldi until he gets his broken auto-responder fixed? Begin forwarded message: From: davide.anza...@netecom.it Subject: ferie estive Date: 25 August 2015 2:33:00 PM NZST To: Pete Mundy p...@fiberphone.co.nz Sono assente per ferie e

Re: [asterisk-users] Fwd: What database should I use, for simple data storing? SQLite or the buitin one?

To some extent the answer depends on how you want to use it overall, and what you already have installed. We did something similar on a project where we created a simple app accessible via AGI, and it stored/retrieved data to/from anXML file. If your access frequency is low enough that

[asterisk-users] Fwd: OpenVZ with asterisk 13

Dear all, Is anyone has experience making Asterisk server with virtual server OPEN-VZ (in proxmox 3.4 box) ? My boss want to build a production server with it, and it will have +/- 300 sip user (concurrent call maybe 150 call) Is it good to go, or not ? I really hope someone who have

Re: [asterisk-users] Fwd: Asterisk pjsip auto dtmf mode

On Sat, Jan 17, 2015 at 12:04 AM, Yaron Nachum nachum.ya...@gmail.com wrote: Hello Asterisk Users, I have been looking for similar auto dtmf mode implementation on pjsip, but didn't see it coming, so I decided to give it a try. My basic plan was to do it as simple as possible with minimum

[asterisk-users] Fwd: Asterisk pjsip auto dtmf mode

Hello Asterisk Users, I have been looking for similar auto dtmf mode implementation on pjsip, but didn't see it coming, so I decided to give it a try. My basic plan was to do it as simple as possible with minimum changes because I am not familiar with all Asterisk code. My idea is to use rfc4733

[asterisk-users] Fwd: ReceiveFax for multiple page (asterisk 13.0.1)

Hi all, I have problem for receiving fax from multiple page fax that sent from fax machine (analog). The error is : WARNING T.30 Page did not end cleanly This is my dialplan [inboundfax] exten = s,1,NoOp( FAX RECEIVED from ${CALLERID(num)} ${STRFTIME(${EPOCH},,%c)} ) exten =

[asterisk-users] Fwd: no ipv6 dns resolution for outbound registration with pjsip/asterisk13.1

3rd attempt to post it to the list, please ignore if it is duplicate I have the following problem When trying to setup asterisk 13.1 with PJSIP to connect to my IPV6 capable SIP provider the registration fails. [code][Dec 22 19:24:24] DEBUG[25247] pjsip: tsx0x110736c .Transaction created

[asterisk-users] Fwd: Fwd: Fwd: Fwd: Fwd: Fwd: Fwd: Fwd: 26.

Anna Crepes: Traubenzucker + Feldsalat spezielles Dressing (bringt selbst mit?) Weitergeleitete Nachricht Betreff: Fwd: Fwd: Fwd: Fwd: Fwd: Fwd: Fwd: 26. Datum: Thu, 11 Dec 2014 15:34:39 +0100 Von: Markus unive...@truemetal.org An: unive...@truemetal.org Geschenke Moritz:

Re: [asterisk-users] Fwd: Fwd: Fwd: Fwd: Fwd: Fwd: Fwd: Fwd: 26.

OMG.. how embarassing.. that was my personal reminder E-Mail for x-mas dinner. Not meant for this list. Please ignore. Shame on me.. *blushing* LOL. Am 12.12.2014 um 21:19 schrieb Markus: Anna Crepes: Traubenzucker + Feldsalat spezielles Dressing (bringt selbst mit?)

Re: [asterisk-users] Fwd: Fwd: Fwd: Fwd: Fwd: Fwd: Fwd: Fwd: 26.

OMG.. how embarassing.. that was my personal reminder E-Mail for x-mas dinner. Not meant for this list. Please ignore. Shame on me.. *blushing* LOL. Am 12.12.2014 um 21:19 schrieb Markus: Anna Crepes: Traubenzucker + Feldsalat spezielles Dressing (bringt selbst mit?)

[asterisk-users] Fwd: Identifying the multiple cards Digium TE820

Hello all! We have 4 pieces TE820 cards in one server. How we can define via settings on server, which card is number 1, which is number 2, which is number 3 and which is number 4? Because if later we add for example 2 cards more in the server, the cards can change their position. for example card

[asterisk-users] Fwd: Regarding packet2packet bridging

Dear concern, I want to configure packet2packet bridging in asterisk. How could I do this any of the tutorial or instructions will help ? I found the setting the canreinvite=yes will do the stuff but it is not working I am using asterisk 12.3 version I am very new to asterisk please help me

[asterisk-users] Fwd: Asterisk 12.1.1 on Ubuntu 12.04 crashing

Hi, I'm writing b/c I'm having a crash that happens a lot... it happened without better_backtraces and then I went ahead and recompiled with it. Then it stopped until today. You can see the crash data at http://pastebin.com/1vbrWepr (bt, bt full and thread apply all bt) The extensions.conf is

[asterisk-users] Fwd: early media (video)

ok then... sorry it took so long, but I already posted it yesterday but it was rejected by the list's moderator as it exceeded 40k in size :( so i'll resend it now: part #1 is this, part #2 will only contain the attachment with working direct-Video (not early media) for comparison... --- 1) i

[asterisk-users] Fwd: early media (video)

part #2 --- SIP read from UDP:10.10.1.144:5060 --- INVITE sip:306@10.10.1.201 SIP/2.0 Via: SIP/2.0/UDP 10.10.1.144:5060;branch=z9hG4bK.UwYy83FQ9;rport From: sip:301@10.10.1.201;tag=XY9jSoWko To: sip:306@10.10.1.201 CSeq: 20 INVITE Call-ID: RYw7fDsLDF Max-Forwards: 70 Supported: replaces, outbound

[asterisk-users] Fwd: issue with speech in IVR

hello i add the following in chan_dahdi and the issue has been solved thanks a lot for your help and support now ican stop the speech and go to my context i really appreciate your help and support immediate = yes echocancel = no dtmfmode = auto -- Forwarded message -- From:

[asterisk-users] Fwd: Goto(${PREV_CONTEXT},s,1) functionality?

I have a moderately complicated ivr menu, implemented in an AEL dialplan. A feature I'm struggling to add now is going back from current context to a previous one, so that user could browse almost the whole ivr menu manually. The difficulty is that when current context has several entry points

Re: [asterisk-users] Fwd: Re: Asterisk T.38 Pass-Through doesn't work

On 01/08/2013, at 2:20 PM, Zoltán Fekete bl...@gyoz.info wrote: 2013/8/1 Joshua Colp jc...@digium.com Larry Moore wrote: On 31/07/2013 8:08 PM, Joshua Colp wrote: Zoltán Fekete wrote: Thank You Larry! I have discussed with my provider. They are not able to insert the T38MaxBitRate

Re: [asterisk-users] Fwd: Re: Asterisk T.38 Pass-Through doesn't work

2013/8/1 Joshua Colp jc...@digium.com Larry Moore wrote: On 31/07/2013 8:08 PM, Joshua Colp wrote: Zoltán Fekete wrote: Thank You Larry! I have discussed with my provider. They are not able to insert the T38MaxBitRate value into the sip answer. :(

Re: [asterisk-users] Fwd: Re: Asterisk T.38 Pass-Through doesn't work

Thank You Larry! I have discussed with my provider. They are not able to insert the T38MaxBitRate value into the sip answer. :( https://gist.github.com/anonymous/6120148 (line 559) That means we are not able to passtrough T38 Faxes with any asterisk version at all? What do you mean? Am I able to

Re: [asterisk-users] Fwd: Re: Asterisk T.38 Pass-Through doesn't work

Zoltán Fekete wrote: Thank You Larry! I have discussed with my provider. They are not able to insert the T38MaxBitRate value into the sip answer. :( https://gist.github.com/anonymous/6120148 (line 559) That means we are not able to passtrough T38 Faxes with any asterisk version at all? What do

Re: [asterisk-users] Fwd: Re: Asterisk T.38 Pass-Through doesn't work

On 31/07/2013 8:08 PM, Joshua Colp wrote: Zoltán Fekete wrote: Thank You Larry! I have discussed with my provider. They are not able to insert the T38MaxBitRate value into the sip answer. :( https://gist.github.com/anonymous/6120148 (line 559) That means we are not able to passtrough T38

Re: [asterisk-users] Fwd: Re: Asterisk T.38 Pass-Through doesn't work

Larry Moore wrote: On 31/07/2013 8:08 PM, Joshua Colp wrote: Zoltán Fekete wrote: Thank You Larry! I have discussed with my provider. They are not able to insert the T38MaxBitRate value into the sip answer. :( https://gist.github.com/anonymous/6120148 (line 559) That means we are not able to

Re: [asterisk-users] Fwd: Re: Asterisk T.38 Pass-Through doesn't work

On 22/07/2013 5:40 AM, Zoltán Fekete wrote: Hi! I have exactly the same problem on asterisk 1.8.22.0 and also on separate 11.2.1 when sending fax to PSTN. Tryed with spa-3102, spa-2102, Patton Smartnode 4634, and Zoiper softphone. SpanDsp also works without any problem on my box. As I

Re: [asterisk-users] Fwd: Re: Asterisk T.38 Pass-Through doesn't work

if that parameter is missing, then the code would in fact default to 2400 as a safe value. Kevin Larsen - Systems Analyst From: Zoltán Fekete bl...@gyoz.info To: asterisk-users@lists.digium.com, Date: 07/21/2013 04:40 PM Subject:[asterisk-users] Fwd: Re: Asterisk T.38 Pass-Through

Re: [asterisk-users] Fwd: Re: Asterisk T.38 Pass-Through doesn't work

On 23/07/2013 6:18 AM, Kevin Larsen wrote: The a=T38MaxBitRate issue you refer to was one that was actually discovered at my company and submitted by a colleague. It was fixed in 11.3.0 and 1.8.21.0. However, I think that it wouldn't help based on the description below being that the parameter

Re: [asterisk-users] Fwd: Re: Asterisk T.38 Pass-Through doesn't work

On 22/07/2013 10:19 PM, Larry Moore wrote: On 22/07/2013 5:40 AM, Zoltán Fekete wrote: Hi! I have exactly the same problem on asterisk 1.8.22.0 and also on separate 11.2.1 when sending fax to PSTN. Tryed with spa-3102, spa-2102, Patton Smartnode 4634, and Zoiper softphone. SpanDsp also works

[asterisk-users] Fwd: Re: Asterisk T.38 Pass-Through doesn't work

Hi! I have exactly the same problem on asterisk 1.8.22.0 and also on separate 11.2.1 when sending fax to PSTN. Tryed with spa-3102, spa-2102, Patton Smartnode 4634, and Zoiper softphone. SpanDsp also works without any problem on my box. As I remember it was a bug in 1.8.1.x that the

[asterisk-users] Fwd: AQuA Meter – waveform analysis to get continous MOS scores for your network

Hi, Although this is a repost from Asterisk biz, we would like to ask if somebody may help us to develop a native Asterisk module using AQuA technology for voice quality monitoring using the same web service AQuA Meter is using. Thanks, Sevana Finland/Estonia -- Forwarded message

[asterisk-users] Fwd: Google Calendar issue

From 23 juanary 2013 on calendar.conf have to change type=caldav to type=ical http://forums.asterisk.org/viewtopic.php?f=1t=85623 http:// Regards Mensaje original Asunto: Google Calendar issue Fecha: Sat, 23 Feb 2013 10:22:17 -0500 De: Bakko asannu...@gmail.com

[asterisk-users] Fwd:

http://95.110.242.215/6bmhxm.php -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello

Re: [asterisk-users] Fwd: Errors Compiling Libpri-1.4.13

2012/11/27 Richard Mudgett rmudg...@digium.com On Mon, Nov 19, 2012 at 08:47:23PM +0100, Adolphus Enaboifo wrote: .. I get errors while trying to compile Libpri 1.4.13. (check attachment} Can you guys please help me prescribe a fix. [snip] gcc -g -Wall -Werror

Re: [asterisk-users] Fwd: Errors Compiling Libpri-1.4.13

2012/11/19 Shaun Ruffell sruff...@digium.com On Mon, Nov 19, 2012 at 08:47:23PM +0100, Adolphus Enaboifo wrote: .. I get errors while trying to compile Libpri 1.4.13. (check attachment} Can you guys please help me prescribe a fix. [snip] gcc -g -Wall -Werror -Wstrict-prototypes

Re: [asterisk-users] Fwd: Errors Compiling Libpri-1.4.13

On Mon, Nov 19, 2012 at 08:47:23PM +0100, Adolphus Enaboifo wrote: .. I get errors while trying to compile Libpri 1.4.13. (check attachment} Can you guys please help me prescribe a fix. [snip] gcc -g -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -fPIC -O2 -MD -MT pridump.o

[asterisk-users] Fwd: Errors Compiling Libpri-1.4.13

Good Day dear members, We are trying to test asterisk in our office to extend the reach of our present proprietary pabx system if successful. I am using an oracle virualbox 4.2.4 as the virtual server platform with ubuntu 12.04.1 server as the operating system. I get errors while trying to

Re: [asterisk-users] Fwd: Errors Compiling Libpri-1.4.13

On Mon, Nov 19, 2012 at 08:47:23PM +0100, Adolphus Enaboifo wrote: .. I get errors while trying to compile Libpri 1.4.13. (check attachment} Can you guys please help me prescribe a fix. [snip] gcc -g -Wall -Werror -Wstrict-prototypes -Wmissing-prototypes -fPIC -O2 -MD -MT pridump.o -MF

[asterisk-users] Fwd: asterisk-users Digest, Vol 95, Issue 33

Thanks I had this line in my /etc/asterisk/chan_dahdi.conf : include=/etc/asterisk/dahdi-channels.conf the file /etc/asterisk/dahdi-channels.conf was generated by /usr/sbin/dahdi_genconf I simply did that : cat /etc/asterisk/dahdi-channels.conf /etc/asterisk/chan_dahdi.conf It works now.

Re: [asterisk-users] Fwd: RTP stats explaination

Hi Dave, On Fri, May 18, 2012 at 11:27 PM, Dave Platt dpl...@radagast.org wrote: In our app we do not forward packet immediately. After enough packet received to increase rtp packetization time (ptime) the we forward the message over raw socket and set dscp to be 10 so that this time packets

Re: [asterisk-users] Fwd: RTP stats explaination

In our app we do not forward packet immediately. After enough packet received to increase rtp packetization time (ptime) the we forward the message over raw socket and set dscp to be 10 so that this time packets can escape iptable rules. From client side the RTP stream analysis shows nearly

Re: [asterisk-users] Fwd: RTP stats explaination

On Fri, 18 May 2012, Dave Platt wrote: A maximum jitter of 230 milliseconds looks pretty horrendous to me. This is going to cause really serious audio stuttering on the receiving side, and/or will force the use of such a long jitter buffer by the receiver that the audio will suffer from an

Re: [asterisk-users] Fwd: RTP stats explaination

On 05/18/2012 12:51 PM, Steve Edwards wrote: On Fri, 18 May 2012, Dave Platt wrote: A maximum jitter of 230 milliseconds looks pretty horrendous to me. This is going to cause really serious audio stuttering on the receiving side, and/or will force the use of such a long jitter buffer by the

[asterisk-users] Fwd: Flashphoner

Thought this deserved a name and shame! ;) Steve Begin forwarded message: From: Pavel Ismailov pavel.ismai...@gmail.com Date: 27 April 2012 06:58:07 GMT+01:00 To: steve-li...@geekinter.net Subject: Flashphoner Hello! My name is Pavel Ismailov and I`m CEO of www.flashphoner.com

Re: [asterisk-users] Fwd: Flashphoner

Subject: [asterisk-users] Fwd: Flashphoner Thought this deserved a name and shame! ;) Steve Begin forwarded message: From: Pavel Ismailov pavel.ismai...@gmail.com Date: 27 April 2012 06:58:07 GMT+01:00 To: steve-li...@geekinter.net Subject: Flashphoner Hello! My name is Pavel

[asterisk-users] Fwd: Realtime asterisk 10.3.0

can anyone help me with this case?   thank you, Leo   Hi there, I setup realtime asterisk 10.3.0 with backend mysql server. everything seems to work fine except when I tried to enable the extensions for dialplan to be obtained from mysql, I got an empty dialplan. I am not sure why this

[asterisk-users] Fwd: Re: Authentication: username and password, also to be from the LAN

On 3/26/2012 1:11 PM, bilal ghayyad wrote: If it possible, then is it possible to be a configuration per user? Just expanding on Jim's answer- to allow user example with password secret from 192.168.0.*, do something like: in /etc/asterisk/sip.conf: [example] type=friend secret=secret

[asterisk-users] Fwd: Do you know how Asterisk came to be?

Apologies for the top post, something is screwed up with my email client, will fix it soon. What a BS story that I have debunked many times. A used Key System could be purchased for a few hundred dollars, a much better investment then writing your own PBX from scratch. A company that is

[asterisk-users] Fwd: Re: Asterisk CLI unresponsive

Hello, is there anyone that can give me some more information on these deadlocks ?! How can these deadlocks occur and what is good practise to avoid these problems ?? Jonas. Original Message Subject:Re: [asterisk-users] Asterisk CLI unresponsive Date: Fri,

Re: [asterisk-users] Fwd: Re: Asterisk CLI unresponsive

On 6 February 2012 10:45, Jonas Kellens jonas.kell...@telenet.be wrote: ** Hello, is there anyone that can give me some more information on these deadlocks ?! How can these deadlocks occur and what is good practise to avoid these problems ?? Jonas. The only way to avoid deadlocks is

Re: [asterisk-users] Fwd: Re: Asterisk CLI unresponsive

On 02/06/2012 12:14 PM, Steve Davies wrote: On 6 February 2012 10:45, Jonas Kellens jonas.kell...@telenet.be mailto:jonas.kell...@telenet.be wrote: Hello, is there anyone that can give me some more information on these deadlocks ?! How can these deadlocks occur and what is

Re: [asterisk-users] Fwd: Re: Asterisk CLI unresponsive

asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Fwd: Re: Asterisk CLI unresponsive -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory

Re: [asterisk-users] Fwd: Re: Asterisk CLI unresponsive

On 02/06/2012 12:25 PM, isr...@gmail.com wrote: Your running into a bug and the only way to solve it is to report it and debug it and hope for a fix There is no way someone can help without it being debugged and knowing what's causing it to lockup The only key to unlcock it when it gets

Re: [asterisk-users] Fwd: Re: Asterisk CLI unresponsive

On 12-02-06 09:15 AM, Jonas Kellens wrote: On 02/06/2012 12:25 PM, isr...@gmail.com wrote: Your running into a bug and the only way to solve it is to report it and debug it and hope for a fix There is no way someone can help without it being debugged and knowing what's causing it to lockup The

Re: [asterisk-users] Fwd: Re: Asterisk CLI unresponsive

On 02/06/2012 03:19 PM, Paul Belanger wrote: On 12-02-06 09:15 AM, Jonas Kellens wrote: On 02/06/2012 12:25 PM, isr...@gmail.com wrote: Your running into a bug and the only way to solve it is to report it and debug it and hope for a fix There is no way someone can help without it being

Re: [asterisk-users] Fwd: Re: Asterisk CLI unresponsive

, 2012 9:18 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Fwd: Re: Asterisk CLI unresponsive On 02/06/2012 03:19 PM, Paul Belanger wrote: On 12-02-06 09:15 AM, Jonas Kellens wrote: On 02/06/2012 12:25 PM, isr

Re: [asterisk-users] Fwd: Re: Asterisk CLI unresponsive

On 12-02-06 09:23 AM, Jonas Kellens wrote: On 02/06/2012 03:19 PM, Paul Belanger wrote: On 12-02-06 09:15 AM, Jonas Kellens wrote: On 02/06/2012 12:25 PM, isr...@gmail.com wrote: Your running into a bug and the only way to solve it is to report it and debug it and hope for a fix There is no

Re: [asterisk-users] Fwd: Re: Asterisk CLI unresponsive

- Original Message - From: Jonas Kellens jonas.kell...@telenet.be To: isr...@gmail.com, Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, February 6, 2012 8:15:50 AM Subject: Re: [asterisk-users] Fwd: Re: Asterisk CLI unresponsive

[asterisk-users] Fwd: Desktops Recieved from Delhi

Ignore this mail sorry all Best Regards, Mahesh Katta *BUZZ**WORKS* Business Services Private Limited BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI 201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri (E) Mumbai 400069 GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax

[asterisk-users] Fwd: Re: Welcome to the asterisk-users mailing list

Hi, there is possible put asterisk to passthrough mode for zrtp ? I tried to manipulate with reinvite/canreinvite without success. Petr -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to

[asterisk-users] Fwd: Re: Securing Asterisk

Original Message Subject: Re: [asterisk-users] Securing Asterisk Date: Wed, 27 Jul 2011 09:28:54 -0700 From: Myles Wakeham my...@techsol.org To: p...@provu.co.uk On 07/27/2011 09:23 AM, asterisk-users-requ...@lists.digium.com wrote: On 23/07/11 18:38, CDR wrote: I beg to

[asterisk-users] [Fwd: Re: Strange network issue]

Forwarded Message From: Ishfaq Malik i...@pack-net.co.uk To: Mike Diehl mdi...@diehlnet.com Subject: Re: [asterisk-users] Strange network issue Date: Fri, 22 Jul 2011 09:55:53 +0100 On Fri, 2011-07-22 at 02:53 -0600, Mike Diehl wrote: On Friday 22 July 2011 2:42:12 am Ishfaq

[asterisk-users] Fwd: FW: loadstar and ok one

Hi All, I have experiancing strenge issue with my production Asterisk system. I'm using asterisk vertion 1.4.28 installed cent OS 5. Issue decription. I have SIP trunk from local carrier to their hosted PBX( broadsoft). Out going calls over this trunk working fine and I can make a

Re: [asterisk-users] Fwd: Re: ControlPlayback's options

On 2011-06-10 07:30, virendra bhati wrote: Hi John, Sorry for wrong information. Actually it's J not P option in ControlPlayBack... http://www.voip-info.org/wiki/view/Asterisk+cmd+ControlPlayback That page is correct for asterisk 1.4 but the feature you need is in 1.6.0 and forward. Have

[asterisk-users] Fwd: Re: ControlPlayback's options

Humm... Seems like my message didn't make it. Here we go again.. /Johan Original Message Subject:Re: [asterisk-users] ControlPlayback's options Date: Sun, 05 Jun 2011 22:19:18 +0200 From: Johan Wilfer li...@jttech.se To: Asterisk Users Mailing List - Non-Commercial

Re: [asterisk-users] Fwd: Re: ControlPlayback's options

Hi John, Sorry for wrong information. Actually it's J not P option in ControlPlayBack... http://www.voip-info.org/wiki/view/Asterisk+cmd+ControlPlayback On Fri, Jun 10, 2011 at 12:23 AM, Johan Wilfer li...@jttech.se wrote: Humm... Seems like my message didn't make it. Here we go again..

Re: [asterisk-users] [Fwd: FW: realtime mysql - p4]

...@lists.digium.com] On Behalf Of Hans Witvliet Sent: 22 May 2011 22:50 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] [Fwd: FW: realtime mysql - p4] On Fri, 2011-05-20 at 10:05 +0100, Andrew Thomas wrote: Post your cdr_mysql.conf and res_mysql.conf and we'll take

Re: [asterisk-users] [Fwd: FW: realtime mysql - p4]

On Fri, 2011-05-20 at 10:05 +0100, Andrew Thomas wrote: Post your cdr_mysql.conf and res_mysql.conf and we'll take it from there. Don't forget to remove any 'private' info first (like passwords). Cheers Tnx for the offer, Wil get the files when got back at the office. I presume that

Re: [asterisk-users] [Fwd: FW: realtime mysql - p4]

Witvliet Sent: 19 May 2011 23:14 To: asterisk-users@lists.digium.com Cc: j.witvl...@mindef.nl Subject: [asterisk-users] [Fwd: FW: realtime mysql - p4] Ok, i tried the suggestion: Instead of: sippuser = resource, database_name, table_name sippeer = resource, database_name, table_name I put in: sippuser

[asterisk-users] [Fwd: FW: realtime mysql - p4]

Ok, i tried the suggestion: Instead of: sippuser = resource, database_name, table_name sippeer = resource, database_name, table_name I put in: sippuser = resource, context, table_name sippeer = resource, context, table_name Unfortunately, with the same results. btw i tried both general as

[asterisk-users] Fwd: write system command output into a variable

I found a solution that works fine for me Set(var1=${SHELL(shellcommand)}) Bye Daniel Von: Daniel Knoll dan...@danielknoll.de Datum: 16. April 2011 13:13:28 MESZ An: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Betreff: write system command

Re: [asterisk-users] Fwd: Re: Asterisk as a Condo door opener/intercom

-Scanned-By: MIMEDefang 2.67 on 205.211.164.50 Subject: [asterisk-users] Fwd: Re: Asterisk as a Condo door opener/intercom X-BeenThere: asterisk-users@lists.digium.com X-Mailman-Version: 2.1.11 Precedence: list Reply-To: dbc_aster...@advan.ca, Asterisk Users Mailing List - Non-Commercial

[asterisk-users] Fwd: Re: Asterisk as a Condo door opener/intercom

Asterisk as a phone system makes perfect sense in a condo. You can get all the DID's you want and eliminate costs for the owners. You can offer standard FXO for people who don't care and IP sets for people who want to upgrade to feature sets. Your door openner is a piece of cake. 1. Create an

[asterisk-users] Fwd: changing port 5060 to 5061

-- Forwarded message -- From: darin iv adari...@gmail.com Date: Mon, 11 Apr 2011 14:33:24 +0530 Subject: changing port 5060 to 5061 To: asterisk-users@lists.digium.com Dear Experts, please send me the ways to change asterisk port from 5060 to 5061 i need to configure it because

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