Re: [asterisk-users] Difference between trunk and released versions

Yehavi, The release branches (1.2, 1.4) were at one time trunk. When it was decided to release 1.4, for example, it was branched off from trunk as the 1.4branch. New functionality continued to be added to trunk after that. Once the release branches are created, they are feature-frozen and only

Re: [asterisk-users] How to loging Agent in asterisk 1.4.13 ?

Search for Agent at http://www.voip-info.org/ On 10/12/07, Walter Willis [EMAIL PROTECTED] wrote: how to loging agent asterisk 1.4.13? thanks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list

Re: [asterisk-users] HOw to call queue ???

Search for Queue at http://www.voip-info.org/ On 10/12/07, Walter Willis [EMAIL PROTECTED] wrote: HOw to call queues in asterisk ? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE

Re: [asterisk-users] DS3 Interface

Not at all! You apparently don't realize (sic) you're talking to is just a subtle way of saying look me up! Matt was just nice enough to do the leg work :-) On 10/12/07, David Boyd [EMAIL PROTECTED] wrote: What a waste of time... dave On Fri, 2007-10-12 at 09:35 -0600, Stephen Bosch

Re: [asterisk-users] Combining Flags in Dial()

You mean like: Dial(Zap/1,10,dtf) ? On 10/12/07, Jeng Yu [EMAIL PROTECTED] wrote: Hi All, I have a quick one for you. Is there a way to mask (i.e. combine) the flags in the Dial() application? In other words, a way to do something like Dial(Zap/1,10,d|t|f) to get the effects of the

Re: [asterisk-users] AstManProxy Host Prefix?

I can do it for $10,000 On 10/24/07, asterisk [EMAIL PROTECTED] wrote: What would be nice if it you could specify the host per user in astmanproy.users Anyone interested in making the change? $$$ Doug -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf

Re: [asterisk-users] Stuck Voicemails?

We have that problem here with Asterisk 1.2.9.1. There is a fix in later versions of the 1.2 branch, but I couldn't tell you which one. You can just delete the .txt file from the user's voicemail folder and it should clear the MWI on the phone. On 10/29/07, Matt [EMAIL PROTECTED] wrote: This

Re: [asterisk-users] Chanspy on Asterisk 1.4.19

with SPYGROUP set to 2000 so not much use to us. You can pass multiple options to a dialplan application, so instead of downgrading ChanSpy, you could have just done: exten = 596,n,ChanSpy(|bg(2000)) Or am I missing something? -- Sean Bright [EMAIL PROTECTED

Re: [asterisk-users] Chanspy on Asterisk 1.4.19

. -- Sean Bright [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Chanspy on Asterisk 1.4.19

: Guys, Sean Bright wrote: Steve Totaro wrote: Should one have to change their dialplan for functionality to remain the same in the same version? I wasn't suggesting it wasn't a regression, just making the OP aware that he can pass multiple arguments to a dialplan application (i.e

Re: [asterisk-users] Monitor v/s MixMonitor

___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Sean Bright [EMAIL PROTECTED

Re: [asterisk-users] [asterisk-dev] Locking, coding guidelines addition

Russell Bryant wrote: You have yet to bring any useful discussion to the table. If I were you, I wouldn't hold my breath waiting for any, either :) -- Sean Bright [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api

Re: [asterisk-users] [asterisk-dev] Locking, coding guidelines addition

in order to improve it's scalability? -- Sean Bright [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing

Re: [asterisk-users] asterisk 1.4.21.1 seg fault

Jerry Geis wrote: What should I do now? silly me it is 1.4.21.1 not 1.2.21.1 If you haven't already, I'd suggest reporting an issue in mantis. http://bugs.digium.com/ -- Sean Bright [EMAIL PROTECTED] ___ -- Bandwidth and Colocation

Re: [asterisk-users] Unable to run make menuselect for asterisk-addons

/config.log /path/to/asterisk/menuselect/config.log Thanks, -- Sean Bright [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http

Re: [asterisk-users] Unable to run make menuselect for asterisk-addons

Jonn R Taylor wrote: I am unable to run make menuselect for asterisk-addons. Works fine for zaptel and asterisk. Here is the output. Sorry, your asterisk-addons (not asterisk) and menuselect config.log files. Thanks, -- Sean Bright [EMAIL PROTECTED

Re: [asterisk-users] asterisk 1.6.0rc6 make menuselect failed.

as attachments to the bug? I've seen this problem crop up before and I would like to get it worked out. Thanks, -- Sean Bright [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25

Re: [asterisk-users] asterisk 1.6.0rc6 make menuselect failed.

Thomas Kenyon wrote: Sean Bright wrote: Thomas Kenyon wrote: In trying to upgrade my test machine from 1.6.0beta9 to 1.6.0rc6 when I try to make menuseletc I get the following error. This is using gcc 4.1, libgtk 2.0, on an intel Core2Duo machine running an up to date Debian etch

Re: [asterisk-users] SVN 1.6.0 / current does not compile

Stefan Gofferje wrote: [CC] chan_agent.c - chan_agent.o chan_agent.c: In function ‘unload_module’: chan_agent.c:2496: error: void value not ignored as it ought to be make[1]: *** [chan_agent.o] Error 1 make: *** [channels] Error 2 Fixed in SVN r143735. Thanks, -- Sean Bright [EMAIL

Re: [asterisk-users] How can Block a pri channel

this can be achieved with Asterisk or not, I don't know. This may be what you are looking for: http://bugs.digium.com/view.php?id=3450 -- Sean Bright [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com

Re: [asterisk-users] How can Block a pri channel

Dwayne Hubbard wrote: Sean is correct I *never* get tired of hearing/reading that. -- Sean Bright [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona

Re: [asterisk-users] DAHDI aaaaaaaaaaaaaaarrrrrrrrrghhhhhhhhh :((((

On Thu, Oct 9, 2008 at 7:31 PM, Remco Barendse [EMAIL PROTECTED] wrote: The information (or lack of it) on upgrading from zaptel to that @*^QW%^%!!! dahdi is very frustrating. I cannot find anything on how to uninstall zaptel, i found an earlier post to this list which suggested make

Re: [asterisk-users] Asterisk CDR Analyser

That query appears in call-log.php around line 232. On Thu, Oct 9, 2008 at 10:20 PM, Klaverstyn, David C [EMAIL PROTECTED] wrote: Hi All, I'm stuck and need some help. I have installed the Asterisk CDR Analyser Version 2.0.1. It mostly works except for the CDR Report. I get the

Re: [asterisk-users] Asterisk M$ SQL Server

Try the A$teri$k user'$ li$t. On 4/22/07, Callum McGillivray [EMAIL PROTECTED] wrote: Hi all, Has anyone successfully set up asterisk to query a M$ SQL Server? I'd like to be able to query one in the dial plan and use the results to tamper with call priorities / CLID etc. If someone could

Re: [asterisk-users] Asterisk M$ SQL Server

Well this is the user's list. Apparently I'm a jacka**. Time for bed. On 4/22/07, Sean Bright [EMAIL PROTECTED] wrote: Try the A$teri$k user'$ li$t. On 4/22/07, Callum McGillivray [EMAIL PROTECTED] wrote: Hi all, Has anyone successfully set up asterisk to query a M$ SQL Server? I'd

Re: [asterisk-users] Re: FastAGI hangs up channel if server is not available

Woops. The documentation problem was my fault. I was basing that on a patched version of asterisk that jumped to another priority on FastAGI failure. I'll update the wiki. Sean On 5/17/07, Tony Mountifield [EMAIL PROTECTED] wrote: In article [EMAIL PROTECTED], Lee Jenkins [EMAIL PROTECTED]

Re: [asterisk-users] make: expand.c:489: allocated_variable_append: Assertion `current_variable_set_list-next != 0' failed.

You need to upgrade `make` to at least v3.8 On 2/1/07, Dennis Kavadas [EMAIL PROTECTED] wrote: hi all i'm getting the below error when trying to compile asterisk-1.4 on redhat-9.0 any suggestions ? make[2]: Leaving directory `/usr/local/src/asterisk-1.4.0/menuselect' make[1]: Leaving

Re: [asterisk-users] make: expand.c:489: allocated_variable_append: Assertion `current_variable_set_list-next != 0' failed.

Sorry, hadn't noticed this was already answered. On 2/5/07, Sean Bright [EMAIL PROTECTED] wrote: You need to upgrade `make` to at least v3.8 On 2/1/07, Dennis Kavadas [EMAIL PROTECTED] wrote: hi all i'm getting the below error when trying to compile asterisk-1.4 on redhat-9.0 any

Re: [asterisk-users] FW: Seamless Multi Office Asterisk Deployment

Why does everyone want to go off-list? Is this not information that could benefit others? On 3/12/07, Bruce Reeves [EMAIL PROTECTED] wrote: Brandon Your on the right track with what is can do. It will also be good to look into what kind of QOS you can do on the T-1 connections between

[asterisk-users] voip-info.org is back!

Looks like the site is back up. Don't all hit it at once, it might go down again ;-) Sean ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:

Re: [asterisk-users] asterisk 1.2 branch revision 53132 failed to compile

I would check to see if you had the latest version of the zaptel library for the version of asterisk you are trying to compile. On 3/16/07, Wilson Pickett [EMAIL PROTECTED] wrote: Has this issue been resolved? I'm having the problem now with the code base downloaded yesterday. On 2/3/07,

Re: [asterisk-users] asterisk 1.2 branch revision 53132 failed to compile

Whats the non-workaround solution? Is there one? On 3/16/07, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Fri, Mar 16, 2007 at 02:46:24PM +0100, Wilson Pickett wrote: Has this issue been resolved? I'm having the problem now with the code base downloaded yesterday. On 2/3/07, Erick Perez

Re: [asterisk-users] Cepstral voices

Best code comment ever, by the way: here's a for loop for i := 1 to iLen do :-) On 3/16/07, Lee Jenkins [EMAIL PROTECTED] wrote: Steve Prior wrote: Julian Lyndon-Smith wrote: what input text ? To what application ? I agree completely with the app_swift suggestion from loopfree as

Re: [asterisk-users] How to get AEL2

The latest release of Asterisk is 1.4.1, or am I missing something? On 3/21/07, Rizwan Hisham [EMAIL PROTECTED] wrote: Hi, im using asterisk 1.4.2. i need to use the AEL2. its not included in 1.4.2. how do i get it? is there any patch for this? -- Regards Rizwan Hisham Software Engineer

Re: [asterisk-users] How to get AEL2

Apparently I was missing something :-) Just saw the mailing list message about 1.4.2. On 3/21/07, Sean Bright [EMAIL PROTECTED] wrote: The latest release of Asterisk is 1.4.1, or am I missing something? On 3/21/07, Rizwan Hisham [EMAIL PROTECTED] wrote: Hi, im using asterisk 1.4.2. i

Re: [asterisk-users] How to get AEL2

Are you sure its not included? I ran 'make menuselect' and under PBX Modules its the first thing listed for me. On 3/21/07, Rizwan Hisham [EMAIL PROTECTED] wrote: its on the ftp link here ftp://ftp.digium.com/pub/asterisk , it was put on yesterday On 3/21/07, Sean Bright [EMAIL PROTECTED

Re: [asterisk-users] Queue application strategy

If you are using Asterisk 1.4 you should look at the autofill configuration option in queues.conf. For versions prior to that, I'm not sure there is a solution. On 4/4/07, Jordan Novak [EMAIL PROTECTED] wrote: I am using rrmemory for my queues. I have noticed that the application will only

Re: [asterisk-users] Polycom and Asterisk

Maybe the firmware uses GPL'd code? ;-) Just a theory, don't sue me Polycom! On 4/5/07, Stephen Bosch [EMAIL PROTECTED] wrote: Andrew Joakimsen wrote: Well I would wonder how Polycom even had any idea whom your vendor is. The vendor made a request for 2.1.0 on my behalf and let it slip that

Re: [asterisk-users] Re: asterisk-users Digest, Vol 33, Issue 12

Since when is Canada part of the rest of the world? I thought it was a US National Park? ;-) On 4/5/07, Stephen Bosch [EMAIL PROTECTED] wrote: john beaman wrote: I too was curious about this, so I copied the text into Babel Fish, and this is the result: I miss of the 2/04/2007 to the

Re: [asterisk-users] Strange error, logger.c: No more room in scheduler...

Does this occur in the latest 1.2.17 release? On 4/10/07, Massimo Nuvoli [EMAIL PROTECTED] wrote: I found no info about this strange error: logger.c: No more room in scheduler logger.c: Asked to delete sched id -1??? Only in verbose mode. Someone know how to solve this? Asterisk 1.2.13 with

Re: [asterisk-users] Execute EAGI script with params from extensions.conf

You need to pass the arguments as separate arguments to the EAGI dialplan application, eg: exten = 492,2,EAGI(InfMsg,-s,1) but I would recommend using pipes... exten = 492,2,EAGI(InfMsg|-s|1) But maybe that's just me. On 4/11/07, equis software [EMAIL PROTECTED] wrote: How can I execute an

Re: [asterisk-users] Execute EAGI script with params fromextensions.conf

Yes you can. Read: http://www.voip-info.org/wiki-Asterisk+AGI On 4/11/07, Griepentrog Scott [EMAIL PROTECTED] wrote: I don't think you can put arguments to the agi. Try is as: exten = 492,2,eagi,InfMsg -Original Message- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED]

Re: [asterisk-users] Why does Asterisk not hangup?

Klaus Darilion wrote: Ok. Just for the info to others: the 10 seconds are hardcoded in pbx.c What line in which version? -- Sean Bright sean.bri...@gmail.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk

Re: [asterisk-users] how to write svn for dahdi-linux and dahdi-tools when using svn 1.4

development version): # svn co http://svn.digium.com/svn/dahdi/linux-complete/trunk/ dahdi-trunk Or one of the tags: # svn co http://svn.digium.com/svn/dahdi/linux-complete/tags/2.2.0-rc1+2.2.0-rc1/ (The line above may wrap) -- Sean Bright sean.bri...@gmail.com

Re: [asterisk-users] Update Caller-ID after Dial()

Benny Amorsen wrote: Does the patch use the non-standard Remote-Party-ID or the proper P-Asserted-Identity? That depends on the value specified for 'sendrpid' in sip.conf. sendrpid=pai ; This will use P-Asserted-Identity sendrpid=rpid ; This will use Remote-Party-ID (the default) -- Sean

Re: [asterisk-users] Installing LUA

/1463560; | patch -p0 $ ./configure $ make menuselect Let me know if that works for you. -- Sean Bright sean.bri...@gmail.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update

Re: [asterisk-users] Installing LUA

tarball (or if you have pulled from SVN just do an 'svn revert configur*')). $ cd path/to/asterisk/src $ wget -O - http://pastebin.ca/raw/1463619; | patch -p0 $ ./configure $ make menuselect Let me know. -- Sean Bright sean.bri...@gmail.com

Re: [asterisk-users] Installing LUA

I supplied will find LUA at configure time but not compile time. This needs some more thought. -- Sean Bright sean.bri...@gmail.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE

Re: [asterisk-users] chan_mobile

The version of chan_mobile in trunk is only compatible with the trunk version of Asterisk, not the 1.4 branch. Whoever maintains the patches on chan-mobile.org will not to update in order for you to compile the trunk version of chan_mobile against Asterisk 1.4. Sean On 10/30/07, Alejandro

Re: [asterisk-users] MySQL() timeout

Find this line: if (mysql_real_connect(mysql, dbhost, dbuser... Add this before that line: int timeout = 10; /* 10 second timeout */ mysql_options(mysql, MYSQL_OPT_CONNECT_TIMEOUT, (const char *) timeout); And recompile. On 10/31/07, Doug Lytle [EMAIL PROTECTED] wrote: Tilghman Lesher

Re: [asterisk-users] AEL2 and Callbacks

Do a 'core show dialplan' and see what the AEL is generating. On 11/1/07, Douglas Garstang [EMAIL PROTECTED] wrote: - Original Message From: Richard Lyman [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday,

Re: [asterisk-users] AEL2 and Callbacks

Douglas Garstang wrote: I am originating a command via the AMI with this... Doug, Were you ever able to resolve this? If so, could you share what the issue was? Thanks, Sean ___ --Bandwidth and Colocation Provided by http://www.api-digital.com--

Re: [asterisk-users] How to play Asterisk .raw file

Gary wrote: I used ChanSpy( ) recorded some test conversations. It has .raw extension. What kind of audio file is this? How can I play it? Gary I believe .raw files are slinear (signed linear). They are effectively wav files without a header. You can use sox to convert them to your

Re: [asterisk-users] Recording calls again

. Where encounter information of how doing it? Are you using Google Translate or Babelfish? -- Sean Bright sean.bri...@gmail.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona

Re: [asterisk-users] install the digium card TC400P howto?

wcb4xxp: done wctc4xxp: done xpp_usb: done wctc4xxp: done -- That's the module for the card. So it is loading. -- Sean Bright sean.bri...@gmail.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009

Re: [asterisk-users] install the digium card TC400P howto?

BERGANZ François wrote: What have I to do? Nothing. The system recognizes the card, and the appropriate module is loading. What is happening that makes you think it isn't working? -- Sean Bright sean.bri...@gmail.com ___ -- Bandwidth

Re: [asterisk-users] install the digium card TC400P howto?

Steve Totaro wrote: No hardware timing source found in /proc/dahdi, loading dahdi_dummy would make me think it is not loading correctly. The TC400P is a transcoder card. It is not a timing source. -- Sean Bright sean.bri...@gmail.com

Re: [asterisk-users] install the digium card TC400P howto?

BERGANZ François wrote: Where have I to insert the mode=g729 ? That depends on the platform. On Ubuntu I added it to /etc/modprobe.d/dahdi: options wctc4xxp mode=g729 -- Sean Bright sean.bri...@gmail.com ___ -- Bandwidth and Colocation

Re: [asterisk-users] help with picking out a digium card.

On 1/17/2010 3:25 PM, shawn bright wrote: Hey all, i love your name, btw. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:

Re: [asterisk-users] FreeTDS (Microsoft MsSQL 2008) and CDR

On 7/26/2010 4:05 AM, Andraž wrote: I have Ubuntu server 10.04 64bit, and Asterisk 1.4.34, compiled from sources. I installed freetds-common,freetds-dev, libct4, libsybdb5, freetds-bin, but, when I run configure and then make menuconfig in section Call Detail Recording - cdr_tds it's

Re: [asterisk-users] Convert wav-file to alaw-file

On 8/17/2010 9:00 AM, Jonas Kellens wrote: Hello list, it seems that Asterisk is unable to convert a wav-file into an alaw-file : [r...@asterisk testing]# asterisk -rx file convert testExtended2.wav testExtended2.alaw Unable to open input file: testExtended2.wav [r...@asterisk testing]#

Re: [asterisk-users] Stuck "channel"

On 10/31/2019 2:13 PM, Carlos Chavez wrote: I assume this is something created by Freepbx.  If I do a "channel request hangup" it tells me the channel does not exist. Any ideas Are you trying to hang up "Message/ast_msg_queu" or are you hitting the tab key to complete it in the CLI?

Re: [asterisk-users] PJSIP crashes

On 2/25/2020 12:40 PM, Saint Michael wrote: PJISP cannot handle the From  field when it does not contain a number. Sure it can, but: sip:Radefeld Dental@8.38.43.67 Is not a valid SIP URI (it can't contain a space). Is Asterisk actually "crashing" or are you just seeing this error in your

Re: [asterisk-users] PJSIP crashes

On 2/26/2020 5:06 PM, Saint Michael wrote: PJSIP should log a warting and continue. That's exactly what it is doing unless I am misunderstanding. You didn't answer my question last time - is Asterisk actually "crashing?" It is causing the CPU usage to spike dramatically. If you are able

Re: [asterisk-users] PJSIP and Grandstream Wave with TSL and SRTP

On 1/23/2020 6:04 PM, hw wrote: This is what mine looks like which works just fine: [transport-tls] type = transport protocol = tls method= tlsv1_2 cipher= ECDHE-ECDSA-AES256-GCM-SHA384,ECDHE-RSA-AES256-GCM-SHA384,ECDHE-ECDSA-AES128

Re: [asterisk-users] PJSIP do not challenge 'options' without username. - silence 'notice' on console.

On 1/23/2020 6:24 AM, Benoit Panizzon wrote: Therefore Asterisk PJSIP cannot match an unsername against an endpoint and prints a notice on the console. Is there a way to silence this kind of notice? No, per RFC 3261, authentication is required for OPTIONS requests just like it would be if an

Re: [asterisk-users] PJSIP and Grandstream Wave with TSL and SRTP

On 1/21/2020 9:18 PM, hw wrote: [transport-tls] type = transport protocol = tls bind = 0.0.0.0:5061 tos = cs5 cert_file = /etc/asterisk/cert/asterisk.pem ca_list_file = /etc/pki/tls/certs/ca-bundle.crt method = sslv23 This is what mine looks like which works just fine: [transport-tls] type

Re: [asterisk-users] Multicast codec

On 2/16/2020 5:38 PM, Jerry Geis wrote: I am trying to find out what codec is used in the asterisk multicast ? It is µ-law by default, but you can override it in the dial string¹. Note that if you do override it, you would need to use a codec that has a static RTP payload code² as no SDP is

Re: [asterisk-users] res_calendar & LetsEncrypt

On 12/24/2019 9:02 AM, Doug Lytle wrote: [Dec 24 07:48:46] WARNING[10679] res_calendar_caldav.c: Unknown response to CalDAV calendar calendar.name.here, request REPORT to /dav/username/Calendar: Server certificate changed: connection intercepted? Would this be considered a bug, or do I have

Re: [asterisk-users] AGI: "Get variable" returns variable VALUE vs "Get full variable" returns variable NAME - bug or my misunderstanding?

On 12/27/2019 2:24 PM, Jonathan H wrote: AGI Rx << SET VARIABLE myVar "Hello World!!!" AGI Tx >> 200 result=1 AGI Rx << GET FULL VARIABLE myVar AGI Tx >> 200 result=1 (myVar) Is this a bug, poor documentation, or my poor understanding of them? I believe the syntax you are looking for is:    

Re: [asterisk-users] AGI: "Get variable" returns variable VALUE vs "Get full variable" returns variable NAME - bug or my misunderstanding?

On 12/27/2019 2:56 PM, Jonathan H wrote: OK, that works - looks like a documentation bug? (Also very confusing!) Should I report it on the page at https://wiki.asterisk.org/wiki/display/AST/Asterisk+17+AGICommand_get+full+variable or on the main tracker? These wiki pages are automatically

Re: [asterisk-users] [SOLVED]Re: TLS/SSL error loading cert file.

Hi, On 4/17/2020 10:34 AM, Olivier wrote: All this came from ast_tls_cert script using 1024 bits-long keys where Debian's defaut was to require at least 2048-long keys ! Simply passing -b 2048 to ast_tls_cert solved it. Yes, this was addressed by two¹ commits² in the most recent releases

Re: [asterisk-users] multicast codec

Should be ulaw On Wed, Apr 1, 2020 at 11:02 AM Jerry Geis wrote: > What is the default multicast codec for multicast in Asterisk 13 ? > > G.729 or G.711 or other ? > > Jerry > -- > _ > -- Bandwidth and Colocation Provided by

Re: [asterisk-users] Troubleshooting load issues

On 4/22/2020 2:55 PM, Dovid Bender wrote: All the calls are using ulaw. The files that I am playing are gsm. I suppose doing a file convert with sox to .ulaw may help You should absolutely do this. -- _ -- Bandwidth and

Re: [asterisk-users] call an IP camera?

On 9/24/2020 12:28 PM, hw wrote: What are the requirements for the URLs that can be used with the 'playlist' option in musiconhold.conf? HTTP(S) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com --

Re: [asterisk-users] memory issues

https://issues.asterisk.org/jira/browse/ASTERISK-28695 On Fri, Sep 25, 2020 at 2:32 PM hw wrote: > > > Hi, > > > > ever since I have switched my server from Centos 7 to Fedora 32, asterisk > > is showing memory issues and no calls are possible. I'm using the asterisk > > that comes with

Re: [asterisk-users] CDR mysql: timeout when remote database unavailable

On 6/7/2020 2:54 AM, Fourhundred Thecat wrote: I would still like to know where the Aterisk mysql timeout duration comes from, and whether it can be configured. In the case of cdr_mysql, the connect timeout is configurable by putting the following in cdr_mysql.conf: [global] timeout = 5 ;

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

Hi, On 1/26/2021 3:12 PM, Ruisheng Peng wrote: Transport: transport-tls: cert_file /home/asterisk/certs/asterisk.crt is either missing or not readable This error means that the file either does not exist or that Asterisk is not able to open it for reading. In your case it looks like the file

Re: [asterisk-users] ADSI - Unable to send CAS...

On 6/7/2021 6:59 AM, Antony Stone wrote: > It makes no sense to me that Asterisk is even considering ADSI (where A stands > for Analogue) on a SIP call. How can I disable this so that the error does > not occur? Don't load the res_adsi.so module. You can add the following to modules.conf:    

Re: [asterisk-users] MulticastRTP and ttl

On 5/12/2021 9:37 AM, Jerry Geis wrote: > Sorry it - may have worked - my person only used a single / not // > Thanks! > > Does this work on version 13 or just version 18 ? In terms of supported versions of Asterisk it works in 16+ Kind regards, Sean --

Re: [asterisk-users] MulticastRTP and ttl

On 5/12/2021 9:19 AM, Jerry Geis wrote: > I tried the 239.1.2.3:20480//t(5) and still using a default of 1. > Is there a config file to set this default TTL ? No, just the syntax I already suggested. It's documented here:

Re: [asterisk-users] MulticastRTP and ttl

On 5/11/2021 4:24 PM, Jerry Geis wrote: > I was using asterisk 13.36.0 and tried to specify a MulticastRTP TTL with > Channel: MulticastRTP/basic/239.1.2.3:20480/5 > where 5 is the ttl Try: MulticastRTP/basic/239.1.2.3:20480//t(5) Kind regards, Sean--

Re: [asterisk-users] ControlPlayBack

On 6/30/2021 9:50 AM, Dovid Bender wrote: > Yes that works. It's an "ugly hack". Would this be classified as a bug > or feature? It's an existing bug: https://issues.asterisk.org/jira/browse/ASTERISK-27871 Kind regards, Sean --

Re: [asterisk-users] ControlPlayBack

On 6/30/2021 8:55 AM, Dovid Bender wrote: > [2021-06-30 08:46:43] WARNING[9661][C-000c8eaa]: file.c:779 > ast_openstream_full: File http://localhost/test.gsm?foo=bar > does not exist in any format > [2021-06-30 08:46:43] WARNING[9661][C-000c8eaa]: file.c:1252 >

Re: [asterisk-users] Strange Codes on Asterisk command line

On 10/24/2021 12:41 PM, cio-al...@playerschool.edu wrote: > Really destroying SIP dialog > '5af3bcf012ac9d574b17f2634e48de54@65.21.137.162:5060' Method: OPTIONS > \U+26504\U+2650A\U+26565\U+26578\U+26569\U+26574\U+2650A\U+2650A If this is still Asterisk 11 as you've mentioned in other threads,

Re: [asterisk-users] SAY_DTMF_INTERRUPT

On 12/23/2021 2:47 PM, Dovid Bender wrote: > Has anyone gotten SAY_DTMF_INTERRUPT to work? Issue created[1] and tentative patch submitted for review[2]. [1] https://issues.asterisk.org/jira/browse/ASTERISK-29816 [2] https://gerrit.asterisk.org/c/asterisk/+/17712 Kind regards, Sean--

Re: [asterisk-users] SAY_DTMF_INTERRUPT

On 12/23/2021 2:47 PM, Dovid Bender wrote: > Has anyone gotten SAY_DTMF_INTERRUPT to work? In 19.0.1 if I do core show > function SAY_DTMF_INTERRUPT it does not show up. Do you mean 19.1.0? It is neither a function nor an argument to the CHANNEL() function - it's just a channel variable, so:

Re: [asterisk-users] Strange Codes on Asterisk command line

On 10/25/2021 1:19 PM, cio-al...@playerschool.edu wrote: > Yes, it's Asterisk 11 on Centos 8, not Centos 7. > Centos 7 cannot be used anymore because of some systemd incompatibility > Can somebody be so nice as to provide a patch for the latest version of > Asterisk 11? Just add --without-libedit

Re: [asterisk-users] extensions.conf [General] settings

On 8/22/2022 11:16 AM, Antony Stone wrote: > Is there any way to find out the values of variables set in the [General] > section of extensions.conf from the Asterisk CLI No. You could use the `!` command to less or cat extensions.conf, but there is no built-in command for it. Kind regards,

Re: [asterisk-users] Multicast codec

On 9/7/2022 1:06 PM, Jerry Geis wrote: > I think multicast uses codec g711 pcmu > is there any way to change or set the codec I want to use - like g722 ? The CHANGES entry for the MulticastRTP channel type[1] has some info on this. Basically you would add the 'c' option to your dial string:    

Re: [asterisk-users] Confbridge for 80 devices

On 10/20/2022 5:17 PM, Jerry Geis wrote: > What is the trick to get "preload => res_timing_dahdi" working ? [modules] autoload = yes noload = res_timing_pthread noload = res_timing_timerfd However, it's unlikely to be a timing problem. Can you share your ConfBridge configuration? Kind

Re: [asterisk-users] Confbridge for 80 devices

On 10/20/2022 5:35 PM, Jerry Geis wrote: > > ;dsp_drop_silence=yes  ; This option drops what Asterisk detects as > silence from >                        ; entering into the bridge.  Enabling this > option will drastically >                        ; improve performance and help remove the > buildup

Re: [asterisk-users] Global variables in global variables

On 1/26/2023 5:16 AM, Antony Stone wrote: > It does not work if it's written in AEL - assigning global variables works, > but the above does not. I've created a JIRA issue[1] for this as well as a proposed patch[2]. Assuming all goes well this should work in future releases. Kind regards, Sean

Re: [asterisk-users] cdr_sqlite3

On Sat, Mar 4, 2023 at 1:29 PM, Fourhundred Thecat <400the...@gmx.ch> wrote: > /var/log/asterisk/master.db > > how can I change the location ? > > If this is not possible to change in the config file, where in the > source code would I change that? cdr/cdr_sqlite3_custom.c line 311 Kind

Re: [asterisk-users] cdr_sqlite3

On 3/28/2023 4:03 AM, Fourhundred Thecat wrote: > > On 2023-03-04 23:11, Sean Bright wrote: > > > > cdr/cdr_sqlite3_custom.c line 311 > > Hello, > > I asked here recently how to change the location where > "cdr_sqlite3_custom" stores the sqlite databa

Re: [asterisk-users] AMI versions

https://docs.asterisk.org/latest/Configuration/Interfaces/Asterisk-Manager-Interface-AMI/Asterisk-Manager-Interface-AMI-Changes/ On Tue, Jul 11, 2023 at 11:54 AM, TTT <[li...@telium.io](mailto:On Tue, Jul 11, 2023 at 11:54 AM, TTT < wrote: > Is there a web page that lists the AMI versions

Re: [asterisk-users] Problems solved

On 5/26/2023 5:41 PM, Steve Matzura wrote: > Doug from this list got me to change my connectivity to my DID provider > from SIP to IAX, and bingo, it all just worked instantly. Looking over your previous messages and the error you were receiving (the one referring to extension 's') it looks like

Re: [asterisk-users] SetCallerPres command gone

On 7/1/2023 11:40 AM, TTT wrote: > I thought it was replaced with CALLERPRES(allowed) but this generated an > error too in Asterisk 20. From UPGRADE.txt¹:     The CALLERPRES() dialplan function is deprecated in favor of CALLERID(num-pres) and CALLERID(name-pres). Kind regards, Sean 1.