[asterisk-users] Asterisk and Teams integration?

    Does anyone know of a good solution to integrate Asterisk and MS Teams?  Something where you can use the MS Teams client as a regular extension? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez +52 (55)8116-9161 --

Re: [asterisk-users] Deleting voicemail by program

    The script included with Asterisk (messages-expire.pl) deletes older messages and then renumbers the rest of the messages.  I guess you need to do the same. On 09/10/23 2:24 PM, Mike Diehl wrote: Unfortunately, I'm using a version of asterisk that is old enough to not benefit from this...

[asterisk-users] Can ShanSpy be used on Local Channels?

    Does anyone know if Chanspy can be used with local channels? Since agents on queues need to use local channels like Local/@from-queue/n to make sure that all of their registered extensions ring we are now having a problem trying to use ChanSpy to listen to calls.  Since we do not know

[asterisk-users] Is there a good Python library for AMI?

    I am switching many of my scripts to python and I found pyst2 in my search for an Asterisk library.  While it seems to work well for AGI acripts it seems very broken when using it for AMI.  Can anyone recommend a good and currently supported AMI library for python? -- Telecomunicaciones

Re: [asterisk-users] Asterisk not replacing private FROM ip with public IP in INVITE

You need to put your external IP in the transport configuration: external_media_address=X.X.X.X external_signaling_address=X.X.X.X external_signaling_port=5060 On 21/06/23 12:36, TTT wrote: I've split this thread off from another (PJSIP authentication) because I think the root cause is

Re: [asterisk-users] PJSIP not performing outbound authentication

    Dis you set "outbound_auth" in your endpoint configuration to Twilio? On 21/06/23 11:19, TTT wrote: I am using Asterisk 20.3.0 with PJSIP. I have setup a trunk to my ISP (Twilio) who requires outbound authentication. My pjsip.auth.conf contains: [Twilio] type=auth auth_type=userpass

Re: [asterisk-users] Multiple phones on same PJSIP account

    Are you using PJSIP_DIAL_CONTACTS in your dialplan?  You need to use that in the Dial command and not just PJSIP/XXX for all extensions to ring. On 21/06/23 9:52, TTT wrote: Ok I've got multiple phone sets registered with the same extension/secret. However, this causes a strange problem.

[asterisk-users] MixMonitor not recording through transfer

    I have the following scenario: Agent calls external number Mixmonitor starts recording call After agent speaks with customer they need to transfer them to an extension that will simply play a message Customer hangs up     The problem is that the recording stops the moment the agent

Re: [asterisk-users] Run asterisk -rx "command" and get plain text output

ns only have colorized output when your are connected to the console (-r) not for remote commands (-rx) On Wed, Aug 3, 2022 at 08:21 Carlos Chavez wrote: I usually like to have the colorized output when looking at asterisk output but I need to get some info by running "asteri

[asterisk-users] Run asterisk -rx "command" and get plain text output

    I usually like to have the colorized output when looking at asterisk output but I need to get some info by running "asterisk -rx" and get just plain text output so I can mail it.  Right now I get ANSI codes in the output.  Is there a way to get plain text output for just that script and

Re: [asterisk-users] Installing and configuring Opus?

    If you compiles Asterisk by hand you need to make sure that codec_opus was selected (make menuconfig to check selections).  If you installed it from another source make sure that Opus is included (maybe an extra package).  Also, make sure that you modules.conf file is not explicitly

[asterisk-users] How to use mixmonitor when transfering a call

    I am having a problem with my recordings.  Mixmonitor is called in the "macro" when you dial an extension.  If that call is transferred to another extension then the recording is reset and we lose the recording for the original call.  How can I tell Mixmonitor to keep recording and not

Re: [asterisk-users] R2 error Seize Timeout

. Regards, Hans Am 08.03.22 um 06:41 schrieb Carlos Chavez:     Last month we switched a Panasonic pbx with a Freepbx 16 appliance.  We use a single E1 in MFC/R2 (Mexico) with Telmex as a provider.  This was connected for a couple of days for testing with no problems before the client moved offices

[asterisk-users] R2 error Seize Timeout

    Last month we switched a Panasonic pbx with a Freepbx 16 appliance.  We use a single E1 in MFC/R2 (Mexico) with Telmex as a provider.  This was connected for a couple of days for testing with no problems before the client moved offices to a new location.  In the new location we are now

[asterisk-users] Local channel sometimes have no audio

    We recently upgraded a very old server from Asterisk 1.8 to 18.9 and we are having a strange issue with calls in queues.  We use Queuemetrics to manage our agents and extensions are configured as Local/@from-queue/n to connect clients to agents.  This is because if you dial PJSIP/

Re: [asterisk-users] automating "make menuselect" options when building

On 08/11/21 11:53, Kingsley Tart wrote: Hi, I realise that this is not really specific to Asterisk, but this seems as sensible a place to ask as any. If I want to create a script to automate the build of my chosen Asterisk setup, what's the best way to automate my selections that I did

Re: [asterisk-users] Delay when dialing...

    Thank you.  The server is running dnsmasq locally for DNS resolution and all queries resolve properly.  I just added the hostname to /etc/hosts and restarted but the delay persists. On 7/23/2021 1:41 AM, Jean Aunis wrote: Le 22/07/2021 à 18:32, Carlos Chavez a écrit :     I started

[asterisk-users] Delay when dialing...

    I started noticing a few days ago that whenever I dial any number or extension there is a delay of 5 to 10 seconds before Asterisk reacts.  I see nothing on the CLI for that time and then the call goes through.  I have checked my network to make sure there is nothing slowing down packets

[asterisk-users] PJSIP_DIAL_CONTACTS and Queues

    Is there a solution to dial multiple contacts for a Queue agent?  Since the pandemic started many of our customers have begun to move agents off site.  Since most of them were using softphones we did not have any problems but now we have one case where the agents have a desk phone in their

Re: [asterisk-users] Some calls drop after 30 seconds

On 08/09/20 4:16, Joshua C. Colp wrote: On Mon, Sep 7, 2020 at 9:35 PM Carlos Chavez <mailto:cur...@telecomab.mx>> wrote: Some users have complained that their calls drop after about 30 seconds.  Not all, just some.  After looking at the log files the only differe

[asterisk-users] Some calls drop after 30 seconds

    Some users have complained that their calls drop after about 30 seconds.  Not all, just some.  After looking at the log files the only difference I can find from the dropped calls is the following line: [2020-09-07 11:29:59] VERBOSE[21666][C-0055] bridge.c: Bridge

Re: [asterisk-users] Channels freeze on Confbridge

On 25/08/20 7:20, Andrew Yager wrote: On Sun, 23 Aug 2020 at 18:23, Antony Stone > wrote: On Saturday 22 August 2020 at 22:51:48, Sebastian Nielsen wrote: > I had a similiar problem, but with calls dropping after 30 sec. > It turned

[asterisk-users] Channels freeze on Confbridge

    I am having a strange problem.  We have an Asterisk 16.12.0 server (we have upgraded at least two versions since we found the problem) where users complain that confbridge calls end after about 30 minutes or so.  The problem is that according to Asterisk the calls are still active.  All

[asterisk-users] One way audio on outgoing calls

    I am having a strange problem with a new provider.  We already have a couple SIP trunks working fine.  We are trying a new provider but we are having one way audio problems with outgoing calls.  Incoming calls do have two way audio, only outgoing calls have this problem.  I do not see

Re: [asterisk-users] No CID between Asterisk using IAX trunk

    Could the difference be that you need to use type=friend for CID to work?  Using type=peer we can forgo auth since we are not using public infrastructure.  My other trunks do not have allowcallerid=yes so I will add that and test it.  Thanks. On 02/03/20 12:54, Doug Lytle wrote: My

Re: [asterisk-users] No CID between Asterisk using IAX trunk

    Not these particular two servers. On 02/03/20 12:16, Doug Lytle wrote: I am trying to troubleshoot two Asterisk servers that have an IAX2 trunk between them. Carlos, Had caller-id ever worked between these two systems? Doug -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos

[asterisk-users] No CID between Asterisk using IAX trunk

    I am trying to troubleshoot two Asterisk servers that have an IAX2 trunk between them.  Calls come and go but there is no CallerID from the remote server either way.  One of the servers is running Asterisk 16 and the other is an older 1.8 install (I know, I am trying to get permission to

[asterisk-users] DTMF not working on incoming calls

    What is  the best way to debug DTMF on a PJSIP trunk?  I have cycled through all available options ('rfc4733','inband','info','auto','auto_info') but my IVR does not recognize any options from the remote end. I have also tried changing codecs from g729 to alaw or ulaw with the same result. 

Re: [asterisk-users] Stuck "channel"

    So a restart is the only way to get rid of it? On 11/1/2019 9:28 AM, Richard Mudgett wrote: On Thu, Oct 31, 2019 at 11:05 PM Carlos Chavez <mailto:cur...@telecomab.mx>> wrote: I have tried both by hand and hitting tab to auto complete: *CLI> channel request ha

Re: [asterisk-users] Stuck "channel"

I have tried both by hand and hitting tab to auto complete: *CLI> channel request hangup Message/ast_msg_queue Message/ast_msg_queue is not a known channel On 31/10/19 14:18, Sean Bright wrote: On 10/31/2019 2:13 PM, Carlos Chavez wrote: I assume this is something created by Freepbx.  If I

[asterisk-users] Stuck "channel"

    Since yesterday I have a stuck channel on my Asterisk server and I do not know how to eliminate it: Message/ast_msg_queu macro-dial-one   s  59 Up  Dial PJSIP/1218/sip:1218@192.1 17:24:07     I assume this is something created by Freepbx.  If I do a

Re: [asterisk-users] Asterisk and CentOS 8

    They only problem I have found so far is while trying to install Alembic for SQLAlchemy (for realtime configs).  Those are the only packages that I cannot get working properly.  Vanilla Asterisk works fine  with the only extra package needed being libedit-devel that is not included in any

Re: [asterisk-users] Polycom BLF Question

    This is done via the custom extension state or hints. Basically you create a custom hint for 444 and monitor that on your phone like any other extension.  You then enable or disable the hint in the same dialplan for 444 and 555.

Re: [asterisk-users] Wanted: professional softphone

On 7/24/19 11:41 PM, Michael Maier wrote: Hello! Does anybody by chance know of a softphone which additionally has a management suite to fully configure it userID based for Windows clients? Any idea is appreciated!     Zulu from Sangoma allows you to generate a QR code that configures

[asterisk-users] Calling GOSUB from Macro on Asterisk 1.8

    I know we should not be running an Asterisk so old but this customer really does not want to replace this particular installation.  I am having a problem when calling Gosub from a macro.  It seems that if I call gosub and return to the macro all Macro related variables like MACRO_EXTEN and

Re: [asterisk-users] why doesn't extension "s" work ?

On 3/28/2019 6:32 PM, sean darcy wrote: I'm using "s" extension in my dialplan: [gv-voice] exten => s,1,Verbose(callerid is "${CALLERID(all)}" or "${CALLERID(num)}") ;Set(Var_TO=${SIP_HEADER(TO)})  ; PJSIP_HEADER(read,To)    same=>n, But when a call comes in to the gv-voice context,

Re: [asterisk-users] AMI mulitple calls quickly

On 3/12/19 11:03 AM, Steve Edwards wrote: On Mon, 11 Mar 2019, Jerry Geis wrote: If I use the AMI interface to originate a call, close the connection, open another connection etc...This works. but is slow... Would opening multiple AMI connections be an option?     You should be able to

Re: [asterisk-users] Various extensions ring once and go to voicemail

//www.ridemcts.com/>___ 1942 N 17th Street | Milwaukee, WI  53205 Check us out on Facebook <https://www.facebook.com/mcts> & Twitter <https://twitter.com/RideMCTS> *From:*asterisk-users *On Behalf Of *Carlos Chavez *Sent:* Monday, January 14, 2019 4:08 PM *To:* asterisk

Re: [asterisk-users] Various extensions ring once and go to voicemail

On 1/14/19 4:02 PM, Duncan Turnbull wrote: Sent from my iPad On 15/01/2019, at 10:34 AM, Thomas Peters > wrote: Duncan: You may have it right—I took one phone and set the ring time to 60 seconds. I now get about 4 rings on that one. I wonder how I can change

[asterisk-users] Is the R2 list still up?

    I am trying to send messages to asterisk...@lists.digium.com but I do not get an error or any messages back.  In the archive I do not see any messages past November 2018. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez +52 (55)8116-9161 --

[asterisk-users] Problem receiving calls with Telmex in Mexico...

    Hi.  I am having a problem when trying to receive calls via en E1 from  Telmex using MFC/R2 (MX Variant).  Outgoing calls are fine.  We are using a PBXact system with a Digium TE420 (5th Gen) card.  Here is a log from the call: [10:46:37:707] [Thread: 140631230322432] [Chan 1] - Call

[asterisk-users] Asterisk 15 and Cepstral

    It seems that app_swift does not work with Asterisk 15 or 16.  I just get errors when trying to compile: [root@pbxoficina app_swift]# ./configure checking gcc... checking swift... checking asterisk... creating Makefile     *  Now run

Re: [asterisk-users] WebRTC as Softphone substitute ?

On 9/26/18 10:20 AM, Matthew Fredrickson wrote: On Wed, Sep 26, 2018 at 9:40 AM Carlos Chavez wrote: On 9/26/2018 4:46 AM, Olivier wrote: Hello, This morning, I asked myself if WebRTC could be a viable alternative to softphone deployment. For me, main issue with Softphones is the amount

Re: [asterisk-users] WebRTC as Softphone substitute ?

On 9/26/2018 4:46 AM, Olivier wrote: Hello, This morning, I asked myself if WebRTC could be a viable alternative to softphone deployment. For me, main issue with Softphones is the amount of work needed for installation and configuration. Also, Softphones must be carefully choosen if

Re: [asterisk-users] AGI timeout option

On 9/13/2018 8:04 PM, Patrick Wakano wrote: Hello list, Hope you all doing  well! Recently, I had an issue with a FastAGI PHP script, which under some specific situation would run into an infinity loop, consuming all CPU resources. This also was preventing Asterisk to terminated the call

[asterisk-users] Segfault on libasteriskpj.so.2

    I just finished installing a brand new server with CentOS 7.5 and Asterisk 13.22.0 and the minute I a call from the PSTN (from a SIP trunk) bridges with any SIP phone Asterisk crashes: Jul 20 10:59:53 localhost kernel: asterisk[2819]: segfault at 188 ip 7f158b9e047c sp

[asterisk-users] No audio on direct call from trunk to SPA-8000

    I am having one of those days.  We just replaced an old Asterisk 1.8 server with a new Asterisk 13.21.1 (both using Freepbx) and almost everything is working except for some incoming calls directed to a Cisco SPA-8000.  The PSTN trunk is SIP.  Only calls coming from the PSTN to a direct

Re: [asterisk-users] Busy indicator for FXO line or extension

On 6/28/18 5:31 AM, bilal ghayyad wrote: Hello; Is it possible to configure one button on the IP Phone (like Polycom or general SIP Phone) to indicate (at the phone display) that the line (the line that is connected for FXO port) is busy or not? If it is not busy, the user can press on the

[asterisk-users] Setting outgoing CALLERID without changing CDR(src)

    I thought I had found and answer to this question by using CALLERID(ani) but it seems that only works on versions prior to 12.  On Asterisk 13 setting CALLERID(num) before dialing to an external trunk always changes CDR(src) to the number you set and the original extension number that

Re: [asterisk-users] Set external CID but retain internal extension in CDR...

On 2/22/18 4:40 PM, Carlos Chavez wrote: On 2/22/18 3:54 PM, Carlos Chavez wrote: On 2/22/18 3:46 PM, Antony Stone wrote: On Thursday 22 February 2018 at 21:41:41, Carlos Chavez wrote: On 2/22/18 1:07 PM, Antony Stone wrote: On Thursday 22 February 2018 at 19:10:51, Carlos Chavez wrote

Re: [asterisk-users] Set external CID but retain internal extension in CDR...

On 2/22/18 3:54 PM, Carlos Chavez wrote: On 2/22/18 3:46 PM, Antony Stone wrote: On Thursday 22 February 2018 at 21:41:41, Carlos Chavez wrote: On 2/22/18 1:07 PM, Antony Stone wrote: On Thursday 22 February 2018 at 19:10:51, Carlos Chavez wrote:    Usually phone companies set

Re: [asterisk-users] Set external CID but retain internal extension in CDR...

On 2/22/18 3:46 PM, Antony Stone wrote: On Thursday 22 February 2018 at 21:41:41, Carlos Chavez wrote: On 2/22/18 1:07 PM, Antony Stone wrote: On Thursday 22 February 2018 at 19:10:51, Carlos Chavez wrote: Usually phone companies set the outgoing CallerID for you but recently we got

Re: [asterisk-users] Set external CID but retain internal extension in CDR...

On 2/22/18 2:05 PM, Antony Stone wrote: On Thursday 22 February 2018 at 20:07:47, Antony Stone wrote: On Thursday 22 February 2018 at 19:10:51, Carlos Chavez wrote: Usually phone companies set the outgoing CallerID for you but recently we got control over that and are now setting

Re: [asterisk-users] Set external CID but retain internal extension in CDR...

On 2/22/18 1:07 PM, Antony Stone wrote: On Thursday 22 February 2018 at 19:10:51, Carlos Chavez wrote: Usually phone companies set the outgoing CallerID for you but recently we got control over that and are now setting the outgoing Calleir ID ourselves. My problem now is that the CDR

[asterisk-users] Set external CID but retain internal extension in CDR...

    Usually phone companies set the outgoing CallerID for you but recently we got control over that and are now setting the outgoing Calleir ID ourselves.  My problem now is that the CDR will put the outgoing CID in the CDR instead of the extension that dialed and that causes problems for

[asterisk-users] Queue playing periodic_announce to agent when they answer

    I have a very strange problem with my queues today.  When the agent answers a call they get the periodic_announce sound played to them.  I have a periodic_announce set to 60 seconds and the caller does hear it if their call is not answered.  Why would it play it to the agent?  At this

Re: [asterisk-users] Duplicate CDR's in Mysql

On 1/14/18 4:22 PM, Mike Diehl wrote: Hi all, I have a problem I've not seen before. My Asterisk server stores CDR's via mysql, and I'm getting duplicate records. For example: mysql> select uniqueid,count(*) from cdr group by uniqueid having count(*)>1;

Re: [asterisk-users] Mixmonitor with b option

On 1/8/18 9:38 AM, Bertrand LUPART - Linkeo.com wrote: Hello Carlos, We have a server that records all calls so we set Mixmonitor with the b option to only record calls that are actually bridged. I notice that we have lost of 44 byte files in /var/spool/asterisk/monitor which

[asterisk-users] Mixmonitor with b option

We have a server that records all calls so we set Mixmonitor with the b option to only record calls that are actually bridged. I notice that we have lost of 44 byte files in /var/spool/asterisk/monitor which correspond to calls that were not answered. If a call is not answered I assume it

Re: [asterisk-users] PJSIP Trunk 401 Unauthorized (Alestra Mexico)

On 12/2/17 4:40 PM, Joshua Colp wrote: On Sat, Dec 2, 2017, at 06:33 PM, Carlos Chavez wrote: I am having a really bad day trying to get incoming calls to work on Asterisk 13 with PJSIP. We just migrated from Asterisk 1.8 where everything was working but there seems that something got

[asterisk-users] PJSIP Trunk 401 Unauthorized (Alestra Mexico)

    I am having a really bad day trying to get incoming calls to work on Asterisk 13 with PJSIP.  We just migrated from Asterisk 1.8 where everything was working but there seems that something got lost in translation.  No matter what I try I always get a 401 Unauthorized message when receiving

Re: [asterisk-users] Confbridge SFU for Asterisk 15

On 11/15/17 11:36 AM, Joshua Colp wrote: On Wed, Nov 15, 2017, at 01:30 PM, Carlos Chavez wrote: Here is more information from the browser about the session: https://pbxoficina.telecomabmex.com/nextcloud/index.php/s/af36iLlljtbYkbF On Asterisk I have icesupport=true in rtp.conf

Re: [asterisk-users] Confbridge SFU for Asterisk 15

On 11/15/17 11:10 AM, Joshua Colp wrote: On Wed, Nov 15, 2017, at 01:05 PM, Carlos Chavez wrote: On 11/14/17 5:23 PM, Joshua Colp wrote: On Tue, Nov 14, 2017, at 07:19 PM, Carlos Chavez wrote: Trace with 3 clients. We can hear each other but no video. https://pbxoficina.telecomabmex.com

Re: [asterisk-users] Confbridge SFU for Asterisk 15

On 11/14/17 5:23 PM, Joshua Colp wrote: On Tue, Nov 14, 2017, at 07:19 PM, Carlos Chavez wrote: Trace with 3 clients. We can hear each other but no video. https://pbxoficina.telecomabmex.com/nextcloud/index.php/s/X0PQ5FrYeqCwwkz Do you see anything in the Javascript console of the browser

Re: [asterisk-users] Confbridge SFU for Asterisk 15

Trace with 3 clients. We can hear each other but no video. https://pbxoficina.telecomabmex.com/nextcloud/index.php/s/X0PQ5FrYeqCwwkz On 11/14/17 5:06 PM, Joshua Colp wrote: On Tue, Nov 14, 2017, at 07:03 PM, Carlos Chavez wrote: On 11/14/17 4:27 PM, Joshua Colp wrote: On Tue, Nov 14, 2017

Re: [asterisk-users] Confbridge SFU for Asterisk 15

On 11/14/17 4:27 PM, Joshua Colp wrote: On Tue, Nov 14, 2017, at 06:25 PM, Carlos Chavez wrote: On 11/14/17 3:55 PM, Joshua Colp wrote: On Tue, Nov 14, 2017, at 05:47 PM, Carlos Chavez wrote: I followed the blog post and I can get video from the conference if I configure the bridge

Re: [asterisk-users] Confbridge SFU for Asterisk 15

On 11/14/17 3:55 PM, Joshua Colp wrote: On Tue, Nov 14, 2017, at 05:47 PM, Carlos Chavez wrote: I followed the blog post and I can get video from the conference if I configure the bridge as follow_talker so I know everything is working on the pjsip side. The only problem

Re: [asterisk-users] Confbridge SFU for Asterisk 15

On 11/14/17 3:38 PM, Joshua Colp wrote: On Tue, Nov 14, 2017, at 05:23 PM, Carlos Chavez wrote: I am trying to get the "Mega Phone" demo working on my office PBX but there seems to be a problem when trying to set the default bridge to sfu mode. I have the following con

[asterisk-users] Confbridge SFU for Asterisk 15

I am trying to get the "Mega Phone" demo working on my office PBX but there seems to be a problem when trying to set the default bridge to sfu mode. I have the following configuration in confbridge.conf in the default_bridge section: video_mode = sfu but when I do a "confbridge show

[asterisk-users] Asterisk 13.8 compile error

I just tried to compile the latest Asterisk 13.8.0 and it stopped with several errors on pjsip. So FYI if you run the install_prereq script and then use ./configure --with-pjproject-bundled you will have the same problem because the prereq script installs an older version of pjproject.

Re: [asterisk-users] PJSIP trunk to Telynx

On 10/20/2017 8:46 PM, Joshua Colp wrote: On Fri, Oct 20, 2017, at 10:17 PM, Carlos Chavez wrote: Has anyone used Telynx as a SIP trunk provider?  It works with chan_sip but it I seem to be having problems trying to set up a PJSIP trunk.  I always get a 401 Unauthorized when they send me

[asterisk-users] PJSIP trunk to Telynx

Has anyone used Telynx as a SIP trunk provider?  It works with chan_sip but it I seem to be having problems trying to set up a PJSIP trunk.  I always get a 401 Unauthorized when they send me a call.  I know my username and password are correct since I can register and PJSIP uses the same

Re: [asterisk-users] speech-recog.agi

On 10/19/17 3:53 PM, Jonathan H wrote: That's because it uses a deprecated API and endpoint. However, funny you should ask this, because I've just finished updating my Google TTS routine to take advantage of the new streamlined API. If you can wait a couple of days, I've stick it up on the

[asterisk-users] speech-recog.agi

I want to try using google for speech recognition in Asterisk and I found a ready made AGI: http://zaf.github.io/asterisk-speech-recog/ I have followed all the steps listed in the web site but I keep getting this error: AGI Tx >> 200 result=99981 (timeout) endpos=22720 AGI Rx <<

[asterisk-users] Cepstral, Swift and Asterisk 13

Anyone here have a working app_swift with Asterisk 13? I purchased my licenses and followed their install procedure but I do not get any audio when I dial a test. Stranger still is that I can get audio on a softphone (Bria) but nowhere else. I have tried several desk phones and

[asterisk-users] Confbridge GUI?

I have a very old server that is used only for conferences on Meetme. To manage the conference rooms we use Web Meetme. Now it is time to upgrade everything but since Meetme is no longer available I need to find a replacement GUI to manage the conference rooms. Anyone know a solution

Re: [asterisk-users] Asterisk 13 on old VMware ESXI 4

On 2017-08-02 07:08, Nathan Anderson wrote: Richard Kenner wrote: But the question here was *Asterisk*, not kernels. User-level code has *way* fewer dependencies. *Precisely*. Unless we're talking DAHDI here (which we're not), Linux & ESXi are red herrings. Carlos Chavez wrote:

Re: [asterisk-users] Asterisk 13 on old VMware ESXI 4

On 2017-08-01 15:48, Doug Lytle wrote: I am having a very tough time trying to replace an Elastix 2.X install running as a virtual machine on ESXI 4 Licensed or free ESXI? I want to say your version is too old. I'm currently running ESXI 6.0 update 3 at home and Asterisk in a VM under debian

[asterisk-users] Asterisk 13 on old VMware ESXI 4

I am having a very tough time trying to replace an Elastix 2.X install running as a virtual machine on ESXI 4. I tried using the Freepbx 14 ISO that installs CentOS 6 along with Asterisk 13.16 but I keep getting random segfaults: [175711.476685] asterisk[2942]: segfault at 188 ip

[asterisk-users] Asterisk install_prereq

Is there a reason for Asterisk 13.17.0 to download and install pjproject-devel-2.3-6.el7.x86_64.rpm when you run the install_prereq script? Since most people will compile asterisk using the bundled version of pjproject this may cause confusion. And it is also an older version than the

Re: [asterisk-users] Asterisk 13.16.0 segfault

lt;mailto:mhter...@jabber.mundoopensource.com.br> https://www.mundoopensource.com.br https://twitter.com/mhterres https://linkedin.com/in/marceloterres On 19 July 2017 at 17:03, Carlos Chavez <cur...@telecomab.mx <mailto:cur...@telecomab.mx>> wrote: On 7/19/17

Re: [asterisk-users] Asterisk 13.16.0 segfault

On 7/19/17 2:37 AM, Marcelo Terres wrote: This is the pjsip library. Is it possible to you to update pjsip for the latest version to test if it solves the problem? On 18 Jul 2017 3:52 pm, "Carlos Chavez" <cur...@telecomab.mx <mailto:cur...@telecomab.mx>> wrote:

[asterisk-users] Asterisk 13.16.0 segfault

I am getting frequent segfaults on a new Asterisk installation. So far the only message I see is: Jul 18 09:02:42 pbxbogota kernel: asterisk[26799]: segfault at 188 ip 7fb2d535723f sp 7fb25a11b5c0 error 4 in libasteriskpj.so.2[7fb2d52e5000+18] Jul 18 09:17:00 pbxbogota kernel:

Re: [asterisk-users] IAX2 getting stuck

On 4/20/17 2:37 PM, Kseniya Blashchuk wrote: If SIP goes to the same provider then yes. Still I would check a packet capture for better understanding. BTW, did you try iax debug? чт, 20 апр. 2017 г. в 19:46, Carlos Chavez <cur...@telecomab.mx <mailto:cur...@telecomab.mx>>:

Re: [asterisk-users] IAX2 getting stuck

that disappeared when the adapter was unplugged... On Thu, Apr 20, 2017, 12:41 AM Antony Stone <antony.st...@asterisk.open.source.it <mailto:antony.st...@asterisk.open.source.it>> wrote: On Wednesday 19 April 2017 at 23:35:24, Carlos Chavez wrote: > On 4/19/17 4:23 PM, Ant

Re: [asterisk-users] IAX2 getting stuck

On 4/19/17 4:23 PM, Antony Stone wrote: On Wednesday 19 April 2017 at 23:14:46, Carlos Chavez wrote: On 4/19/17 4:09 PM, Antony Stone wrote: On Wednesday 19 April 2017 at 22:54:51, Carlos Chavez wrote: I have a server that had been operating for a few years now with IAX2 trunks

Re: [asterisk-users] IAX2 getting stuck

On 4/19/17 4:09 PM, Antony Stone wrote: On Wednesday 19 April 2017 at 22:54:51, Carlos Chavez wrote: I have a server that had been operating for a few years now with IAX2 trunks to several other servers. Since yesterday all IAX2 trunks now say UNREACHABLE. ...snip... So far

[asterisk-users] IAX2 getting stuck

I have a server that had been operating for a few years now with IAX2 trunks to several other servers. Since yesterday all IAX2 trunks now say UNREACHABLE. No configuration changes have been made and no upgrades have been done. The server is running 11.16.0 (yes, we are planning

Re: [asterisk-users] Triggering an AGI script when a queued call is answered

On 11/24/16 11:20 AM, A J Stiles wrote: Many years ago, I used to have an AGI script that fired on an incoming call, did some database lookups and ended up raising a notification on the screen of the person whose phone was ringing, with the details looked up from the incoming caller ID. All

[asterisk-users] sorcery.conf mappings

Is there some documentation for all the available sorcery.conf mappings for realtime? Asterisk already includes some tables in the database that are not enabled by default on the sorcery.conf like transports and outbound registrations. There are no examples in the file on how to enable

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

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 46:24:59 Sotelo Sotelo IAX2/to-CD-20713 I have tried "hangup request

Re: [asterisk-users] Conference Call like conference done by mobile!

On 10/5/16 7:04 AM, Mandar Khire wrote: hi, I trying to solve one scenario:- As I can make call from mobile phone to my friend1. As he accept it, I put him on hold, & dial friend2. As he also accept it, I put him on hold & follow same procedure till friend6. The I click on 'Merge call' & I

Re: [asterisk-users] Mysql PJSIP realtime > 13.10?

On 9/12/16 4:21 PM, George Joseph wrote: On Mon, Sep 12, 2016 at 3:01 PM, Carlos Chavez <cur...@telecomabmex.com <mailto:cur...@telecomabmex.com>> wrote: On 9/12/16 3:39 PM, George Joseph wrote: On Mon, Sep 12, 2016 at 2:31 PM, George Joseph <gjos...@digium.c

Re: [asterisk-users] Mysql PJSIP realtime > 13.10?

On 9/12/16 3:39 PM, George Joseph wrote: On Mon, Sep 12, 2016 at 2:31 PM, George Joseph <gjos...@digium.com <mailto:gjos...@digium.com>> wrote: On Mon, Sep 12, 2016 at 2:14 PM, Carlos Chavez <cur...@telecomabmex.com <mailto:cur...@telecomabmex.com>> wrote:

Re: [asterisk-users] Mysql PJSIP realtime > 13.10?

On 9/12/16 3:21 PM, Annus Fictus wrote: Hello, is there any reason you don't use ODBC with MySQL? Regards El 12/09/2016 a las 15:14, Carlos Chavez escribió: Has anyone successfully used Mysql realtime PJSIP with Asterisk 13.11? I have tried 13.11, 13.11.1 and 13.11.2 but I always get

[asterisk-users] Mysql PJSIP realtime > 13.10?

Has anyone successfully used Mysql realtime PJSIP with Asterisk 13.11? I have tried 13.11, 13.11.1 and 13.11.2 but I always get the following error now: Sep 12 14:42:35] WARNING[24498]: res_config_mysql.c:1162 require_mysql: Realtime table general@ps_contacts: column 'qualify_timeout'

[asterisk-users] How to get a list of DAHDI channels

Anyone know an efficient way to get a list of the DAHDI channels? Is there an AMI or ARI variable to get a list of all the channels? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez +52 (55)9116-91161 --

Re: [asterisk-users] Asterisk 13.11 realtime problem registering phones

On 2016-09-04 10:11, Max Grobecker wrote: Hi, Am 02.09.2016 um 22:48 schrieb Carlos Chavez: I upgraded my office installation from 13.10 to 13.11 yesterday and now I am having problems registering phones. Here is what I get on the CLI: [Sep 2 15:38:46] WARNING[2098

[asterisk-users] Asterisk 13.11 realtime problem registering phones

I upgraded my office installation from 13.10 to 13.11 yesterday and now I am having problems registering phones. Here is what I get on the CLI: [Sep 2 15:38:46] WARNING[2098]: res_config_mysql.c:1162 require_mysql: Realtime table general@ps_contacts: column 'qualify_timeout' cannot be

[asterisk-users] PJSIP hints unreliable...

I find that using hints with PJSIP on Asterisk 13 is very unreliable compared to regular SIP. I see many phones as unavailable when they are in fact available. Usually hints will work fine for a while after a phone registers but after a while it will remain at unavailable while it is

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