[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 tha

[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 softphone

[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 << VERB

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 rep

[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 inform

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 a

[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. Ma

[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 profil

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 config

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 is that

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

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 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

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 11/15/17 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 and

[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] 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] 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] 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 correspond

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; +--+--

[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 point

[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 repor

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

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 the

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 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 the

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

[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 diale

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 b

[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 DID

[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 7f1568789820

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 pro

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 Deskpho

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

[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 '

[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 start

[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 -- _

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 th

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] 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 send

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, "s"

[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] 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 every

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. https://wiki.asterisk.org/wiki/display/AST/Extension+State+an

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 "

[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 "chann

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

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

[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. 

[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 up

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

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 Asteri

[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 anyth

[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 use

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 out

[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 14410400-5e04-4358-af

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] 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

[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 be

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

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 interac

[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/ d

[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 havi

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] 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 rese

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 blockin

[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 not

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

wer versions 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 &q

[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 tra

[asterisk-users] Strange RTP problem...

I have a new installation where an Asterisk server is connected to an Avaya PBX via a PRI E1. We are having a problem that I attribute to their firewall but I just want to make sure. When we make a call from the Avaya to a SIP extension there is only sound on the receiving end. F

[asterisk-users] NAT issue with Fortinet Firewall

I have a customer with a Fortinet Firewall that is having stability issues with Asterisk and SIP endpoints (PAP2T) outside his network. The first issue I see is that Asterisk sees all phones as the IP address of the Fortinet. Since the parameter "localnet" defines the local netw

[asterisk-users] G729 license count...

I need a refresher course on how many licenses I need to buy. I have an Asterisk server that receives calls by SIP (G729) and then sends them to the PSTN via 32 Zap interfaces on an Astribank. I cannot remember if the license is per channel or per call so I do not know if I need 32 or 64

[asterisk-users] One way audio...

I have a big headache. I have an Asterisk server connected to an Avaya PBX. Everything is working between those two. The problem is that I have 45 PAP2T adapters and 45 SPA3102 adapters that connect via the Internet to the Asterisk server through a Fortinet firewall. When calling from a

[asterisk-users] One way audio...

I am still having a very frustrating problem win an Avaya-Asterisk system. I have written about this before but I am expanding the description of the problem just in case someone can give me some insight. This installation is an Asterisk 1.4.19.1 server connected to an Avaya PBX u

Re: [asterisk-users] One way audio...

ternip and localnet parameters correctly. > 2) Also in sip.conf, try the following on the PAP2's sections: > > disallow=all > allow=alaw:10 > > In case that fails, try also > > disallow=all > allow=alaw:20 > > > > Att > Vinícius Fontes > Desenvo

Re: [asterisk-users] One way audio...

at fails, try also > > disallow=all > allow=alaw:20 > > > > Att > Vinícius Fontes > Desenvolvimento > Canall Tecnologia em Comunicações Ltda. > > - "Carlos Chavez" <[EMAIL PROTECTED]> escreveu: > > > I am still having a very frustrating p

[asterisk-users] externip not working...

I have an Asterisk 1.4.19.1 server that is behind a Fortinet firewall. Localnet is 192.168.2.0/255.255.255.0 and all external sip devices look as if they are on the same local network because the Fortinet rewrites the incoming IP as its own address. The problem I have is that when

[asterisk-users] More one way audio...

I am a bit desperate trying to solve this problem. Sorry if I am abusing the list a bit with the same king of question. The problem I am having is very specific which is why it is very difficult to diagnose and fix. Basically an Asterisk server is connected via E1 PRI to an Avaya

[asterisk-users] Busy out a zap channel?

Is there a way to busy out a Zap channel? I have a customer who is having problems with a line connected to a TDM800 card and we would like to busy out that line. Since that line is the head of the hunt group I cannot simply disable that channel, I need to busy the line so calls will come

Re: [asterisk-users] Busy out a zap channel?

er() > exten => s,n,Busy() > > > Att > Vinícius Fontes > Desenvolvimento > Canall Tecnologia em Comunicações Ltda. > > - "Carlos Chavez" <[EMAIL PROTECTED]> escreveu: > > > Is there a way to busy out a Zap channel? I have a customer who is &

Re: [asterisk-users] Busy out a zap channel?

Thank you. Unfortunately the phone Company in Mexico is not very helpful when it comes to those services. On Tue, 2008-05-20 at 16:48 -0500, Tilghman Lesher wrote: > On Tuesday 20 May 2008 16:08:19 Carlos Chavez wrote: > > The problem is that I do not have physical acce

Re: [asterisk-users] addons-1.6 not seeing installed MySQL packages

Don't know about Debian but in Fedora or CentOS you need to install mysql-devel to compile Mysql support in Asterisk-Addons On Wed, 2008-05-21 at 14:31 -0500, JR Richardson wrote: > Hi All, > > I'm poking around with 1.6, tried to compile the addon package, but it > doesn't see mysql_conf

Re: [asterisk-users] Similar extension numbers for multiple users

As long as each tenant has its own context you can use the same numbering plan. The only thing you need to keep unique are the names for the SIP devices. If you want your tenants to be able to call each other then you would need to set up a special prefix for each tenant. On Thu, 2008-06

[asterisk-users] TE110P with 40,000 IRQ missess

? -- Carlos Chavez Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options

[asterisk-users] Zaptel version on Asterisk website...

Since Zaptel 1.4.11 has been released, why is the link on the Asterisk website pointing to 1.4.10.1? Is there a problem with the newest version or just someone forgot to update the link? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52

Re: [asterisk-users] Red Alarm TE420 with E1s - R2

Your problem seems to be that the card is in T1 mode and you need it to be in E1 mode. Check the jumpers on the card and change them to the E1 position. Or you can send the module a parameter to put the card in E1 mode. On Mon, 2007-12-03 at 13:14 -0200, Roger C. Beraldi Martins wrote: >

Re: [asterisk-users] Setting custom field in CDR

On Thu, 2007-12-06 at 10:37 -0500, Mike wrote: > Hi, > > The Asterisk Wiki (page: http://www.voip-info.org/wiki/view/Asterisk > +func+cdr) mentions I can set any custom CDR field I want. Here is > the example it gives: > > ; Update our accountcode field and then save some random music facts t

[asterisk-users] Limit participants in Meetme...

Is there an easy way to limit the number of participants on a Meetme room? Lets say we only want 10 people to be able to enter a particular meetme conference, how can I prevent number 11 from entering this conference? We will not have a pin to enter. -- Telecomunicaciones Abiertas de Mé

Re: [asterisk-users] Asterisk and NAT

On Tue, 2007-12-11 at 00:14 -0800, bilal ghayyad wrote: > Hi All; > > My Asterisk has a public IP address, how can we let > two IP Phones in different site and both are behind > NAT (each one has a private IP address) to call each > other? > > In other words, > Assuming Asterisk IP Address is 1

[asterisk-users] Calling Party Category Field

ountcode=Alestra group=1 switchtype=euroisdn callerid=asreceived signalling=pri_cpe pridialplan=unknown faxdetect=both channel=1-15,17-31 Any ideas on how to solve this problem? -- Carlos Chavez Director de Tecnología Telecomunicaciones Abiertas de México S.A. de

[asterisk-users] Calling Party Category Field

Alestra group=1 switchtype=euroisdn callerid=asreceived signalling=pri_cpe pridialplan=unknown faxdetect=both channel=1-15,17-31 Any ideas on how to solve this problem? -- Carlos Chavez Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 s

Re: [asterisk-users] Queue calls drop to voicemail intermittantly

On Mon, 2007-12-17 at 11:45 -0700, Robert Norton - SophMedia LLC wrote: > Are the agents “ignoring” the calls while their ringing? > > > > -- > Robert Norton > SophMedia LLC Operations Manager > Cell: 480-234-4312 Office: 480-626-5449 (x300) > P.O. Box 7755 Tempe, AZ 85281 > http://www.XStrea

Re: [asterisk-users] Lamps on Snom phones

On Thu, 2008-01-03 at 18:00 +, Russell Brown wrote: > Quoth "Phil Knighton" <[EMAIL PROTECTED]> > > > >I've incorporated the kind responses from other list members, such as > >setting call limits but to no avail! I've checked the function key > >settings on the Snom, and adjusted it to match

Re: [asterisk-users] MFC/R2

On Mon, 2008-01-28 at 17:03 -0700, James Finstrom wrote: > -BEGIN PGP SIGNED MESSAGE- > Hash: SHA1 > > Followed the instructions at > http://www.voip-info.org/wiki/view/Asterisk+MFC+R2 > > I dead end at patching the channels Makefile. There have been some > changes since these instructio

[asterisk-users] Asterisk 1.4.17 and Teliax DTMF

I am having a problem with DTMF when sending calls through Teliax (SIP). In the peer for teliax I defined dtmfmode=rfc2833 and for the most part it is working. The problem always happens when a user is trying to call a conference system. They simply cannot get into the conference because

[asterisk-users] R2 with Alestra in Mexico...

I am trying to set up Astunicall 1.4.16 with a link from Alestra in Mexico City. I have done everything I usually do for other links in Mexico but this one simply will not send or receive calls. I just get Protocol error. Anyone has any experience with R2 and Alestra? -- Telec

Re: [asterisk-users] R2 with Alestra in Mexico...

On Wed, 2008-02-06 at 08:17 -0600, Moises Silva wrote: > Carlos, I have some spare time today in case you want me to check it. > > Is this your first time with Alestra? > Thank you for the offer. Yes this is the first time I use Alestra for R2. I have another customer that uses

Re: [asterisk-users] Asterisk and fax

I would recommend you use Iaxmodem / Hylafax / Avantfax for your needs. We use this with several customers and it works very well. This way you do not have to patch Asterisk with spanDSP. You can set up as many virtual fax machines as your machine will handle. On Wed, 2008-02-13 at 18:

[asterisk-users] AMD on a SIP trunk...

We have an Asterisk server with a small outgoing call center. We use AMD and it usually works very well on Zap channels (E1 PRI). We added a couple of SIP trunks to reduce long distance costs but now AMD gets stuck when the call goes out through the SIP channels. Here is an example call

Re: [asterisk-users] About faxes recived through a PRI and passed to a fax machine connected to a FXS port

You do not have to do anything else. When Asterisk detects a fax tone it will disable echo cancellation on those channels so the fax can go through. Just make sure that the Astribank is the sync source for timing and you should be able to send and receive faxes. In your dialplan

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