Re: [asterisk-users] Alternative to Local channel

You can't set the variable in globals? I don't know if functions work in globals, but it is worth a try. [globals] LSESSION=${STRFTIME(${EPOCH},,%Y-%m-%d %H:%M:%S)} On 8/16/23 20:39, Federico wrote: I used to use the local channel to create a global variable (dialplan) [default] exten =>

Re: [asterisk-users] Setting codec on originating (calling) channel with chan_pjsip (SIP_CODEC_INBOUND equivalent)

I suspect most people simply don't care. Transcoding between ulaw and g722 is not CPU intensive and Direct Media doesn't work when NAT is involved (which would the case for most people). On 7/5/23 17:22, Michael Ulitskiy wrote: Well, I'm trying to migrate to chan_pjsip so that I don't have

Re: [asterisk-users] AGI script commands

You have to read stdin to accept the data Asterisk sends when the AGI starts before you can send any AGI commands to Asterisk. Also, "agi set debug on". On 6/30/23 21:52, TTT wrote: I have an AGI script written in PHP that worked great with Asterisk 13. I’m porting it to an Asterisk 20

Re: [asterisk-users] Get channel variables via ARI/AMI

If you want to do more than just get headers from the INVITE, then you'll need to use an actual SIP Proxy like Kamailio. On 6/26/23 15:03, TTT wrote: It looks like if I call Getvar and pass PJSIP_HEADERS() I can get the entire SIP header for a channel.  I also read (on stackoverflow) that

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

type=endpoint rewrite_contact=yes force_rport=yes rtp_symmetric=yes On 6/21/23 14:36, TTT wrote: I've split this thread off from another (PJSIP authentication) because I think the root cause is something different.I think the problem is the following FROM line in my SIP INVITE

Re: [asterisk-users] Confbridge for 80 devices

https://wiki.asterisk.org/wiki/display/AST/Timing+Interfaces On 10/20/22 17:35, Jerry Geis wrote: [modules] autoload = yes noload = res_timing_pthread noload = res_timing_timerfd SO I "dont" want to load res_timing_anything ??? I have preload on res_timing_dahdi, then res_timing_pthread and

Re: [asterisk-users] Muliticast not connecting

On 10/13/22 13:25, Joshua C. Colp wrote: On Thu, Oct 13, 2022 at 2:16 PM Jerry Geis > wrote: I have a simple dialplan with asterisk 18.14.0 exten => 141,1,Answer exten => 141,n,Noop(MC) exten => 141,n,Playback(beep) exten =>

Re: [asterisk-users] externnotify script not running

If you are using systemd /tmp might be a private /tmp On 3/16/22 16:10, Mike Diehl wrote: Hi all, I'm trying to build a custom voicemail delivery system using externnotify in voicemail.conf. But, the configured script doesn't seem to run. I have:

Re: [asterisk-users] Get context with hangup handler

This might not help, but you don't have to use different contexts when using GoSub, here is an example: https://pastebin.com/ftwWwpKt On 1/5/22 22:22, Dovid Bender wrote: Steve, I thought of this but that would mean I would need to add this to the beginning of every context which I can do,

Re: [asterisk-users] Exec two commands with ExecIf

The dialplan is a very complicated config file. It is often repetitive and ugly. Don't expect it to work like an actual programming language. If you want that, use an AGI. In this case, just call it an "Asterisk-ism" and move on. You'll find plenty more of them. On 12/23/21 01:37, Dovid

Re: [asterisk-users] Dial() after the h extension has been invoked?

Create a spool file from the 'h' extension to generate the call. On 11/12/21 11:56, Antony Stone wrote: Hi. I have a setup which comprises some "front-end" Asterisk servers which have SIP trunks to external providers, and very simple dial plans, and some "back- end" servers which only talk to

Re: [asterisk-users] Between a dumb client and a capable server...

On 8/20/21 4:24 PM, Antony Stone wrote: On Friday 20 August 2021 at 19:06:09, George Joseph wrote: On Fri, Aug 20, 2021 at 8:33 AM Antony Stone wrote: So, if I have Asterisk registered as a SIP client to some remote server, how can I get Asterisk to tell that remote server to put the call

Re: [asterisk-users] Failed to authenticate

You could switch to PJSIP and avoid most of this silliness. I love Asterisk, but the peer/user/friend model in chan_sip is simply terrible. PJSIP is different so there is a learning curve, of course. On 8/9/21 11:05 AM, Jerry Geis wrote: On Mon, Aug 9, 2021 at 8:32 AM Jerry Geis

Re: [asterisk-users] SIP Source Port

Kamailio is useful when you want to do weird, non-standard, or unusual stuff with SIP. You could send your outgoing connections to Kamailio, which could then send the connection out with the required source port. Have you considered using a not stupid provider? On 7/10/21 3:44 PM, Joshua C.

Re: [asterisk-users] Loading Json values into asterisk as variable values

You can set variables in pjsip.conf on specific endpoints. See pjsip.conf.sample On 2/26/21 9:56 AM, Dovid Bender wrote: Steve, What language are your AGI's written in? I have been using PHP for a long time and every time it's launched there seems to be a run on the CPU. I wonder if I

Re: [asterisk-users] upgrade asterisk 11 to 13 or 11-16

"asterisk -vvr" what it is doing? On 12/07/2020 11:50 AM, Eric Wieling wrote: Read UPGRADE.TXT in v13 and v16.  Then read it again. I upgraded from Asterisk v11 to Asterisk v13.  Once all issues were resolved, then I switched to PJSIP.   Once all the issues with PJSIP were resolved, then

Re: [asterisk-users] upgrade asterisk 11 to 13 or 11-16

Read UPGRADE.TXT in v13 and v16. Then read it again. I upgraded from Asterisk v11 to Asterisk v13. Once all issues were resolved, then I switched to PJSIP. Once all the issues with PJSIP were resolved, then I upgraded from v13 to Asterisk v16. This was done over the course of about a

Re: [asterisk-users] Digium TE134 compatibility issues with new Dell server - Zero interrupts

Make sure selinux is set to permissive or disabled. On 10/22/20 11:44 AM, Richard Reina wrote: Dell T40 -- http://help.nyigc.net/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new

Re: [asterisk-users] Stir Shaken

I used their scam checking service. Below is part of the dialplan I used. I don't know how their STIR/SHAKEN service works the same. same = n,GosubIf($[${LEN(${CALLERID(num)})} == 11]?scam-check,${EXTEN},1) same = n,Goto(from-pstn,${EXTEN},1) [scam-check] exten = _XX.,1,Noop same =

Re: [asterisk-users] Codec question

turn off g726. On 6/17/20 4:34 PM, Jerry Geis wrote: Ok - updating the firmware on teh device - factory reset, re-config. Capabilities: us - (g726|slin16|ulaw|alaw|gsm), peer - audio=(ulaw|alaw|g726|slin16)/video=(nothing)/text=(nothing), combined - (g726|slin16|ulaw|alaw) Looking much

Re: [asterisk-users] Stir-Shaken for asterisk

Transnexus has some STIR/SHAKEN related features. https://transnexus.com/clearip/ We are evaluating them for robocall blocking. On 5/28/20 12:10 AM, Jeff LaCoursiere wrote: A few weeks... like in a year and a few weeks: https://transnexus.com/blog/2020/fcc-mandates-stir-shaken/ Some

Re: [asterisk-users] Troubleshooting load issues

Try setting transcode_via_sln=no in /etc/asterisk/asterisk.conf and restart Asterisk. A reload will NOT apply the new value. Setting it to no seems to smooth out CPU usage on one of my servers. On 4/22/20 2:01 PM, Dovid Bender wrote: Hi, I have an Asterisk box which has an IVR that plays

Re: [asterisk-users] DAHDI not loading

If the kernel changes, then DAHDI needs to be recompiled. It sounds like the kernel did not change, but you might want to check. On 3/16/20 12:58 PM, Jerry Geis wrote: Just a guess, Recompile Dadhi. I'd rather not have to do that step. I "desire" to make the image and copy to the

Re: [asterisk-users] USB dahdi fxo ?

People make it overly complicated. Things which provide dialtone are plugged into FXO ports. Phone lines PBX analog extensions Things which expect to be provided with dialtone are plugged into FXS ports. Analog Phones Fax Machines FXO ports can handle the 90 volt ring singnal FXS

Re: [asterisk-users] clarification on gosub, macros and AEL

For all new dialplan, I recommend using Gosub. From https://www.voip-info.org/asterisk-ael2/ : This will complement the fact that Macros will be implemented with Gosub() calls instead of Macro() calls from now on, to avoid restricted memory issues. [I think this started in Asterisk 1.4)

Re: [asterisk-users] Increasing variables - Changes v13 vs v16

Verify ${myCpt} is not empty. On 10/1/19 10:24 AM, Administrator TOOTAI wrote: Hi list, on asterisk 13 I use same => n,Set(__myCpt=$[${myCpt} + 1]) which is working well. On an Asterisk 16 I get, for this same command [2019-10-01 16:15:01] WARNING[28197][C-0008]: ast_expr2.fl:470

Re: [asterisk-users] Lightweight ODBC DB

It is not tough to run multiple instances of MySQL using different data directories and different ports/sockets.Google for mysqld_multi. I have an MySQL instance which holds call routing information and is replicated to the Asterisk servers. I have a second MySQL instance on the same

Re: [asterisk-users] Forking AGI or GoSub

In PHP something like: $pid = pcntl_fork(); if ($pid != 0) { // we are the parent // do parent stuff exit; } // we are the child, detatch from terminal $sid = posix_setsid(); if ($sid < 0) { die; } // do child stuff On 04/19/2019 02:00 PM, Mark Wiater wrote: On 4/19/2019 1:49

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

Think of "s" as meaning "stupid" because calls from devices too stupid to send the dialed number are routed to the "s" extension. Any incoming calls which includes the dialed number would NOT be sent to extension "s", those calls will match whatever the dialed number is. On 03/28/2019 08:32

Re: [asterisk-users] Paging systems?

These seem to work well: http://www.vikingelectronics.com/product_docs/product_manuals/VE__PA-15__15_Watt_Paging_Amplifier_with_Background_Music_and_Loud_Ringing_486.pdf On 03/21/2019 02:53 PM, Michael Munger wrote: Does anyone have an (overhead) paging system that they like that works with

Re: [asterisk-users] branching in extensions.conf?

If you want your dialplan code to look pretty, use AEL. On 02/20/2019 11:41 AM, Brian J. Murrell wrote: Is there any less cumbersome way of doing conditionalized/branching in extensions.conf other than something like: exten => s,n,GotoIf($["${SIP}" = "PJSIP" ]?pjsip) exten =>

Re: [asterisk-users] PJSIP DNS ISSUE

Make sure the IP of every interface address is listed in /etc/hosts Use dnsmgr Install local BIND, which you already did. On 02/20/2019 11:29 AM, John T. Bittner wrote: Anyone know how to disable DNS in asterisk so PJSIP still works when the internet goes down. I tried a few things but

Re: [asterisk-users] trouble removing + sign

The format of Set is Set(MYVAR=myval) not Set(MYVAR = myval) On 02/13/2019 06:12 PM, sean darcy wrote: I'm using BLACKLIST() to check numbers, which does not like leading + signs. I want to test if there is a plus sign, and then remove it. I tried:  ;  strip leading plus sign   same => n,

Re: [asterisk-users] Freepbx / Asterisk PJsip multipe devices

That question was answered long ago.. https://wiki.asterisk.org/wiki/display/AST/Dialing+PJSIP+Channels On 02/06/2019 09:16 AM, basti wrote: In other words. I there a way that both phones are ring with only one extension? On 06.02.19 15:05, basti wrote: both phones are in the same net. when

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

From https://wiki.asterisk.org/wiki/display/AST/Timing+Interfaces: res_timing_dahdi uses timing mechanisms provided by DAHDI. This method of timing was previously the only means by which Asterisk could receive timing. It has the benefit of being efficient, and if a system is already going to

Re: [asterisk-users] Is there any way to pass caller id to cell phone?

block it. It is worth testing if they will allow any outbound CLID or need it to match the account. The problem will get worse when SHAKEN'STIR comes into effect and they need to certify that the call came from your office. The reason they would block it is to prevent both spam calls and fraud. Eric

Re: [asterisk-users] How to defer SDP in ACK for unit testing purposes

Have you looked at https://sourceforge.net/projects/sipsak.berlios/ On 10/10/2018 07:11 AM, Olivier wrote: Le mer. 10 oct. 2018 à 12:26, Joshua Colp > a écrit : On Wed, Oct 10, 2018, at 4:27 AM, Olivier wrote: > Hello, > > I think I met a case

Re: [asterisk-users] AGI timeout option

I don't know AGIspeedy, but I have some PHP scripts where I set a connect timeout using streams. Example using https, but should be easily adaptable to non-s http.: $pbxsh_bin = @file_get_contents("https://blah.blah.blah;, FALSE, @stream_context_create(array('https' => array('timeout' => 5,

Re: [asterisk-users] change dialing process on live call

Another way, using Local/ [do_dial] exten => s,1,Noop exten => s,n,Dial(SIP/1000/1001@do_dial,20) exten => s,n,Hangup() exten => 1001,1,Noop exten => 1001,n,Wait(20) exten => 1001,n,Dial(SIP/1001) exten => 1001,n,Hangup More detailed:

Re: [asterisk-users] getting real sip status after dial

en => n,Return On 06/09/2018 03:10 PM, Khalil Khamlichi wrote: Thanks for your response Eric, Here is some testing code, as you can see ${HANGUPCAUSE(${ARG1},tech)} is empty if number is not found (HANGUPCAUSE=1) or if sip request times-out (HANGUPCAUSE=0) (a dead far end for example) so I ha

Re: [asterisk-users] getting real sip status after dial

I think HANGUPCAUSE is channel agnostic. See: core show function HANGUPCAUSE Some thing like this IIRC: Set(my_cause=${HANGUPCAUSE(${CHANNEL(name)},tech)}) Remember the incoming leg of the call and the outgoing leg of the call are different channels. Make sure you are giving HANGUPCAUSE

Re: [asterisk-users] getting real sip status after dial

I think HANGUPCAUSE is channel agnostic. See: core show function HANGUPCAUSE Some thing like this IIRC: Set(my_cause=${HANGUPCAUSE(${CHANNEL(name)},tech)}) Remember the incoming leg of the call and the outgoing leg of the call are different channels. Make sure you are giving HANGUPCAUSE

Re: [asterisk-users] How to execute priorities following a caller hangup in a successful Dial?

added the following after the Dial, and it doesn't appear in cli >> after peer hangup: >> >> same => n,NoOp(After Dial ${AddressToReachPeer}) >> >> I also tried putting 'g' before the 'b'. >> >> I also tried removing the context headers of the hangup hand

Re: [asterisk-users] How to execute priorities following a caller hangup in a successful Dial?

id P wrote: Thanks, Eric. I just tried a hangup handler, but it's showing a similar problem: When the peer hangs-up, the hangup handler is not invoked and the caller channel remains open.  same => n(callPeer),Set(GLOBAL(Peer${IndexIntoPeers}CurrentCallsCount)=$[${PeerCurrentCallsCount} +

Re: [asterisk-users] How to execute priorities following a caller hangup in a successful Dial?

Use hangup handlers, they work around the issues with the 'h' extension. On 06/05/2018 05:33 AM, David P wrote: Thanks, Anthony. I added both 'g' and 'F' options. Now, when the caller hangs-up, my cleanup code is run by both the caller channel and the peer channel, but I only want the caller

Re: [asterisk-users] What does pct mean?

Could this gap in sequence numbers caused by a codec change generate errors like the one below? [2018-02-13 12:57:43] WARNING[4917][C-0004c2cb] codec_sangoma.c: [526559][g722toulaw] Got Seq 15944 but expecting 10106 (time since last read = 0ms), dropped 5838 packets On 02/13/2018 01:24 PM,

Re: [asterisk-users] Handling a long-running agi on hangup-handler?

Asterisk (after 1.4?) sends the AGI a HUP when the call hangs up. Try setting your script to ignore the HUP signal and make it fork and go into the background so Asterisk thinks the process has completed. In PHP ignore HUP:     pcntl_signal(SIGHUP, SIG_IGN); In PHP fork and become a short

Re: [asterisk-users] Asterisk 13.19.0 Now Available

See https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions for information regarding the release cycle.   It shows v14 went into security fix only mode on Sept 26 2017. On 01/12/2018 02:02 PM, Binarus wrote: Thanks for taking the time, but ... On 12.01.2018 12:04, Joshua Colp

Re: [asterisk-users] Answered time on channel

on the location of dialplan execution when the call hangs up, hangup handlers are attached to the call channel. You can attach multiple handlers that will execute in the order of most recently added first. On 12/26/2017 04:43 PM, Steve Edwards wrote: On Tue, 26 Dec 2017, Eric Wieling wrote: Don't use

Re: [asterisk-users] SIP invite timeouts : how is someone sending invites from our server ??

On 01/02/2018 05:30 PM, sean darcy wrote: On 12/30/2017 08:18 PM, Dovid Bender wrote: Script kiddies trying to find vulnerable systems that they can make calls on. Lock down the box with iptables and use fail2ban to block them. The via is probably bogus unless a box at the DoD was

Re: [asterisk-users] Answered time on channel

Don't use an 'h' extension, use a hangup handler. Use the MASTER_CHANNEL() function to set variables to ensure they are always set in the "top most" channel.  Below is an untested example, but is inspired by dialplan code I use in production.  Maybe it will help. [outbound] ; this is called

Re: [asterisk-users] General Kernel practices on CentOS

That only applies to DAHDI, not Asterisk. I add exclude=*kernel* to /etc/yum.conf so the kernel doesn't get upgraded accidentally and break DAHDI. On 12/20/2017 05:30 AM, Abdul Basit wrote: Olivier If you installed asterisk from source, you need to recompile it after kernel version

Re: [asterisk-users] Any impact on VoIP from loss of Net neutrality

and Congress is being forced to review the decision (and potentially finally create a proper law). ( http://uproxx.com/news/senate-democrats-cc-net-neutrality-fight/ ) So it is worth it to contact your Senator and let them know what you think they are supposed to be doing in your name. Eric Klein

Re: [asterisk-users] Asterisk download stats

I have tried to find these in the past, best I came up with was using Shodan.io search Looking for Asterisk I get: TOTAL RESULTS 42,036 TOP COUNTRIES United States 12,914 Russian Federation 3,173 Brazil 2,356 United Kingdom

Re: [asterisk-users] Having problem getting Asterisk to work on CentOS 7

This is what I hacked up for my CentOS 7 Asterisk server. It does NOT act like safe_asterisk. [Unit] Description=Asterisk PBX and telephony daemon Documentation=man:asterisk(8) After=network-online.target multi-user.target [Service] Type=simple User=root Group=root

[asterisk-users] Digium Announced the Call for Speakers is Now Open for AstriCon 2017

/02/astricon-2017-october-orlando/ - Announcement PR: http://www.prnewswire.com/news-releases/digium- announces-call-for-speakers-now-open-for-astricon-2017-300401428.html -- Eric Klein VP Operations & EMEA Sales GreenfieldTech Email eric.kl...@greenfieldtech.net Skype: EricLK

[asterisk-users] Digium Announced the Call for Speakers is Now Open for AstriCon 2017

/02/astricon-2017-october-orlando/ - Announcement PR: http://www.prnewswire.com/news-releases/digium- announces-call-for-speakers-now-open-for-astricon-2017-300401428.html -- Eric Klein VP Operations & EMEA Sales GreenfieldTech Email eric.kl...@greenfieldtech.net Skype: EricLK

Re: [asterisk-users] RS485 Audio device

If cable can be pulled , you have a couple of options. Long Reach Ethernet from Cisco is rated for 5,000 feet. Multi-mode fiber with fiber/ethernet media converters on each end would work and electrically isolate the two ends of the cable. Both are way overkill from a capacity standpoint, but

Re: [asterisk-users] Asterisk use with verizon hotspot

A few years ago I ran into something similar. Using TLS seemed to fix it, but it was a while ago so I might be wrong. On 10/14/2016 11:35 AM, Greg Woods wrote: On Fri, Oct 14, 2016 at 9:06 AM, Dovid Bender > wrote: Changing your port

[asterisk-users] [asterisk-dev] VoIP monitoring tools

(CPU, hard disk, processes, etc.), connectivity (ping, dropped packets, etc) and Asterisk processes (number of calls, channels, etc.). -- Eric Klein VP of Customer Experience GreenfieldTech Mobile +972-54-666-0933 Email eric.kl...@greenfieldtech.net Skype: EricLKlein Web: http

Re: [asterisk-users] Multiple phones when one is unregistered

The dial application dials devices not extensions. The only way to "dial" an extension from the dialplan is to use chan_local. On 09/01/2016 06:55 AM, D'Arcy J.M. Cain wrote: So does the Dial command go directly to the registered device or does it use the extension? I was assuming that it

Re: [asterisk-users] Multiple phones when one is unregistered

On 08/30/2016 11:31 AM, D'Arcy J.M. Cain wrote: Paste the Asterisk console output of a call showing the problem, Here is what the log shows. I can't put the unregistered user back at the moment. Perhaps I can do it overnight when no one is going to the building. [Aug 23 10:20:55]

Re: [asterisk-users] Multiple phones when one is unregistered

The dialplan below cannot go to voicemail, either something else is going on or the dialplan below is not the actual dialplan. Calls only go to voicemail when the dialplan runs the VoiceMail application. Paste the Asterisk console output of a call showing the problem, maybe someone can help.

Re: [asterisk-users] Need ISDN call generator

with BRI ( http://www.pikatech.com/warp-telephony-all-in-one/) Both ship world wide. There are other brands and models out there and depending on where you are located they may be better options based on shipping etc. Hope this helps, Eric On Sun, Aug 28, 2016 at 12:20 PM, Hooman Fazaeli <hoomanf

Re: [asterisk-users] DAHDI on CentOS 7

"make config" should also install the init script. On 08/15/2016 11:36 AM, Jerry Geis wrote: >On my Fedora 24 system, the "dahdi-tools" package contains an old-style >init script /etc/rc.d/init.d/dahdi, and this seems to work just fine with >systemd. I realize that CentOS != Fedora but if you

Re: [asterisk-users] Trouble applying regex to dialplan variable that contains double-quotes

How Set handles quotes can be changed with the 'app_set' setting in the [compat] section of /etc/asterisk/asterisk.conf. See also: https://wiki.asterisk.org/wiki/display/AST/Application_Set Perhaps you have the value left over from an old Asterisk setup. On 08/08/2016 04:31 PM, Alex

Re: [asterisk-users] Calls are dropped after 15 minutes

I've seen calls drop after 10 mins when SIP session timers are enabled. Try setting them to refuse in sip.conf. On 07/30/2016 02:07 PM, Keith Heppner wrote: We have a problem in that calls are dropped after 15 minutes (on both internal and out going calls, incoming calls do not seem to have

Re: [asterisk-users] Function SHELL not registered

Maybe Asterisk dialplan apps and functions don't work in the [globals] section. On 07/05/2016 11:40 AM, John Kiniston wrote: If you just need the name of the system it may be contained in the variable ${SYSTEMNAME}. This is assuming you have the systemname set in asterisk.conf

[asterisk-users] Anyone have problems with HPE 5130 EI Switch Series

phones. Seems that only the Yealinks are repeatedly rebooting whenever the user clicks the speaker button, otherwise the phones work properly. Their Cisco and Polycom phones work fine, and the Yealinks were fine prior to the upgrade. Anyone see anything like this before? -- Eric Klein Sr

Re: [asterisk-users] Migrating asterisk 11 to 13: some callers get no ringback tone any more

I don't know the default setting for progressinband in the code, but it is documented in Asterisk 11's sip.conf.sample as defaulting to never. Maybe the docs were fixed since Asterisk 11. from 11.21.x sip.conf.sample: ;progressinband=never ; If we should generate in-band ringing

Re: [asterisk-users] Debian 8.4 : dahdi startup scripts ?

hare/dahdi/dahdi_handle_device, which executes the scripts in /usr/share/dahdi/handle_device.d/ For my (simple) setup, this will probably "just work". I'll try removing /etc/init.d/dahdi next time I reboot my asterisk machine. -- Eric

Re: [asterisk-users] Debian 8.4 : dahdi startup scripts ?

't have it. Want to file a bug report or shall I? -- Eric Cooper e c c @ c m u . e d u -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory web

Re: [asterisk-users] Debian 8.4 : dahdi startup scripts ?

ion. The dahdi package (version 1:2.10.0.1-1) contains /etc/init.d/dahdi -- Eric Cooper e c c @ c m u . e d u -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Joi

Re: [asterisk-users] How to diable fax header (OR: what module could add that header?)

:05" (at the left border of the page) and "001/002" > (at the right border of the page). Are you using ReceiveFax()? If so, are you sure the unwanted information is already in the TIFF file? Or could it be produced by whatever program is being used to convert and/or view the TIFF

[asterisk-users] Final call for Astricon NASA trip

> > Hi everyone, > Sorry if you get this multiple times, cross posting to several lists > I need to give final count and payment on before Astricon starts, so need to know who is in and that they have paid. It is getting close to our trip to NASA and time to finalize the details and headcount.

[asterisk-users] Final update and Pricing for Astricon NASA trip

Hi everyone, Sorry if you get this multiple times, cross posting to several lists It is getting close to our trip to NASA and time to finalize the details and headcount. So here is the final information. If you have questions or want to join please contact me directly. *Trip date* Friday,

Re: [asterisk-users] How many Asterisk deployments?

Asterisk 73,819 AsteriskNOW 2,083 Astimax 46 Elastix 187 Freepbx 42 Other 32 Piaf 16 PIKA 899 TrixBox 95 VoiceOne 51 VoipNow 754 XVB 265 *Total Asterisk based PBXs * *78,428* Hope this helps, Eric Date: Fri, 7 Aug 2015 15:59:47 +0100 From: A J Stiles asterisk_l

Re: [asterisk-users] Looking for PRI Card with automatic fail over

/ PBXs not lines I am not sure that this would benefit us. So far I have found 1 company that seems to have a gateway card and that is the beroNet PCI / PCIe Gateway Cards which they say has the card level failover. I will update the list should this meet the use case. Eric

[asterisk-users] Looking for PRI Card with automatic fail over

I only have the 1 Asterisk box acting as a gateway and passing some calls out over SIP and IAX2. Any suggestions would be appreciated. Thanks Eric -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New

Re: [asterisk-users] Call Center

. We have done it for some large and complicated applications, and offer it as a service it you like. Eric Date: Sat, 1 Aug 2015 17:57:23 + From: Murthy Gandikota murth...@hotmail.com To: asterisk-users@lists.digium.com asterisk-users@lists.digium.com Subject: [asterisk-users

Re: [asterisk-users] showing sip number insted of pri number

that will block calls that are not from the assigned block of DIDs. So using JG's example if you send anything but one of the assigned numbers from the range of 1234-0 to 1234-99 they will block the call. This can be because it is the wrong length, format, or not in your assigned range. Eric

Re: [asterisk-users] Organizing a Pre Astricon road trip (Eric Klein)

Sorry, apparently I forgot that we are looking at the Monday before Devcon for this trip. -- Message: 4 Date: Wed, 24 Jun 2015 11:59:29 +0300 From: Eric Klein eric.kl...@greenfieldtech.net To: asterisk-users@lists.digium.com Subject: [asterisk-users] Organizing

Re: [asterisk-users] Receiving faxes with spandsp question

my [incoming] context and in my [screen-callers] context (equivalent to your auto-attendant, I guess). I think the time for fax detection can vary, so it might occur in either context. Then I define all the actual fax handling in the [fax] context. -- Eric Cooper e c c @ c m u . e d u

Re: [asterisk-users] asterisk email to fax

-sDEVICE=tiffg4. You'd have to define some syntax for specifying the outgoing number and cover page info in the email body, and extract the MIME attachment for the document from the email. Hope this helps. -- Eric Cooper e c c @ c m u . e d u

Re: [asterisk-users] asterisk email to fax

particular obsolete version of Windows. Since the O.P. said he's using it for his home office, I think he'll be able to control user expectations :-) -- Eric Cooper e c c @ c m u . e d u -- _ -- Bandwidth and Colocation

[asterisk-users] Organizing a Pre Astricon road trip

, as we get closer we will discuss total price and how people can pay it. Looking forward to seeing you in Orlando in October. Eric -- Eric Klein Sr. Consultant GreenfieldTech Mobile +972-54-666-0933 Email eric.kl...@greenfieldtech.net

Re: [asterisk-users] chanspy for group extension

Using Gosub / Return is well documented on voip-info.org, Asterisk The Definitive Guide, and many other places. Rehashing it on the mailing list would not be helpful. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Salaheddine

Re: [asterisk-users] switching from SIP to Skype..or not

Which wideband codec did you use when testing SIP? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ron Wheeler Sent: Thursday, March 12, 2015 9:39 AM To: asterisk-users@lists.digium.com Subject: Re:

Re: [asterisk-users] chanspy for group extension

This is one of the drawbacks to using macros. There are workarounds for macros, but the correct solution is use the Gosub / Return dialplan applications -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of

Re: [asterisk-users] Caller ID Names

Are the phones exposed to the internet (even using NAT)? If so there is a good chance these calls are not coming through your PBX but are coming in direct from someone, usually scammers. Polycom has a config option to disable accepting calls from unknown devices. No idea if Cisco has

[asterisk-users] Failsafe AGI using AEL

For the mailing list archive and for anyone else interested. A few years ago we needed to automatically run a second AGI if the first AGI failed i.e. a failsafe setup. Mainly because I'm not a very good programmer. 8-| The code below is very similar to what we use in production. This code

Re: [asterisk-users] Res_fax - FAXOPT(faxdetect)

I solved the issue by not answering the call as I assume others have done. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Administrator TOOTAI Sent: Wednesday, February 18, 2015 12:50 PM To:

Re: [asterisk-users] Debugging some DTMF Weirdness.

Is it possible your transmit or receive gain is too high and Asterisk is detecting the echo of a DTMF as a new digit cause by an analog leg of the call somewhere? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Kiniston Sent:

Re: [asterisk-users] Investigating international calls fraud

I’ve seen the following exploits of Asterisk / FreePBX boxes: 1) Default PlcmSpIp username and password for Polycom provisioning 2) Insecure SIP usernames and secrets 3) FreePBX GUI accessable from the internet 4) OS remote exploit (maybe ssh/ssl exploit) Mitigation options: 1) Don’t

Re: [asterisk-users] sip show channelstats reliable?

I've seen something similar with Adtran SIP gateways.When a re-invite happens the Adtran gets all confused about call stats and marks the pre-reinvite leg of the call as losing large numbers of packets.BTW, IIRC reinvites happen when a codec changes or the channel switches to T.38.

Re: [asterisk-users] Passing literals with commas to subroutine [SOLVED]

The easiest way is to escape the commas is with a \ (backslash). -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of A J Stiles Sent: Thursday, December 11, 2014 7:41 AM To: Asterisk Users Mailing List -

Re: [asterisk-users] Finish extension (avoid dialplan to silently continue in the next priority of another extension)

Hangup handling in Asterisk is horrible enough to make the Buddha cry. The (mostly working) solution I came up with was 1) Attach a hangup handler to the source channel as soon as possible in the dialplan to do whatever post call work which needs to be done. 2) Use the “g” and “F”

[asterisk-users] Weird SIP stuff

We are seeing something weird we haven't come across before. It seems they are sending us a different IP in the SIP from URI, than the IP they are actually sending us the traffic from. Basically, the traffic is coming from 65.211.180.237 but the header is: f:

Re: [asterisk-users] SIP call drops after 32 seconds, but only when....

Try setting directmedia=no in sip.conf. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Yves A. Sent: Saturday, November 22, 2014 8:06 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] SIP

Re: [asterisk-users] Upgraded to 13 and now Mailbox is empty in sip show peers

I doubt the person cares, if you don't like people top posting then stop reading their messages. If someone top posts, nothing you do will make them stop top posting. Complaining about something you cannot change just wastes everyone's time.I have a rule which deletes messages with top

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