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 => s

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 t

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 site

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 th

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 transaction

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 => 141,n,Dial(Multicas

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: externnotify=/home/phones/commands/deliver_vm

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, b

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 t

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 o

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 woul

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

rminal via "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 resol

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 yea

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 A

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 = n,S

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 better

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 interesti

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 r

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 physical

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 po

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) O

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 ast_yy

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 serv

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 P

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 P

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 SI

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 => s,n,Dial(${ARG2},

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 nothi

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, V

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 u

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 simi

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&Local/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: https://wiki.asterisk.org/wiki/display/AST/Delay+Dia

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

=${HANGUPCAUSE(${ARG1},tech)}) same => n,Goto(End) same => n(NotFound0),Set(SIPcause=SIP 500 Server Internal Error) same => n,Goto(End) same => n(NotFound1),Set(SIPcause=SIP 404 Not Found) same => n,Goto(End) same => n(End),Set( ${SIPcause} / $

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 t

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 t

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

;>> >>> [handleHangupByCaller] >>> ; Ensure that GLOBAL(CB${IndexIntoPeers}CurrentCallsCount) is >>> decremented after hangup, and end-of-call-epoch is set. >>> exten => s,1,NoOp(${PeerBeingConsidered} caller channel: Entered >>> handleHang

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

 same => n,Return() I've also tried replacing the Dial above with:  same => n,Dial(${DialForPeer},,g) Cheers, David On Tue, Jun 5, 2018 at 7:38 AM, Eric Wieling <mailto:ewiel...@nyigc.com>> wrote: Use hangup handlers, they work around the issues with the '

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, A

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 l

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 wrote:

Re: [asterisk-users] Answered time on channel

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'

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 comprimised

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 on

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 upgrad

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 Environment=HOME=/var/lib/ast

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 should fix all your worr

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 wa

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] WARNING[-1]

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] 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 h

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 Villací

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 th

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 https://wiki.a

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 al

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 Elhari

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 Adminis

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: [asterisk-u

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 somethin

[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: asterisk-users@l

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: Th

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 us

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. Also

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” opt

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 - Non-Comme

[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: ;tag=4-45026-159e4a6-995949f-159e

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 ca

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 p

Re: [asterisk-users] AGI and AMI in PHP -- What's current?

t: Re: [asterisk-users] AGI and AMI in PHP -- What's current? On Tue, 18 Nov 2014, Eric Wieling wrote: > Other than a few minor patches, we use stock phpagi. Can you spare me a flat spot on my forehead and share the wealth? -- Than

Re: [asterisk-users] AGI and AMI in PHP -- What's current?

Other than a few minor patches, we use stock phpagi. If you want simple, phpagi is the way to go. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Tuesday, November 18, 2014 3:34 PM To:

Re: [asterisk-users] Lost audio on forwarded calls

Any chance this is a simple directmedia and/or NAT issue? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Totaro Sent: Friday, October 03, 2014 4:14 PM To: tjrl...@live.com; Asterisk Users Mailing List - Non-Commercial Discussion S

Re: [asterisk-users] PBX hacked: why hundred of calls to the same number ?

We set up our servers to allowguest=yes and autocreatepeer=yes and use a global context setting to point any of those calls to an IVR jail.Attempts stop reasonably quickly. An empty "room" with an unlocked "door" is far less interesting than a room with the door locked. From: asterisk-user

Re: [asterisk-users] how to strip +1 out of incoming number

I prefer using FILTER() so if somehow CallerID arrived with something nasty it will be filtered out. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Kiniston Sent: Thursday, October 02, 2014 2:09 PM To: Asterisk Users Mailing List

Re: [asterisk-users] Asterisk removes ice lines in sdp when calling between webrtc clients

Asterisk is not a SIP Proxy. It is a B2BUA and will *always* replace the SDP with its own. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olli Heiskanen Sent: Thursday, October 02, 2014 9:06 AM To: Asterisk Users Mailing List - Non-C

Re: [asterisk-users] Change codec when dial from SIP to DAHDI

You will find not transcoding much less useful that one might imagine. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of d tbsky Sent: Thursday, September 25, 2014 2:57 PM To: Asterisk Users Mailing List - Non-

Re: [asterisk-users] read digits from the user through php agi script

I’m not going to help you debug your code, but I wanted to post part of a function from one of our internal AGIs which reads auth codes using a simple IVR. The code is ugly but it might be helpful to you. This code is released to the public domain. // no pin provided, get pin from caller

Re: [asterisk-users] Asterisk 11.9.0 PRI no ring indications

Ringback problems are a pain in the neck to troubleshoot. You don't mention your endpoint, but if the endpoint is sip, play around with the prematuremedia and progressinband options in sip.conf.The comments for these two settings in sip.conf.sample are completely and totally confuzing. Tr

Re: [asterisk-users] Asterisk prefix code to dial a high fraud country - security mechanism

: Re: [asterisk-users] Asterisk prefix code to dial a high fraud country - security mechanism absolutely not what I meant, I really meant to say thank you for respectfully pointing that out. -Motty On Thu, Sep 18, 2014 at 2:32 PM, Eric Wieling mailto:ewiel...@nyigc.com>> wrote:

Re: [asterisk-users] Asterisk prefix code to dial a high fraud country - security mechanism

. I had been working on Asterisk for the last 4 years, I am always learning something knew. - Motty On Thu, Sep 18, 2014 at 2:15 PM, Eric Wieling mailto:ewiel...@nyigc.com>> wrote: Your question demonstrates a fundamental lack of Asterisk concepts and knowledge. You should start by readin

Re: [asterisk-users] Asterisk prefix code to dial a high fraud country - security mechanism

Your question demonstrates a fundamental lack of Asterisk concepts and knowledge. You should start by reading http://www.asteriskdocs.org/ and go from there.Asterisk is not something you can learn in a few days. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@l

Re: [asterisk-users] Asterisk- VoIP Phones ring from Ext. 101 but that Ext. does not exist in extensions.conf

See: http://community.polycom.com/t5/VoIP/100-EXTERNAL-CALLS-UNWANTED-NUMERAL/td-p/50841 From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of motty cruz Sent: Tuesday, September 16, 2014 12:04 PM To: Asterisk Users Mailing List - Non-Commerc

Re: [asterisk-users] if statement recording - after hours

See ExecIf in the output of "core show applications". The IF function might be useful, see "core show functions". I assume the Asterisk Book also covers this. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf O

Re: [asterisk-users] Special functionality for Secretary/Boss

Sounds like you are running FreePBX. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mitul Limbani Sent: Thursday, September 04, 2014 6:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Speci

Re: [asterisk-users] Asterisk secure fine tune - stop attack

If we don't need to allow access from outside the USA we block access from all non-ARIN IP addresses by using iptables. This takes care of at least 80% of attacks. I enabled guest access and pointed all guest calls to an IVR which auto disconnects the call after a while (2 min seems good) if

Re: [asterisk-users] Does Asterisk 1.8. Supports Video Calls

risk-users-boun...@lists.digium.com] On Behalf Of Patrick Laimbock Sent: Tuesday, September 02, 2014 9:36 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Does Asterisk 1.8. Supports Video Calls On 02-09-14 22:52, Eric Wieling wrote: > A co-worker was doing video, I dislike video. The p

Re: [asterisk-users] Does Asterisk 1.8. Supports Video Calls

: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Patrick Laimbock Sent: Tuesday, September 02, 2014 7:54 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Does Asterisk 1.8. Supports Video Calls On 02-09-14 21:15, Eric Wieling

Re: [asterisk-users] Does Asterisk 1.8. Supports Video Calls

As long as you are NOT transcoding video should work in Asterisk. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Patrick Laimbock Sent: Tuesday, September 02, 2014 6:39 PM To: asterisk-users@lists.digium.com

Re: [asterisk-users] Custom SIP-header not present in call Asterisk to Asterisk

: Tuesday, September 02, 2014 12:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Custom SIP-header not present in call Asterisk to Asterisk On 02-09-14 14:22, Eric Wieling wrote: Try Hangup(123) where 123 is whatever "hangup cause" you want to

Re: [asterisk-users] Custom SIP-header not present in call Asterisk to Asterisk

Try Hangup(123) where 123 is whatever "hangup cause" you want to send back to the caller. The calliing Asterisk server will get the valuse back in HANGUPCAUSE variable. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Se

Re: [asterisk-users] RDNIS with tel: vs. sip: header

Looks like this was resolved recently. https://reviewboard.asterisk.org/r/3349/ -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Patrick Laimbock Sent: Thursday, August 28, 2014 12:02 PM To: asterisk-users@lis

Re: [asterisk-users] Dispatching calls question

-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Wednesday, August 20, 2014 9:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dispatching calls

Re: [asterisk-users] PRI timing settings

Do you also dial only 7 digits when calling from your cellphone when it works? Have you tried using the whole number in your dial? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Wednesday, August 20, 2014 5:29

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