Re: [asterisk-users] meetme conference playback of random sound file

Check out the Random Application and the RAND function, Here is a quick untested example for either. exten = s,1,Answer exten = s,n,Background(privacy-please-stay-on-line-to-be-connected) exten = s,n,Random(33:${CONTEXT},s,FILE1) ; 33% Num1 exten = s,n,Random(33:${CONTEXT},s,FILE2) ; 33% Num2

Re: [asterisk-users] Multiple public address to one Asterisk server behind NAT?

You could run two copies of asterisk on different private IP addresses. Have your current install bound to the first private IP with the externalIP set to the first public and the second install running on the other IP with the other externalIP set. On Tue, Feb 22, 2011 at 2:34 PM, Michelle

Re: [asterisk-users] Is it possible to dial an automated message when Asterisk receives an email?

You could use a procmail recipe to create a call file and then move it to the /var/spool/asterisk/outgoing directory. Below is a untested example .procmailrc: :0: * ^to.trig...@example.com | /usr/local/bin/callout.sh where callout.sh would look like this perhaps: !/bin/bash sleep 5

Re: [asterisk-users] asterisk hint SIP presence

On Tue, May 24, 2011 at 10:26 PM, Deka, Rajib IN MAA SL rajib.d...@siemens.com wrote: Hello List, Asterisk CLI command “core show hints” gives the list of hint extension configured and its presence status. In command output there is a field called “watchers” and it contains a numeric value

[asterisk-users] 1.4.38 passing a Regular expression containing a pipe character to a macro ?

Howdy, I'm working on a macro that authenticates the calling extension against a list of allowed extensions but it looks like the Expression I'm attempting to send of pipe separated extensions is showing up as additional arguments to my macro. I expected to have 4 arguments to the below macro,

Re: [asterisk-users] GOIP GSM to SIP Gateway?

On Tue, Dec 20, 2011 at 12:39 PM, Matt mhop...@gmail.com wrote: Is there anyway (short of defining dial an 8 from this phone for this trunk to this SIM and a 9 from this phone for a trunk to this SIM) to get it to use certain SIM cards when calls are made from certain phones? You could

Re: [asterisk-users] Mute DTMF

On Thu, Mar 29, 2012 at 12:09 PM, Todd Routhier fonema...@gmail.com wrote: I have been breaking my head on this, can't find a solution. Anyone know a way to mute DTMF on SIP? I have already tried changing the dtmfmode option and messing with different codec/dtmfmode settings but so far, not

Re: [asterisk-users] Dial Plan - Routing via Caller ID

On Thu, Apr 5, 2012 at 10:35 AM, list...@gmail.com wrote: If I wanted to route a call from a particular DID and the CALLERID from a specific A/C this doesn't seem to work for me: exten = 614000/_702XXX,n,Goto(context1,s,1) exten = 614000/614999,n,Goto(context2,s,1) exten =

Re: [asterisk-users] Get voicemail box password from dialplan?

On Mon, Jun 11, 2012 at 8:34 AM, Chet W. Stevens cwstev...@interact.ccsd.net wrote: Also, related to this question; is there a best or recommended method to determine the dialing extensions voice mail box? I have been using variations of ${CUT(CHANNEL,-,1)} which does work but I just have to be

[asterisk-users] AMI Reload action, returning generated errors?

Howdy, I'm building a webapp to allow my techs to do minor dialplan edits and trigger a reload on my PBX's running 1.8 I have no problem triggering a 'reload pbx_config.so' via manager, The problem is how can I see the results of my reload? For example a missing close parenthesis which would

Re: [asterisk-users] Checking messages from outside the network

You could add an 'a' extension to the context you are receiving calls in and remove the call to VoiceMailMain after your call to Voicemail. exten = a,1,VoiceMailMain(1001) The VoiceMail app listens for users to press '*' and directs callers to the 'a' extension in the same context if pressed.

Re: [asterisk-users] Invalid options

I think what Asmaa is indicating here is that inside app_voicemail (s)he's able to select options that are not relevant to the current menu (s)he is playing. That wouldn't be a dialplan problem or a DTMF issue, It would be inside the app_voicemail.c source. It looks like there is a big switch

Re: [asterisk-users] Add SIP Header for 1 SIP peer when calling a group of SIP peers

Use a LOCAL Channel and redirect that one peer through some dialplan Something like this: Dial(LOCAL/inno0@addheaderSIP/inno4SIP/inno6,30) [addheader] exten = inno0,1,SipAddHeader(foo) exten = inno0,n,Dial(SIP/inno0) exten = inno0,n,Hangup On Thu, Nov 14, 2013 at 9:35 AM, Jonas Kellens

Re: [asterisk-users] issue with speech in IVR

Look into the Authenticate application https://wiki.asterisk.org/wiki/display/AST/Application_Authenticate exten = 600,1,Ringing(2) exten = 600,n,Answer exten = 600,n,Authenticate(1234) exten = 600,n,Goto(home,s,1) On Thu, Dec 5, 2013 at 3:49 AM, Salaheddine Elharit salah.elharit...@gmail.com

[asterisk-users] Paging in waves.

I've been working on writing a subroutine to page groups of phones at once and I'm having some difficulty. My goal is to have a user call an extension, I record the page they wish to play, I then page out that recorded file to the phones in groups. [sub-masspage] exten = s,1,NoOP same =

Re: [asterisk-users] Paging in waves.

})is),10) same = n,Hangup ;end sub-masspage On Thu, Dec 5, 2013 at 5:36 PM, John Kiniston johnkinis...@gmail.comwrote: I've been working on writing a subroutine to page groups of phones at once and I'm having some difficulty. My goal is to have a user call an extension, I record the page

Re: [asterisk-users] Question about Asterisk 12

I looked on http://www.voip-info.org - maybe I missed it? The Digium/Asterisk site - I see all sorts of cool things about the REST API, but CLI - maybe I missed it!!?? - again, I could be looking in the wrong place? https://wiki.asterisk.org/wiki/display/AST/Home To my knowledge the

Re: [asterisk-users] Application Queue context that calls the extensions

app_queue dials Channels and not extensions unless your adding them to the queue as members using a local channel. I believe you can call Macro's and Gosubs from app_queue to set variables before the channels are bridged. On Mon, Jan 27, 2014 at 11:17 AM, Eduardo Leones

Re: [asterisk-users] Integration with outlook

We have used this commercial software to dial via our IP phones at my office. It's about $10 a license IIRC. http://www.theteletrigger.com/ On Tue, Jan 28, 2014 at 12:13 PM, bilal ghayyad bilmar...@yahoo.com wrote: Hello; Is there a method way to be able to dial the phone number through

[asterisk-users] Rejecting a call as if the extension does not exist.

I'm trying to address a problem with users transferring to invalid destinations. In my sip peer I'm setting both __FORWARD_CONTEXT and __TRANSFER_CONTEXT to a context with a extension defined below to set some CDR variables before the call is transferred. [customer-forward] exten =

Re: [asterisk-users] Rejecting a call as if the extension does not exist.

On Fri, Feb 7, 2014 at 2:16 AM, A J Stiles asterisk_l...@earthshod.co.ukwrote: You are suffering from classic Namespace Pollution. You need to put the extensions for which you are testing into their own separate context, e.g. customer-realexts; and include -that- context into your

Re: [asterisk-users] Pass Sound files as Argument to Macro Asterisk 1.8

It should be. I'd write something like the below: [macro-test] exten = s,1,NoOp exten = s,n,GotoIf($[${STAT(e,/var/lib/asterisk/sounds/${ARG1}.ulaw)} = 0]?NOPROMPT:PLAYBACK) exten = s,n(NOPROMPT),Background(nothing-recordedforpm-prompt-number) exten = s,n,SayPhonetic(${ARG1}) exten =

Re: [asterisk-users] quoting arguments to System command in dialplan

How about using the FILTER function to strip out anything you don't like from the CALLERID variables? Set(CIDNAME=${FILTER(A-Z,${CALLERID(NAME)})}) Set(CIDNUM=${FILTER(0-9,${CALLERID(NUM)})}) On Wed, Jul 2, 2014 at 2:25 PM, Eric Cooper e...@cmu.edu wrote: I'm trying to invoke a program to

Re: [asterisk-users] Dynamic Parking Lots. Music on Hold Class

Have you tried just setting the music on hold for the channel before you park your call? Set(CHANNEL(musicclass)=classical) I just did a quick test and it appears to work on 1.8 however I am not using a dynamic lot. On Thu, Aug 21, 2014 at 7:19 AM, Bryant Zimmerman brya...@zktech.com wrote:

Re: [asterisk-users] Pattern Extension not working in Dialplan

The first issue I see is you are attempting to insert your pattern match in the middle of your 's' extension, That's going to break your 's' extension. The second issue is that you are matching on XX which will match two digits, You need to match on _X instead if you are attempting to match on

Re: [asterisk-users] Polycom DND + Intercom/Paging Override?

On Wed, Sep 17, 2014 at 10:06 PM, Nathan Anderson nath...@fsr.com wrote: BUT Polycom handsets cannot be configured to just listen to RTP being multicasted to a particular multicast IP like many other IP phones can...the signalling for Polycom multicast paging and PTT functionality is

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

Try the Filter function Set(cid=${FILTER(0123456789,${CALLERID(NUM)})}) On Thu, Oct 2, 2014 at 10:52 AM, motty cruz motty.c...@gmail.com wrote: Hello, our VoIP send us caller ID +1(area)(number) for instance +16024224334 is there a way to strip +1 out of caller ID? -- Thanks for your

[asterisk-users] sip.conf to pjsip.conf conversion script

Howdy, I'm trying to get my feet wet with pjsip using the conversion script mentioned on the Wiki on this page: https://wiki.asterisk.org/wiki/display/AST/Migrating+from+chan_sip+to+res_pjsip I'm using the copy of the script that's included with Asterisk 13

Re: [asterisk-users] make asterisk do something when an outgoing call is picked up

Lee I recommend you use the MixMonitor application along with a combination of Playback to play your message to the Calling party and the A argument to Dial to play a file to the called party. So for your outbound calls: exten = _NXX,1,NoOP() same =

Re: [asterisk-users] PlayTones while in call

Henry, Both Montior and MixMonitor have a 'B' option that plays a periodic tone. B([interval]): Play a periodic beep while this call is being recorded. interval - Interval, in seconds. Default is 15. https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_Monitor

[asterisk-users] Function to get mailbox for a PJSIP Endpoint?

Howdy, I'm trying to re-write my voicemail check extension. I formerly used the SIPPEER function to get the mailbox for a peer with ${SIPPEER(${peer},mailbox)} Is there a way to do this with PJSIP now that I've converted over? I see a function PJSIP_ENDPOINT and it has a mailboxes subset but

Re: [asterisk-users] Function to get mailbox for a PJSIP Endpoint?

[7102](aor_dynamic) mailboxes=7102@default On Thu, Nov 6, 2014 at 6:09 PM, Joshua Colp jc...@digium.com wrote: John Kiniston wrote: Howdy, Kia ora, I'm trying to re-write my voicemail check extension. I formerly used the SIPPEER function to get the mailbox for a peer with ${SIPPEER

Re: [asterisk-users] Function to get mailbox for a PJSIP Endpoint?

(congestion) same = n,Congestion(8) same = n,Hangup() On Fri, Nov 7, 2014 at 6:26 PM, George Joseph george.jos...@fairview5.com wrote: On Fri, Nov 7, 2014 at 6:20 AM, Joshua Colp jc...@digium.com wrote: John Kiniston wrote: Here's my config, I am configuring the mailboxes as you see

[asterisk-users] pjsip phoneprov realtime?

Howdy, Is there a way to use realtime with phoneprov.com and pjsip? I've got a working pjsip realtime config currently but I have to add a phoneprov section to my pjsip.conf for each phone I want to provision. I was hoping the Sorcery page in the wiki would help possibly but it's blank :(

Re: [asterisk-users] pjsip phoneprov realtime?

, John Kiniston johnkinis...@gmail.com wrote: Howdy, Is there a way to use realtime with phoneprov.com and pjsip? Not yet. I forgot that bit in the initial version of the res_pjsip_phoneprov_provider module. I have a patch ready but it's tangled up in other stuff. I should be able

[asterisk-users] Get the status of a PJSIP endpoint?

Is there an equivalent to ${SIPPEER(${peer},status)} for PJSIP? The closest I've been able to get is to use AST_SOURCERY to see if they have a contact ${AST_SORCERY(res_pjsip,aor,${peer},contact) but I'm not certain if I'll still have a contact entry after a phone has gone unreachable? -- A

Re: [asterisk-users] Get the status of a PJSIP endpoint?

...@fairview5.com wrote: On Tue, Nov 18, 2014 at 10:55 AM, George Joseph george.jos...@fairview5.com wrote: On Mon, Nov 17, 2014 at 1:52 PM, John Kiniston johnkinis...@gmail.com wrote: Is there an equivalent to ${SIPPEER(${peer},status)} for PJSIP? I could swear there was a non-obvious way to get

Re: [asterisk-users] Asterisk13 don't execute h exten inside macros

Alonso, I can't answer if the 'h' extension inside macros was deprecated but I do see a change to CDR processing in the Upgrade notes for 13. https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+13 The endbeforehexten

Re: [asterisk-users] How to get BEEP BEEP BEEP when underline sends 486 Busy Here.

You don't actually do it in your Dial command, You do it after your dial command. Your Dial will return a DIALSTATUS variable, Check it and then process your busy. Here's some sample code I just banged out real quick for you. Untested but it should do the job. [sub-dialout] exten = s,1,NoOP()

Re: [asterisk-users] Problem with odbc connector with cdr

I notice you have MySQL-asterisk as your definition in your odbc.ini but you are trying to connect to simply 'MySQL' with your 'isql' command. Does isql work with 'MySQL-asterisk' as the DSN instead of simply 'MySQL' ? I have machines that use /etc/odbc.ini and machines that use

[asterisk-users] Debugging some DTMF Weirdness.

I'm attempting to find where my extra long DTMF Tones are coming from. I'm dialing from my sip handset through my proxy to my Asterisk box which is my PSTN Gateway. I'm pressing 4 to select a menu and everything is fine. [Feb 12 16:58:18] DTMF[29762] channel.c: DTMF begin '4' received on

[asterisk-users] Determining if a queue member is paused in Dialplan logic. [1.8]

Howdy, I'm looking at enabling autopause on one of my queues where my queue members are bad about leaving their desks without pausing. The problem I see is that when the queue pauses an Member it doesn't jump into the dialplan to do so which means my handy device state and asterisk database

Re: [asterisk-users] Determining if a queue member is paused in Dialplan logic. [1.8]

Thank you Kevin, I've looked at your solution and while I agree it's not ideal it does appear to be something that might work for me. I'll see if I can maybe backport the QUEUE_MEMBER stuff to 1.8 from 11. I'm also exploring an idea with a co-worker of using an AMI listener that will fire off

Re: [asterisk-users] OT - How does the blind transfer function work on Snom-870?

Take a look at two variables you can set on your SIP peer. TRANSFER_CONTEXT and FORWARD_CONTEXT You should be able to use this syntax in your sip.conf setvar=_TRANSFER_CONTEXT=kiniston-xfr You can then create the logic you need in your dialplan to change the ring using something like exten =

Re: [asterisk-users] Understanding the right way to get started with multiple trunks/extensions

In the 'home-number' example that was provided the caller ID was being replaced with the string 'Home' It's easy to prepend the caller ID instead however. Set(CALLERID(name)=Home-${CALLERID(name)}) You could even get fancy and set it based on what number was called, This would prepend the

Re: [asterisk-users] Dialing multiple channels with confirm

Thinking about it I don't think you want to do what you are asking, It sounds to me like you would create a race condition. Otherwise what happens when the Person A answers and accepts the call and Person B also answers and accepts the call? Which channel do you bridge your call with? Person A

Re: [asterisk-users] situation with ivr and four-channel gateway

I'd recommend using DEVICE_STATE On your extension 101, Check the DEVICE_STATE of peer SIP/101, If it's not 'NOT_INUSE' then dial it, Otherwise dial SIP/102 exten = 101,1,ExecIf($[${DEVICE_STATE(SIP/101)}=NOT_INUSE]?Dial(SIP/101,40)) same = n,Dial(SIP/102,40,t) same = n,Hangup()

Re: [asterisk-users] Asterisk Inbound calls, multiple SIP accounts, calledID

Can you show us the CDR record for that call? And maybe what your s priority of your incoming context is? It should be easy to get what number was dialed, Try: ${CUT(PASSTHRU(${SIP_HEADER(TO):5}),@,1)} Normally I display the callers number on my phones, Not the number they dialed? On Wed, Apr

Re: [asterisk-users] dial out with channel variable; sub-string usage

BABY appears to be a global variable in your example. In your CLI output testcarrier is a peer, It's not a variable at all. The context field for your peer testcarrier is where incoming calls from testcarrirer will be routed to. Here is some example dialplan showing how you can use one context

Re: [asterisk-users] Asterisk Inbound calls, multiple SIP accounts, calledID

Andrew, Instead of your SET and GOTO blocks I'd recommend using the Asterisk DB to make things easier to maintain. You could make two database entries for each of your DID's database put 4259981810 name JohnPersonal database put 4259981810 target kiniston-extern,john-personal,1 Then you could

Re: [asterisk-users] Manipulate extension state in 1.8.x

You can use a custom device state to do it. [dnd] ;DND Toggle exten = *363,1,Answer() same = n,Set(CURRENT_PRESENCE=${DEVICE_STATE(Custom:DND${CHANNEL(peername)})}) same = n,GotoIf($[${CURRENT_PRESENCE}=NOT_INUSE]?*78,1:*79,1) ;DND On exten = *78,1,NoOP(Turning DND On) same =

Re: [asterisk-users] Voice mail and caller ID

Try this for CHAN_SIP: same = n,Set(Peer=${SIPCHANINFO(peername)}) ; Get the peer same = n,Set(MailBox=${SIPPEER(${Peer},mailbox)}); Get the mailbox same = n,VoicemailMain(${MailBox}@LocalSets,s) ; If we have a mailbox defined log into it If you are using PJSIP it's more complex

Re: [asterisk-users] Voicemail: saycid without prefix

The easiest solution may be to strip the leading zero's off your caller ID before your caller enters the Voicemail app to leave you a message. ExecIf(REGEX(^[0][0]. ${CALLERID(NUM)})?Set(CALLERID(num)=${CALLERID(NUM):2})) On Fri, Jul 3, 2015 at 10:53 PM, Luca Bertoncello lucab...@lucabert.de

Re: [asterisk-users] Asterisk pin code for out-going international calls (safeguard against fraud)

The Authenticate application will do this for you. https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_Authenticate You can either give it a single PIN to use for all calls, Authenticate using a value in the Asterisk Database, Or use a plain text file for the PIN's On Mon, Jul

Re: [asterisk-users] Voicemail: saycid without prefix

Nice! I didn't know what dialing rules may apply to his location, Your code does look like an improvement on mine tho. I love the REGEX function. Even better, if the first 4 digits are 0049, you could replace them with 0 as though it was an inland call: ExecIf(REGEX(^0049.

Re: [asterisk-users] Asterisk pin code for out-going international calls (safeguard against fraud)

I don't see that the Authenticate application has return values for failure cases or returns call control on a failure case. Sorry I don't think you will be able to do what you want with it. On Tue, Jul 7, 2015 at 12:22 PM, Motty Cruz motty.c...@gmail.com wrote: Here is what i have, exten =

Re: [asterisk-users] Queues don't follow dialplan if no members are registered

Martin amar...@xes-inc.com wrote: - Original Message - From: John Kiniston johnkinis...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, July 28, 2015 12:12:05 PM Subject: Re: [asterisk-users] Queues don't

Re: [asterisk-users] Queues don't follow dialplan if no members are registered

In your queues.conf do you have a leavewhenempty and joinempty set? in queues.conf [myqueue] leavewhenempty = strict joinempty = strict strategy = ringall ringinuse = no On Tue, Jul 28, 2015 at 9:58 AM, Andrew Martin amar...@xes-inc.com wrote: Hello, I am running Asterisk 11 on CentOS 6.x.

Re: [asterisk-users] no ringing tone with Dial option r

Have you checked your indications.conf? I've seen a missing or misconfiguration in the zone definition cause this. On Tue, Nov 3, 2015 at 11:07 AM, sean darcy wrote: > On 11/01/2015 12:38 PM, sean darcy wrote: > >> I'm not getting any ringing when I use option r with Dial:

Re: [asterisk-users] Sending E-Mail from voicemail

Deliver your voicemail to two boxes, One set to email the attached file and other set not to attach. exten = 700,1,Voicemail(701@users702users,u) [users] 701 = 1234,Hello There,he...@how.are.you,,attach=no 702 = 1234,Hi Hi,he...@how.are.you,,attach=yes|delete=yes ​ --

Re: [asterisk-users] Queue grouping - how can it be implemented?

Use Local Channels and hints to combine SIP/MOM and SIP/MOMMobile into a single extension you add to the queue. extensions.conf: [queue-phones] exten => MOM,1,Dial(SIP/MOM/MOMMOBILE,60,tkw) exten => MOM,hint,SIP/MOM/MOMMOBILE exten => DAD,1,Dial(SIP/DAD/DADMOBILE,60,tkw) exten =>

Re: [asterisk-users] How to simulate 100 phones in a lab ?

You could use Custom Device States to create your 100 extensions to watch and update their status by changing the states manually. With Asterisk 13.0 and PJSip I had the RLS feature working with some Yealink phones (T28P and a T41P to be specific), on the phone I set:

[asterisk-users] Asterisk 13.7.0 AutoMixMonitor

On my older Asterisk installs I'm still using Automon because I can set MONITOR_EXEC to run my post process command and use MONITOR_EXEC_ARGS to send it some options I need by adding those to my sip.conf entries with SetVar lines. On my Asterisk 13.7.0 box I want to use the

Re: [asterisk-users] PJSIP Stun/ICE

Just an idea for a work around, Have you thought about putting a proxy between your PBX and the Internet such as openSIPS or Kamilio? That way you may not need to change your IP inside pjsip, Let your proxy handle it. I gave up switching my edge asterisk to pjsip at least twice because I >

Re: [asterisk-users] Asterisk 13.7.0 Pickup with namedcallgroup/namedpickupgroup

On Tue, Feb 2, 2016 at 11:11 AM, Richard Mudgett wrote: > > Since you didn't specify the channel driver, I took a quick look at the > chan_dahdi, chan_sip, and chan_pjsip channel drivers to see if they > set any default groups. I didn't see any of those channel drivers set

[asterisk-users] Asterisk 13.7.0 Pickup with namedcallgroup/namedpickupgroup

Should setting a namedcallgroup & namedpickupgroup supersede numeric callgroups and pickupgroup ? I've got 5 peers on my 13.7.0 box, Three of them have a namedcallgroup & namedpickupgroup of 'kiniston' and Two of them have a namedcallgroup & namedpickupgroup of 'sanday'. I'm not specifying a

Re: [asterisk-users] Voice recognition IVR Is it possible?

I think I saw an old page on the voip-info wiki on how to use CMU Sphinx with asterisk. http://www.voip-info.org/wiki/view/Sphinx IMHO It's not going to be anywhere as good as a commercial product without a lot of work. On Mon, Feb 22, 2016 at 11:34 AM, Daniel Chavez

Re: [asterisk-users] 11.21,2 : how to transfer to Jolly Roger ?

You can do this with setting up an application map using DYNAMIC_FEATURES and enabling it on your incoming call paths. https://wiki.asterisk.org/wiki/display/AST/Custom+Dynamic+Features If you don't want to even answer the calls you could try doing this with 'ex-girlfriend logic', I personally

Re: [asterisk-users] Voice recognition IVR Is it possible?

I saw Lumenvox offering Speech Recognition for asterisk at a past Astricon. http://www.lumenvox.com/partners/digium/Asterisk.aspx On Mon, Feb 22, 2016 at 11:00 AM, Daniel Chavez wrote: > Hello list, > I was wondering if it were possible for asterisk to do a voice

Re: [asterisk-users] GSM Gateway behind SIP ATA?

Have you turned on sip debugging? Do you see the caller ID in the invite from your Gateway to your PBX? On Tue, Jan 26, 2016 at 2:07 AM, Belal wrote: > Dear sir, > > what about receiving call from a GSM gateway. I didn't see the caller ID?. > is it happen to you?

Re: [asterisk-users] Using Asterisk to play Icecast streams

Could you setup a local instance of Icecast and point your PBX to it? It's been years since I did any streaming but I recall my icecast relay would only consume bandwith when it had listeners connected to it. Then you wouldn't have to worry about how many people were listening to a single

Re: [asterisk-users] asterisk admin interface

You could explore using ARI with it's Push configuration. https://wiki.asterisk.org/wiki/display/AST/ARI+Push+Configuration On Mon, May 16, 2016 at 11:33 AM, Goke Aruna wrote: > hi all, > can anyone give me a guide on any asterisk admin solution / interface for > config

Re: [asterisk-users] Asterisk 13 Realtime Voicemail frustrating issue

Have you tried using the table definition that comes with the Asterisk source? the file mysql_config.sql is located in contrib/realtime/mysql and defines a very different voicemail table than what you have in your configuration. On Tue, May 3, 2016 at 3:10 AM, Michele Pinassi

[asterisk-users] Call a subroutine via Originate?

Howdy everyone, I'm writing a little click to dial type tool and I've run into a snag where my Originate command needs to call a Sub routine to do a database lookup and some other stuff. I can't seem to get the syntax right to call Gosub with Originate Just testing with the command line I've

Re: [asterisk-users] Function SHELL not registered

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.asterisk.org/wiki/display/AST/Asterisk+Main+Configuration+File That said, for SHELL support you probably need to set :

Re: [asterisk-users] VoiceMail - Allow * for only some users

I think you almost have it. In your vmfwd context have a wildcard match that sends the caller back to the originating voicemail and then define specific extensions that are allowed to forward. [vmfwd] exten => _,1,Voicemail(box@context,option) same => n,Hangup ; Andrew Ruthven

Re: [asterisk-users] how to join 2 channels using AGI/AMI

I think a simpler way to do this would be to define an member in your queues.conf that points to a local channel that calls the remote users cell phone. You can use the M option in your dial to run a macro to prompt the user to accept the call. Here's my connector macro, I call it with:

Re: [asterisk-users] how to join 2 channels using AGI/AMI

g. > -- Executing [s@macro-myconnector:3] GotoIf("SIP/pbx2-04b2", > "1?REJECT,1") in new stack > > Any idea? > > > > > > > 2016-06-30 21:50 GMT+02:00 John Kiniston <johnkinis...@gmail.com>: > >> I think a simpler way to do this w

Re: [asterisk-users] How to remove unused custom hints?

You can delete them from the astdb with database del. do a 'database show' and your devices should show up in the tree under 'CustomDevstate' On Mon, Aug 15, 2016 at 9:42 AM, Tomas Holy wrote: > Hello list members, > after programing of dialplan I have some messy

Re: [asterisk-users] Dial and start music on hold after timeout

This seems like the obvious answer but maybe I'm misunderstanding the question. exten => s,1,Dial(SIP/alice,20) same => n,Playback(myannouncement) same => n,NoOP(Whatever else you want to do goes here) On Mon, Aug 22, 2016 at 8:36 AM, Jean Aunis wrote: > Hello, > >

Re: [asterisk-users] Dial and start music on hold after timeout

22/08/2016 à 17:42, John Kiniston a écrit : > > This seems like the obvious answer but maybe I'm misunderstanding the > question. > > exten => s,1,Dial(SIP/alice,20) > same => n,Playback(myannouncement) > same => n,NoOP(Whatever else you want to do goes here) > >

Re: [asterisk-users] Recording "Never" on extension not stopping recording

I'd recommend trying the support resources for the GUI that is managing your asterisk installation. This looks like it might be FreePBX dialplan logic to me, Most likely it won't be something that the list will be able to help you modify. http://community.freepbx.org/ is the FreePBX Community

Re: [asterisk-users] Dial and start music on hold after timeout

, 2016 at 8:45 AM, Jean Aunis <jean.au...@prescom.fr> wrote: > Sorry, I forgot to write that the SIP peer must keep ringing while the > announcement is being played. > > Le 22/08/2016 à 17:42, John Kiniston a écrit : > > This seems like the obvious answer but ma

Re: [asterisk-users] Blacklist callers from file

Here is a quick and dirty bash script to do it that I wrote you. #!/bin/bash if ( asterisk -rx "database deltree blacklist") then echo "Blacklist Cleared" else err "ERROR Failed to clear Blacklist, Exiting." exit 1; fi while IFS=,

[asterisk-users] res_pjsip parkinglot configuration?

I'm trying to find where you configure the parking lot used by phones registered via pjsip. In sip.conf you could set the default lot for call parking with the 'parkinglot=mylot' setting but I don't see an equivalent in pjsip.conf Do I need to use setvar to set CHANNEL(parkinglot) on my endpoint

[asterisk-users] Finding the user agent of a channel using PJSIP?

I'm working on my sip to pjsip translation. Right now I do some functionality based on what the user agent is on the calling phone using: ${SIPPEER(${CHANNEL(peername)},useragent)} I'm trying to replace it with PJSIP_CONTACT(${CHANNEL(contact)},user-agent}) but I'm not getting any data

Re: [asterisk-users] Any way of creating a file to write to from the dialplan, or must I use AGI?

I'm able to use the FILE function to create files just fine. Set(FILE(${CALLFILE},,,al,u)=Extension: s) On Fri, Nov 4, 2016 at 2:26 PM, Jonathan H wrote: > Seems I can write to an existing file, but is there really no way of > creating a new file to log some data to,

Re: [asterisk-users] Any way of creating a file to write to from the dialplan, or must I use AGI?

> that directory. Hmmm.... > > On 4 November 2016 at 21:50, John Kiniston <johnkinis...@gmail.com> wrote: > > I'm able to use the FILE function to create files just fine. > > > > Set(FILE(${CALLFILE},,,al,u)=Extension: s) > > > > On Fri, Nov 4, 2016 at

Re: [asterisk-users] Say duration of alaw file?

Alright... How about: exten => 100,1,NoOP() same => n,Set(Duration=$[CEIL(${STAT(s,/var/lib/asterisk/moh/reno_project-system.alaw)} / 8000)]) same => n,NoOP(Duration is $[FLOOR(${Duration} / 60)] Minutes, $[REMAINDER(${Duration},60)] Seconds) same => n,Hangup() On Tue, Oct 18, 2016

Re: [asterisk-users] IAX - Equivalent of SipAddHeader

<ad...@tootai.net> wrote: > Le 24/10/2016 à 18:46, John Kiniston a écrit : > >> You can do it with IAXVAR. >> >> https://wiki.asterisk.org/wiki/display/AST/Asterisk+14+Function_IAXVAR >> >> >> exten => 100,1,Set(IAXVAR(myvar)=foo) >> exten

Re: [asterisk-users] IAX - Equivalent of SipAddHeader

You can do it with IAXVAR. https://wiki.asterisk.org/wiki/display/AST/Asterisk+14+Function_IAXVAR exten => 100,1,Set(IAXVAR(myvar)=foo) exten => 100,n,Dial(OTHERHOST/201) on OTHERHOST exten => 201,1,Set(myvar=${IAXVAR(myvar)}) exten => 201,n,NoOP(My variable is ${myvar}) On Mon, Oct 24, 2016

Re: [asterisk-users] Say duration of alaw file?

This is the most compact I can make it just by tidying up your variables and playbacks: same => n,Set(ARRAY(minSpeech,playFile)=minutes,/var/lib/asterisk/sounds/en/abandon-all-hope) same => n,Gosub(setup) same => n,set(playFile=/tmp/reno_project-system) same => n,Gosub(setup)

Re: [asterisk-users] CHANNEL(accountcode) only stores 19 chars

Line 161 of include/asterisk/channel.h:#define AST_MAX_ACCOUNT_CODE20 /*!< Max length of an account code */ That's where your limit is coming from, I see some other places where the code doesn't use that definition ( chan_ooh323, cdr_sqlite, cdr_tds , res_config_sqlite) So as long as you are

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

I always set a TIMEOUT(absolute) on calls across trunks to something reasonable like 10 hours, that way calls should end in a sane amount of time even if something weird happens. Otherwise I've always had to do a reload when I couldn't hang up from the CLI. On Thu, Nov 3, 2016 at 9:16 AM, Carlos

Re: [asterisk-users] Tranfer the called number in 3 way call

You could use the IAXVAR() function to set variables before dialing your IAX peer on the initial PBX that then get retrieved by the 2nd PBX. PBXA: same => n,Set(IAXVAR(CALLERID)=${CALLERID(num)}) same => n,Set(IAXVAR(DNID)=${CALLERID(DNID)}) PBXB: same => n,NoOP(CallerID is ${IAXVAR(CALLERID)}

Re: [asterisk-users] sip show [general]?

'sip show settings' may do what you want. On Wed, Jan 11, 2017 at 3:32 AM, Thufir Hawat wrote: > I appreciate that the console lets you see the details for a peer with > "sip show peer foo". Certainly, I can look in sip.conf to see the > [general] context, but can I

Re: [asterisk-users] anveo, a different kind of trunk provider?

On Mon, Jan 2, 2017 at 5:26 AM, Thufir Hawat wrote: > > But, their registration string with Asterisk is: > > Locate [general] secion and add the following > register => ACCOUNT_NUMBER:sip_passw...@sip.anveo.com:5010 > > > Wouldn't this send every outbound call through

Re: [asterisk-users] SIP peer authentication

type=peer matches on the IP of the specified host, If you want to match on the username use type=user. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at:

Re: [asterisk-users] asterisk name in mysql

You can use func_odbc to do this. https://wiki.asterisk.org/wiki/display/AST/Getting+Asterisk+Connected+to+MySQL+via+ODBC2 There is a good chapter in the Asterisk book about using ODBC for hotdesking that may help you understand ODBC as well.

Re: [asterisk-users] moh reload not reloading/reading new musiconhold files

Your new file is in the 'myfolder/1'' subdirectory of the MOH directory. Either move the file into the MOH directory or define a new class in musiconhold.conf that is for your directory. On Fri, Mar 3, 2017 at 7:19 AM, Jonas Kellens wrote: > Hello > > using Asterisk

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