Re: [asterisk-users] Digium or Sangoma? What happened to Digium cards

Sangoma purchased Digium. You can find Sangoma cards at https://www.sangoma.com/telephony-cards/ On Tue, Jan 12, 2021 at 2:29 PM bilal ghayyad wrote: > Hello All; > > We were using Digium cards, now I am not able to reach for digium website > that contains the telephony cards and Asterisk

[asterisk-users] Is anyone using autohints=yes with Queue hints and PJSIP?

Hello, I'm working on converting my 18.0.1 test system from SIP to PJSIP and I've run into something odd. I have a queue defined named acme-test that has two agents in it, PJSIP/7001acme and PJSIP/7002acme. I have autohints=yes in my acme-intern context, I have not defined hints for either of

Re: [asterisk-users] AMI vs. Dialplan Originate

You could do the old school method and create and move a .call file from your dialplan. exten => writefile,1,NoOP() same => n,Set(CALLFILE=/var/spool/asterisk/tmp/${FileName}-${ARG1}.call) same => n,Set(FILE(${CALLFILE},,,al,u)=Channel: SIP/bob) same => n,Set(FILE(${CALLFILE},,,al,u)=WaitTime:

Re: [asterisk-users] Problem with OPTIONS requests.

t; to accept a 404 reply to a SIP:OPTIONS as a valid response. > > > Hope it helps. > > Cheers, > Joel. > > > On Thu, Jul 16, 2020 at 5:04 PM John Kiniston > wrote: > >> I'm implementing a SBC with my Asterisk PBX but the keeps disabling the >> trunk

[asterisk-users] Problem with OPTIONS requests.

I'm implementing a SBC with my Asterisk PBX but the keeps disabling the trunk group I've configured and I think it may be because Asterisk is returning a 4r04 to the OPTIONS. I've created a test context and have put in a wildcard pattern match to try and catch those options but it doesn't seem to

Re: [asterisk-users] Redis in place of astdb

Dovid, You could use func_odb + a ODBC Redis driver to keep from having to shell out. On Wed, Jul 8, 2020 at 4:37 AM Dovid Bender wrote: > Hi, > > Does anyone know of any projects that would allow you to use Redis in > place of AstDB? By in place of I don't mean for what Asterisk needs but to >

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

Nice, Do you have the code up on GitHub? I'd love to see it. What's the source of the data? Something API driven I hope? Have you thought about implementing your project via curl instead of func_odbc? On Wed, May 27, 2020, 8:52 PM Saint Michael wrote: > In a few weeks, no SIP call is going to

Re: [asterisk-users] Dialplan - using multiple AND or OR in set is it possible ?

Use the ARRAY version of Set. same = n,ExecIf($["A" = "B"]?Set(ARRAY(C,D)=1,2)) On Tue, Apr 21, 2020 at 3:56 AM Administrator wrote: > Hello, > > we want to use something like > > same = n,ExecIf($["A" = "B"]?Set(C=1) & Set(D=2) & ...) > > Problem is that result gives C=1) & Set(D=2) & ... > >

Re: [asterisk-users] Help with FUNC_MATH

m looking at the wiki won't STRFIME just give me what I need based on > the unix time that I put in? What I am actually looking to do is convert > over from 12 hour format to 24 (unless strftime does just that and I don't > kow what am I am doing?). > > > > On Thu, Feb 13,

Re: [asterisk-users] Help with FUNC_MATH

Try using the STRFIME function instead of doing this by hand. https://wiki.asterisk.org/wiki/display/AST/Function_STRFTIME *%H* The hour as a decimal number using a 24-hour clock (range 00 to 23). *%I* The hour as a decimal number using a 12-hour clock (range 01 to 12). On Thu, Feb 13, 2020

Re: [asterisk-users] Site to site VPN problems

Ira, What version of Asterisk are you using, and what channel driver? There has to be a better way than to create hundreds of peer entries. On Thu, Dec 12, 2019 at 12:26 PM Ira wrote: > Hello Jan, > > Tuesday, December 3, 2019, 8:49:28 PM, you wrote: > > Jan> The next thing to look at is

Re: [asterisk-users] Better audio in than just 8k

Jerry, What if you specify a higher bitrate to mpg123? You are limiting it to 8k with the 'r' option. I convert my source audio files with sox to 16khz signed linear for wideband hold music. sox -c1 hold.wav -r 16000 -c 1 -e signed-integer -r 16k hold.raw Then I rename the .raw file to a

Re: [asterisk-users] Hacking

On Sun, Jun 16, 2019 at 3:37 PM John T. Bittner wrote: > Anyone know how someone can hack an asterisk box and register with every > single account on the box. > > This box only has 3 accounts, with very complex passwords. Have VoIP > blacklist setup and fail2ban… > I've seen this happen when

Re: [asterisk-users] Read like Application but to limit allowed digits

The solution you don't want to use (testing the result of Read) is what I personally use. same => n(PROMPT),Read(RESPONSE,playback_or_callback,1,,3,8) same => n,GotoIf($["${RESPONSE}" = "1"]?RETURN,1) same => n,GotoIf($["${RESPONSE}" = "2"]?PLAY,1:PROMPT) On Mon, Apr 8, 2019 at 11:45 AM

Re: [asterisk-users] Asterisk 1.8.7.0 connectivity to Avaya SM

Thomas, Does the Asterisk box need to do anything other than handle calls for this one specific IVR? IE does it ever originate calls? If it's only recieving calls then I'd turn on guest access and not even bother with a peer. Just set [general] context=transit-ivr allowguest=yes On Tue, Feb

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

at 12:14 PM Brian J. Murrell wrote: > On Wed, 2019-02-20 at 11:46 -0700, John Kiniston wrote: > > Use the IF function to evaluate and change the dial command directly. > > Thanks for taking the time, but that doesn't actually answer the > question I asked. It in fact an

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

Use the IF function to evaluate and change the dial command directly. My braces and parens may be off in this example sorry if it doesn't work out of the box. exten => s,n,Dial(${IF($["${SIP}" = "PJSIP"]? ${PJSIP_DIAL_CONTACTS(${STRREPLACE(ARG2,"PJSIP/","")})}{ARG2})},20,TtWw) On Wed, Feb 20,

Re: [asterisk-users] (NAT) direct media to host on local net when registering from external address

How is your endpoint currently configured in asterisk? Have you tried rtp_symmetric to see if the endpoint sends audio to asterisk if asterisk can send audio back to the client? Alternatively if your SIP Proxy is also a Media proxy you could set the media_address on the endpoint to be your proxy

Re: [asterisk-users] PJSIP_HEADER - Diversion header manipulation

Davor, Have you attempted to do a Set(PJSIP_HEADER(remove,Diversion)=)) in your add_diversion context to remove the header Asterisk is passing through? On Mon, Dec 10, 2018 at 3:04 AM Davor Jovanovic wrote: > Hi all, > > I’m trying to rewrite Diversion header when call forwarding is done on >

Re: [asterisk-users] Queues and penalties

This should work, How are you defining your timeouts in the queues.conf ? And to verify, in your extensions.conf you are calling Queue with the queue name and the ruleset to apply from queuerules.conf? On Wed, Nov 28, 2018 at 12:45 PM Paddy Grice wrote: > Hi All > > I have been looking at this

Re: [asterisk-users] Queue not dialing out to cell phone for some reason

0 > retry=1 > > Which means if I send it to "Eric" - it will go to his voicemail after 30 > seconds. Should I change timings? > > Thank you! > > -- > *From:* John Kiniston > *To:* Ivan Demkovitch ; Asterisk Users Mailing > List - Non-Commercial Discussion >

Re: [asterisk-users] Queue not dialing out to cell phone for some reason

hannel? Can you be more specific? I'm not > very good at this :) > > > > This is logger.conf. Where(which section) should I place logging > configuration? > > [general] > dateformat=%F %T > > [logfiles] > console => notice,warning,error,dtmf > messages =

Re: [asterisk-users] Queue not dialing out to cell phone for some reason

what does the output of 'queue show sales' show? Do you have queue logging enabled? Have you looked in the queue log to see what events are firing? On Thu, Nov 15, 2018 at 9:55 AM Ivan Demkovitch wrote: > Hello, > > I have queues.conf setup with a group like so: > > [Sales](StandardQueue) >

Re: [asterisk-users] BLF screen alerts

Good Morning John, Most likely you would need to make changes on your phones themselves. PJSIP is just sending NOTIFY messages when there is a state change, The phone itself is choosing how to handle the message. On Mon, Nov 5, 2018 at 10:40 AM John T. Bittner wrote: > Anyone know how to

Re: [asterisk-users] Dropped calls when all DAHDI lines in use

You could use GROUP & GROUP_COUNT to track how many channels you are using before you attempt to dial out and send back a Busy/Congestion/Whatever to your endpoint when you are at your limit. On Mon, Oct 8, 2018 at 4:33 PM Andrew Martin wrote: > Hello, > > I am running Asterisk 11.17 with DAHDI

Re: [asterisk-users] feeling n00b again

It's a configuration issue with the peer (or the peer it's matching) in sip.conf You have configured the peer to use avpf but your phone is not attempting to do avpf. Either configure the phone to match the entry in sip.conf or change sip.conf to match the phone. On Mon, Aug 27, 2018 at 4:56

Re: [asterisk-users] Is there a way to remove launching shell command from Asterisk CLI

I use sudo to limit this. Cmnd_Alias CAPTAGENT = /sbin/service captagent stop, /sbin/service captagent start, /sbin/service captagent restart Cmnd_Alias ASTERISK = /sbin/service asterisk stop, /sbin/service asterisk start, /sbin/service asterisk restart, /usr/sbin/rasterisk, /usr/sbin/asterisk,

Re: [asterisk-users] G729

On Fri, Jul 20, 2018 at 11:41 AM Saint Michael wrote: > ​The community would benefit if a non/licensed version of G729 would be >> included with Asterisk​, since the license expired. The current codec >> source code posted still requires licensing. >> > ​I am sure Digium would not prefer to ​ >

Re: [asterisk-users] How to steal an answered call?

David, You should be able to use the Bridge dialplan application to do what you want. https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_Bridge I use the CHANNELS function and the IMPORT function to find the channel to bridge to my caller. On Sun, Jul 8, 2018 at 8:17 PM David

Re: [asterisk-users] Start audio call and enable video later

I don't believe it's supported to start an audio call and then re-invite to add video, I believe Asterisk requires the call to start as a video call. On Mon, Jun 11, 2018 at 6:42 AM, Stefan Tichy wrote: > Hi, > > when some phone inititates an audio call and sends a re-invite with > audio and

Re: [asterisk-users] More testing

I got excited when I saw 8 new messages on the Asterisk list-serve this morning, What discussions must be happening I thought! You are a tease sir. On Tue, May 22, 2018 at 7:58 PM, Matt Fredrickson wrote: > More testing. Test test test. :-) > > -- > Matthew Fredrickson >

Re: [asterisk-users] Looking for better fax handling

Lock files. Create one when you start sending the fax, on your retry process check for a lock file and if one exists don't retry. On Mon, May 21, 2018 at 10:49 AM, D'Arcy Cain wrote: > I am having troubles with sending faxes. I hope someone can help me > work out a

Re: [asterisk-users] Wanted: WebRTC tutorial

On Tue, Apr 24, 2018 at 9:56 AM, Bruce Ferrell wrote: I'd REALLY like to get it working. And for the record, I REALLY HATE pjsip. > > I've been twiddling Asterisk (and other VOIP systems) since 2002; Linux > since '93 and telecom since 1980. The config is so opaque,

Re: [asterisk-users] Random, uncommanded blind transfers

I'd suggest doing packet captures on the T21P's themselves at the affected branches and see if you can catch it happening. The Yealinks themselves will regenerate DTMF if they get signaled for it. On Thu, Feb 8, 2018 at 2:19 AM, Stefan Viljoen wrote: > Hi Guys > > > >

Re: [asterisk-users] asterisk mysql contacts

use func_odbc, create a new function that does a lookup. [CALLERID] prefix=LOOKUP dsn=MyDB readsql=SELECT CALLERID from MyNames where CallerIdNum = '${SQL_ESC(${ARG1})}' exten => s,n,Set(CALLERID(NAME)=LOOKUP_CALLERID(${CALLERID(NUM)})) On Wed, Jan 17, 2018 at 6:16 AM, Atux Atux

Re: [asterisk-users] how do i enable call features??

+Function_DB On Wed, Jan 10, 2018 at 11:19 AM, Atux Atux <atuxn...@gmail.com> wrote: > That is the general idea. But how do i make it work? is there somewhere > ready? > > > On Wed, Jan 10, 2018 at 6:39 PM, John Kiniston <johnkinis...@gmail.com> > wrote: > >> Def

Re: [asterisk-users] how do i enable call features??

Define your *72 and *73 extensions in your internal context, Have them set a value in the ASTDB that you then check when dialing your handsets. The same can be done for call forwarding, store a number in the ASTDB and check if it's present, if it is forward the call to that number. On Wed, Jan

Re: [asterisk-users] How to supervise a Voicemail box with a BLF button ? What does "State:Unavailable" exactly means ?

eaves VoiceMailMan > application ? Using externnotify ? > 2. What is MWI:101@default expression for (see [2] ? > > Cheers > > [2] https://wiki.freepbx.org/display/FPG/Subscribe+a+BLF+ > button+to+Monitor+a+Voicemail+Box > > 2017-11-21 17:58 GMT+01:00 John Kiniston <john

Re: [asterisk-users] How to supervise a Voicemail box with a BLF button ? What does "State:Unavailable" exactly means ?

Hello Olivier, I may be incorrect but I don't believe you can hint on a mailbox like that. I've always used custom device states and dialplan logic for my shared voicemail boxes that are not being watched directly by a endpoint natively. On Mon, Nov 20, 2017 at 5:07 AM, Olivier

Re: [asterisk-users] Call preemption

The easiest thing to do might be to limit the peer to 11 calls in your sip.conf or pjsip.conf and then use the GROUP and GROUP_COUNT functions to limit the extension to 10 concurrent calls, Then when you get call #11 who is high priority you can allow it through or otherwise return busy.

[asterisk-users] Realtime extensions, Multiple SQL lookups?

I'm toying with the idea of replacing a statically generated file I include in my extensions.conf with a realtime lookup against my database. I've got it working but something seems off in my logs, It looks like I'm getting two lookups for every priority? [Oct 12 16:45:24]

Re: [asterisk-users] PJSIP, NAT and STUN/ICE

Well, I can answer one of your questions. To see the current external_media_address and external_signalling_address from the CLI you can issue a 'pjsip show transport ' Your router could have a built in sip helper that is rewriting the contact for your packets. On Mon, Oct 9, 2017 at 2:56 PM,

Re: [asterisk-users] Voicemail: search for name in a phonebook

Yes, You could do easily this either with the internal asterisk database or with something like func_odbc as a source for the data. In the context you receive your incoming calls you do a lookup against one of the above data sources using the CALLERID(NUM) and change CALLERID(NAME) to be the name

Re: [asterisk-users] SIP-Header Pass-Through

My only suggestion would be you could reduce your line count by replacing your GotoIf statements with ExecIF statements. exten => addheader,1,ExecIf($["x${ARG1}" != "x"]?Set(PJSIP_HEADER(add,Route)=${ARG1})) same => n,ExecIf($["x${ARG2}" != "x"]?Set(PJSIP_HEADER(add,P-Charging-Vector)=${ARG2}))

[asterisk-users] Pre-Dial Handler return something like GOSUB_RESULT?

I'm messing around with pre-dialer handlers today and running into a wall. Dial has the U option where I can execute a Gosub when the channels bridge and there I can set the variable GOSUB_RESULT to BUSY to make Dial act like the channel I called was Busy. I want to do something similar with a

Re: [asterisk-users] Writing CDR's to two database servers

You can do this with cdr_adaptive_odbc. On Tue, Jun 20, 2017 at 5:58 AM, Marcelo Terres wrote: > Well, you could create and AGI and run it after the normal CDR was > inserted. > Marcelo H. Terres > IM: mhter...@jabber.mundoopensource.com.br >

Re: [asterisk-users] Using queue priorities to add agents

I have a real ugly queue that has this in it's rules [mrule] penaltychange => 20,2,2 penaltychange => 40,3,3 penaltychange => 80,4,4 penaltychange => 120,5,5 penaltychange => 150,6,6 penaltychange => 180,1,1 penaltychange => 200,2,2 penaltychange => 220,3,3 penaltychange => 240,4,4 penaltychange

Re: [asterisk-users] Dial an extension to modify dialplan

You could use the DIALGROUP function for this and not need to shell out. https://wiki.asterisk.org/wiki/display/AST/Asterisk+14+Function_DIALGROUP On Mon, May 8, 2017 at 7:35 AM, Frank Vanoni wrote: > Hello > > I have the following scenario: > > [mynicecontext] >

Re: [asterisk-users] asterisk name in mysql

Well, My suggestion was to use FUNC_ODBC but instead you went with APP_MYSQL which has been depricated. Did you compile APP_MYSQL? It's not enabled by default. On Sat, Apr 22, 2017 at 1:25 PM, Atux Atux wrote: > Thanks a lot for the reply. > I did follow that already, but i

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] 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] 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

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] 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] 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] 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] 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] 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] 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] 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

[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] 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] 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=,

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] 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] 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] 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] 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] 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] 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 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] 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] 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

[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] 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

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] 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 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] 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] 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:

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

[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] 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] 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] 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] 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] 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] 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] 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] 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] 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

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