Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa
It worked with my test. I'm on Asterisk 18.19.0
-- Executing [517xxx@voipms:4] System("IAX2/voipms-15815", "asterisk
-rx 'core restart now'") in new stack
-- Remote UNIX connection
Asterisk uncleanly ending (0).
Executing last minute cleanups
== Destroying musiconhold processes
== Manager unregistered action DBGet
== Manager unregistered action DBGetTree
== Manager unregistered action DBPut
== Manager unregistered action DBDel
== Manager unregistered action DBDelTree
Preparing for Asterisk restart...
Asterisk is now restarting...
asterisk*CLI>
Disconnected from Asterisk server
Asterisk cleanly ending (0).
Executing last minute cleanups
After a hung call, can you run core restart now from the asterisk console?
Doug
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Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa
>> Is there a dial plan call that can "exit asterisk" or completely reload >> everything - killall active calls and start over ? Using system() you could issue a asterisk -rx 'core restart now' Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa
>>> How do we get this working For the time being, go back to 18.14.0 Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 18.14.0 vs 18.18.0 and chan_console/chan_alsa
In the old days when I was using console/dsp, I would have to use
alsamix from the console to verify that the output wasn't muted. Maybe
it's still the same.
Doug
On 9/7/23 03:43 PM, Jerry Geis wrote:
ok switching to "Console/default" does show the text
--- <("<) --- Call to device 'default' on console from 'default'
<2564286000> --- (>")> ---
--- <("<) --- Auto-answered --- (>")> ---
However I don't hear any audio.
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Re: [asterisk-users] asterisk 18.18.0 and chan_console
On 9/6/23 03:23 PM, Jerry Geis wrote: I am trying to just play on PulseAudio actually. This used to work - I have just recently updated to 18.18.0, so I'm puzzled. All of my Asterisk installs are running in virtual machines, so I have no way to test. Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 18.18.0 and chan_console
What is the device that you're connecting to? Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 18.18.0 and chan_console
In a past work life, I did use console/dsp to connect to a sound card that hooked up to a bogan paging amp. I still have access to the programming and everything I have show as using a lower case c for console Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 18.18.0 and chan_console
>>> hi Doug - so what device do you use? I am getting and error for Console/dsp I don't use it; just figured I'd try to help. Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 18.18.0 and chan_console
>>> Thanks doug - I did that - still showing XXX for chan_console Just to verify that you did rerun configure after installing the libraries? Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 18.18.0 and chan_console
On my debian 11 install I needed to install portaudio19-dev Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple phones on same PJSIP account
>>> I am creating a dialplan where a single user (Alice) has two offices. Both >>> of her phones should ring if her extension is called. On my home Asterisk, I have created a home queue and made both of my phones a member. The first phone that picks up get that call. Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Expanding my answering-machine system
On 6/17/23 08:47, Steve Matzura wrote:
Both Background() and WaitExten() allow the caller to enter DTMF
digits. Asterisk then attempts to find an extension in the current
context that matches the digits that the caller entered. If Asterisk
finds a match, it will send the call to that extension.
My question then is, is "*" a valid exension, as in:
I'd have to assume yes. I don't use WaitExten() and I set
autofallthrough=no in the /etc/asterisk.conf, since that is the way I've
always expected Asterisk to work; my dialplan examples are based on that.
The below example shows a call coming into a DID, playing background
prompts and excepting input during play.
;
;* Auto attendant
;
exten => 5175551212,1,Gosub(check-blacklist,s,1)
same => n,Gosub(check-hours,s,1)
same => n,Gosub(holiday-check,s,1)
same => n,Gosub(get-callerid,s,1)
same => n,Goto(auto-attend,s,1)
[auto-attend]
include => dial-by-extension
;*
;* Set timeouts
;*
exten => s,1,Set(TIMEOUT(response)=8)
same => n,Set(TIMEOUT(digit)=2)
same => n,Set(LOOPCOUNT=0)
same => n,GotoIf($["${Holiday}" = "YES"]?HOLIDAY:BEGIN)
same => n(BEGIN),Answer()
same => n,Wait(1)
;
;* Play the 'Welcome message' and office hours message
;
same => n,Background(${voice}/welcome)
same => n,Background(${voice}/business_hours)
same => n,Background(${voice}/8am_5pm)
same => n(HOLIDAY),Background(${voice}/dial_anytime)
same => n(DIRECTORY),Background(${voice}/directory_assist)
same => n,Background(${voice}/press_1)
same => n,Background(${voice}/to_ring_after_hours)
same => n,Background(${voice}/press_2)
same => n,Background(${voice}/absence_delay)
same => n,Background(${voice}/press_3)
;
;* If 1 is pressed, go to Dial by name
;
exten => 1,1,Goto(directory,s,1)
;***
;* If 2 is pressed, dial the Foyer phone
;***
exten => 2,1,Goto(dial-by-extension,4255,1)
;***
;* If 3 is pressed, dial absence/delay extension
;***
exten => 3,1,Gosub(cellphone-callerid,s,1)
exten => 3,n,Voicemail(3888@sip,us)
exten => 3,n,Hangup()
;
;* If 8# is pressed, go to Voicemail Main menu
;
exten => 8#,1,VoiceMailMain(@sip)
exten => 8#,2,Hangup()
This is not the complete dialplan; I also have error checking and a loop
counter.
Doug
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Re: [asterisk-users] Expanding my answering-machine system
On 6/16/23 20:29, Steve Matzura wrote: As always, thanks in advance for a kick in the right direction. For both capabilities, you can use Background() instead of Playback() for audio prompts. Background() allows for interrupting the prompts and continue on with your dialplan. Doug-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question on ring count on incoming circuits
On 5/28/23 14:20, Steve Matzura wrote: Who controls how many times an incoming call from an external (DID) provider will ring before Asterisk picks up the call and handles it internally Asterisk and this is defined with your timeout on the dial command, mine is 26 seconds so around 5 rings. Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Function DENOISE not registered
On 5/26/23 01:15, Fourhundred Thecat wrote: how do I fix this? What do I have to do to "register" denoise ? confbridge.conf states: "Requires func_speex to be built and installed." I am guessing you have not fulfilled that requirement. Doug-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems Solved, two left
On 5/24/23 09:56, Steve Matzura wrote: I don't understand your explanation because in the two files whose contents I posted, there's nothing routed to anything called just 's'. However, I've seen that in the error messages and it stumped me, too. No 'start' either. Steve, Please make sure you reply back to the list, so others can help also. As for why it's sending to the start extension, I cannot say since I am using IAX trunking with voip.ms and I get a DID for inbound matching. Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems Solved, two left
On 5/24/23 08:03, Steve Matzura wrote:
*** extensions.conf ***
[general]
[globals]
; Make sure to include inbound prior to outbound because the
_NXXNXX handler will match the incoming call and create a loop
include => voipms-inbound
include => voipms-outbound
[voipms-outbound]
exten => _1NXXNXX,1,Dial(PJSIP/${EXTEN}@voipms)
exten => _1NXXNXX,n,Hangup()
exten => _NXXNXX,1,Dial(PJSIP/1${EXTEN}@voipms)
exten => _NXXNXX,n,Hangup()
exten => _011.,1,Dial(PJSIP/${EXTEN}@voipms)
exten => _011.,n,Hangup()
exten => _00.,1,Dial(PJSIP/${EXTEN}@voipms)
exten => _00.,n,Hangup()
; inbound context example for your DID numbers, do not add the number
1 in front
[voipms-inbound]
exten => {redacted},1,Goto(hello,200,1) ; My DID
[phones]
exten => 101,1,Dial(PJSIP/yealink)
[hello]
exten => 200,1,Answer()
same => n,Playback(hello-world)
same => n,Hangup()
Your inbound is being sent to s (start extension) instead of your DID,
so it's not matching. So, you'll need to find out where in your
dialplan it's being mapped to s.
Did you know that voip.ms supports IAX2 natively? Working much better,
in my opinion, that SIP.
Doug
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Re: [asterisk-users] Problems Solved, two left
On 5/23/23 19:22, Steve Matzura wrote:
voipms: Call (UDP:208.100.60.12:5060) to extension 's' rejected
because extension not found in context 'voipms-inbound
Steve,
Could we see your dialplan for voipms-inbound?
I'm using voip.ms as well, but have not converted from chan_sip yet. My
voip-ms inbound extensions.conf below (Phone number changed to protect
the innocent)
[voipms]
include => voicemail
exten => 5175551212,1,Answer()
same => n,Gosub(check_blacklist,s,1)
same => n,Gosub(get_callerid,s,1)
same => n,Gosub(check_for_direct,s,1)
same => n,Set(_ARG1=4259)
same => n,Gosub(extension_timeouts,s,1(${ARG1}))
same => n,Queue(home,WwtTkKr,,,23)
same => n,NoOP(Dial Status: ${QUEUESTATUS})
same => n,NoOP(Hangup Cause: ${HANGUPCAUSE})
same => n,Gosub(s-${QUEUESTATUS},s,1(${ARG1}))
Doug
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Re: [asterisk-users] log custom variable in cdr
On 4/6/23 01:34, Fourhundred Thecat wrote: my question is, how can I log this filename in my cdr ? Set(CDR(userfield)=yourcontent) Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] mailing list working?
>>> there are new versions of Asterisk but mailing list is empty I think they've been having issues, I've noted recent mail coming across that was from several days ago. Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail Transcription with openai/whisper
On 11/27/22 09:22, Greg Troxel wrote: Thanks for posting. As I'm running asterisk on a PC Engines apu2, I don't need the details as it is obviously unworkable, but it's great to see non-cloud progress. Greg, Just a note, This would work if you have the API server running on a Linux x86 box. Then Asterisk would be using curl and python to communicate with that API Linux box. Doug-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voicemail Transcription with openai/whisper
Everybody, I've recently discovered openai/whisper and have been trying in earnest to get this working with Asterisk for voicemail transcriptions (Currently using the NerdVittles script with IBM Watson) https://github.com/openai/whisper After spending several hours today, I've successfully integrated my home Asterisk 16 voicemail with Whisper. Once I have followed the instructions for setting up an API server https://blog.deepgram.com/how-to-build-an-openai-whisper-api/ Initially, I setup a quad core VM to test this with, but discovered that without a dedicated card for the inference that it was horribly slow. So, I've set up testing on my desktop (Kubuntu 20) since I have an nVidia GTX 1060 installed. For the integration with Asterisk, I'm using a slightly modified script from nerdvittles IBM Watson script sendmailibm That can be found on their website https://nerdvittles.com/free-asterisk-voicemail-transcription-with-ibms-stt-engine/ I will probably find a low cost nVidia video card and get a stand alone Linux box running to handle this project. If you're interested in the details, let me know. Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to escape the & in BackGround
On 1/16/22 2:19 PM, Dovid Bender wrote: Does anyone know a way of telling Asterisk that & is part of the URL and to pass it along as a string? Try enclosing the URL in single quotes, Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk playback ogg files (SOLVED)
>>> asterisk doesn't support .ogg file format (digged through Yes it does, if it's complled in with it. Under make menuselect => Format Interpreters You'll see the development libraries that need to be installed before re-compiling for ogg playback support Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk playback ogg files
>>> exten => _[*+#0-9].,n,Playback(/var/lib/asterisk/sounds/output-ogg) If the actual filename is output.ogg then the code should be exten => _[*+#0-9].,n,Playback(/var/lib/asterisk/sounds/output) You'll also need to confirm that you compiled Asterisk with Vorbis support. Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk playback ogg files
>>> exten => _[*+#0-9].,n,Playback(/var/lib/asterisk/sounds/output.ogg,) Do not use the .ogg when describing the filename. Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Notifying missed calls
>>> but if the called hangs up prior the timeout for the voicemail, the >>> Subrouting "noanswer" will not called... You can use the h priority for that. https://wiki.asterisk.org/wiki/display/AST/Special+Dialplan+Extensions Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 18.7.1 - can't load res_fax, can't stop app_fax
>>> so I re-did make and make install and then a full asterisk restart, but >>> I still got the same "missing dependency: res_fax" error in the log. You should probably do a make distclean And then run configure again before re-compiling. Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco pricing and lead time
On 10/10/21 9:31 AM, Dovid Bender wrote: Hi, I see that you have pricing for the 12 C1000-48T-4X-L C I take it this is an ps moment *grin* Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] originate call in dial plan to join confbridge
>>> How do I do that ? I want all 3 ringing at the same time - and then as they >>> answer they are brought into the conference. I'd use call files, Others I'm sure would use AMI. Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Conference bridge recording file name
According to the wiki, you can disable the timestamp record_file_timestamp Append the start time to the record_file name so that it is unique. https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Configuration_app_confbridge Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Between a dumb client and a capable server...
>>> I do not want to build a SIP server / PBX myself which can itself perform >>> call hold >>> & transfer etc (I know how to do that with Asterisk) I assumed we were talking about an Asterisk server. Ignore what I just suggested, Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Between a dumb client and a capable server...
>>> I'm looking for something which I can place in the network path between the >>> client and the server, which can send these call control commands on to the >>> server, so that it can then put calls on hold, transfer them, etc. Install Flash Operator Panel https://www.fop2.com/ Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AST-2021-008: Remote crash when using IAX2 channel driver
>>> Asterisk Project Security Advisory - AST-2021-008 Downloading asterisk-16-current.tar.gz is still showing Asterisk 16.19.0 Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TON values
Mike, The below link turned up for me in a Google Search https://www.voip-info.org/asterisk-config-chandahdiconf/ Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] STIR/SHAKEN
On 3/7/21 1:43 PM, Greg Troxel wrote: So I wonder if your asterisk instance is connecting to the PSTN as a top-level carrier, or, more likely, I am confused in some way. Greg, I think this is the case for quite alot of those here. For me though, I just manage the on premise PBX and my carrier handles the STIR/SHAKEN part. Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ast_app_exec_macro: Cannot run 'Macro(atb)'. The application is not available.
>>> OK, both combination worked but still silence until the all numbers are >>> dialed. I have never used the U option on the dial command to call a sub-routine, Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ast_app_exec_macro: Cannot run 'Macro(atb)'. The application is not available.
>>> Gosub(check-number-forwarding,Dial(SIP/718xxpstn-5665,20,U(atb-sub))
>>> Is ARG1 = atb-sub ?
No.
My complete line
exten => _45XX,1,Set(_ARG1=${EXTEN}
same => n,Gosub(check-number-forwarding,s,1(${ARG1}))
So, if someone were to dial a 4 digit number starting with 45 (i.e. 4522), it
would jump to the sub-routine called check-number-forwarding and supply the
variable of 4522 to that sub-routine.
It could have been just as easily written as
same => n,Gosub(check-number-forwarding,s,1(4522))
Your sub-routine will need to pass what dialing options you are wanting to use.
A good source of information
https://wiki.asterisk.org/wiki/display/AST/Gosub
Doug
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Re: [asterisk-users] ast_app_exec_macro: Cannot run 'Macro(atb)'. The application is not available.
>>> How do you enable the phone speaker on the Gosub?
>>> I had:
>>> Dial(SIP/718x@pstn-5665,20,m(default)M(atb))
You can provide variables to your gosub routine, for an example
Gosub(check-number-forwarding,s,1(${ARG1}))
Doug
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Re: [asterisk-users] ast_app_exec_macro: Cannot run 'Macro(atb)'. The application is not available.
>>> app.c:280 ast_app_exec_macro: Cannot run 'Macro(atb)'. The application is >>> not available. Macros are no longer built by default in Asterisk 16. This was documented in the UPGRADE.txt file app_macro: - The app_macro module is now deprecated and by default it is no longer built. Users should migrate to app_stack (Gosub). A warning is logged the first time any Macro is used. Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk Unknown DYNAMIC_FEATURES item 'automon' on channel
Review your features.conf file in /etc/asterisk Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] I found an excellent guide: Configure Asterisk VoIP IP PBX SIP Server with Cisco IP Phones
>>> Also, you will need a TFTP server working on your Asterisk box My suggestion would be to get a refurbished Polycom VVX 301 phone (With power brick if no POE is avaiable) for around $27 US. Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Timing source for Asterisk
The wiki page has some information on timing and troubleshooting https://wiki.asterisk.org/wiki/display/AST/Timing+Interfaces Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] which linux for asterisk?
On 12/9/20 2:03 AM, Dmitry Melekhov wrote: But because Centos is declared dead, what is best choice ? Oracle? Ubuntu? And for those that have no idea as to what he is referring to (I didn't), here is the Register article https://www.theregister.com/2020/12/09/centos_red_hat/ Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk compile in VM move to actual hardware get illegal instruction
On 8/8/20 8:35 AM, Jerry Geis wrote: The VM is Intel box (host) and the physical box is a celeron. So something is not right there. What would be a good ./configure option that asterisk can compile with on the VM image so this illegal instruction does on occur ? Jerry, Under Compiler Flags uncheck BUILD_NATIVE Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Log queue threshold (1000) exceeded. Discarding new messages.
On 6/26/20 4:16 PM, Antony Stone wrote: Where can I set this threshold? /etc/asterisk/logger.conf ; All log messages go to a queue serviced by a single thread ; which does all the IO. This setting controls how big that ; queue can get (and therefore how much memory is allocated) ; before new messages are discarded. ; The default is 1000 Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ODBC connection failure - can it be fatal?
>>> Is there any way I can tell Asterisk that an ODBC connection problem is a >>> fatal error Your be best bet would be to do that check in the script that starts up Asterisk and maybe a CRON job that periodically tests connectivity. Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Controlling Asterisk from within the dialplan
>>> other than using the System() command? Not that I am aware of, Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mail2Fax
On 6/19/20 4:23 AM, basti wrote: Fax is not send. No Sip stuff is show in log. I don't know what is wrong here. Best regards Basti, This really belongs on the iaxmodem mailing list or the HylaFAX+ Mailing list. Lee Howard is the author of both packages and very responsive. Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR mysql: timeout when remote database unavailable
>>> Instead, the call still terminates if mysql cannot be reached. I just tested this, I'm using cdr_odbc, by shutting down mysql and I did not experience the call being dropped. The console logged the mysql failure, but the call continued. You may want to consider moving to cdr_odbc instead. Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call replicating
On 6/5/20 12:24 PM, Marek Greško wrote: How can this behavior been overriden? I do not expect this is problem on provider side, since it was working normally using chan_sip. Console output and dial plan snippets are always useful when diagnosing, Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Notification when on the phone
On 5/29/20 2:28 AM, Administrator wrote:
You could also use DEVICE_STATE
I am using DEVICE_STATE to identify when a phone is in use:
exten => s,n,GosubIf($["${DEVICE_STATE(SIP/${ARG1})}" =
"INUSE"]?SHOWBUSY,s,1(${ARG1}))
I'm trying to figure out the best way to display that information to the
person that is calling that in use extension.
Doug
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Re: [asterisk-users] Notification when on the phone
>>> But if you've already got the caller on the phone, then you might consider >>> the CONNECTEDLINE function in Asterisk... And that we don't. It's the third party that would like the notification the the destination phone is currently busy with another call. CONNECTEDLINE only functions after a channel has been answered. I was successful with using CONNECTEDLINE when issuing an Answer() first, but it added delay and the displayed message didn't show for very long. And, with the Polycom phones setup with multi-line, a call never rings busy unless the user press the DND (Do not disturb) button. Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Notification when on the phone
Everybody, I've had a request from my manager that I figure out how to get our Asterisk 13.x system using chan_sip to be able to display on the Polycom VVX series phone display (firmware 5.9.5), when an extension is called and the person on the other end is on the phone. He said, "Our old Analog phone system could do it, how hard can it be?" I've gone down the path of trying to use MessageSend, but for the life of me, cannot get the VVX 501 or VVX 601 phones to enable Instant messaging, Enabling the feature with feature.instantMessaging.enabled="1" seems to do nothing. Further investigation shows that I can send messages to the phones using curl after enabling Push messaging. This works easy enough, but figured I'd ask others if they are doing something similar and maybe I can avoid re-inventing the wheel. All comments are welcome! Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk : CDR Analyzer Updated
On 5/25/20 5:56 AM, Mitul Limbani wrote: Maybe you can have it uploaded on GitHub.com as a repository ? With a README.md file on how to install it for PHP7 ? Anybody that would like to do this would be most welcome. I have no plans on supporting it. Basic instructions and attachment will follow shortly, Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk : CDR Analyzer Updated
Everybody, I've been using the old Asterisk CDR Areski GUI CDR-Stats for at least a dozen years, it was easy to configure and didn't requite installing 'connectors' on anything or adding tables on the DB server. It's based off of PHP5 and the only reason I still keep around a Debian 7 system, since it won't work with the newer PHP7. A friend of mine is learning PHP7 and offered to update Asterisk Stats to work with the PHP7 as a learning experience. I've currently got the updated Asterisk Stats running on Debian 10 (Buster) without issues. Anybody wanting a copy, just reply to this email and I'll provide the updated archived install. Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Meaning of RTT in channelstats
On 5/16/20 9:57 AM, Michael Maier wrote: On 15.05.20 at 14:31 Doug Lytle wrote: Google says Round Trip Time https://www.voip-info.org/asterisk-rtcp/ That doesn't answer my question (I know the abbreviation RTT). Therefore I'm trying again: I'm just wondering what the RTT *exactly* means. Where are the exact measuring points located? => How are the RTT values exactly calculated? Which values are actually used for? => What about the processing time between the inbound leg and the outbound leg (transcoding, ...)? Somebody else more knowledgeable then me will have to chime in here, but my guess would be, that since TCP is stateful, it's the amount of time that a RTCP packet taken to be acknowledged the recipient. The measure points located would probably be each hop in the path, which typically can be visualized with traceroute. Similar to ping; the math behind it I would have no clue. And again, just a guess. Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Meaning of RTT in channelstats
Google says Round Trip Time https://www.voip-info.org/asterisk-rtcp/ Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mute conference participants
On 4/26/20 10:48 AM, Dovid Bender wrote: Hi, Looking at https://wiki.asterisk.org/wiki/display/AST/ConfBridge+Configuration there is an option for admin_toggle_mute_participants however the non admin users can still toggle toggle_mute. Is there any option for the admin to disallow non admins from using toggle_mute to unmute themselves? If there isn't such an option on there any devs here that can ping me off line what it would cost/take to get it done? Dovid, My guess would be to redefine their menu map and take away the option completely, Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Troubleshooting load issues
>>> All the calls are using ulaw. The files that I am playing are gsm. I >>> suppose doing a file convert with sox to .ulaw may help but it should be >>> able to do 500 calls without an issue. Can it possibly be a bug? if not how >>> do >>> I profile which call(s) can be causing the spike? One of the things that come to mind is that the operating system is flushing your SSDs at the time of the spike. You could always use iotop to watch what the file system is doing at the time of the spike. Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] audio problem with asterisk and meetme conference
>>> Can I adjust the talk or listen volume for another user? I've never used the volume controls, but it would appear. https://wiki.asterisk.org/wiki/display/AST/ConfBridge+Configuration Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] audio problem with asterisk and meetme conference
>>> I never moved to confbridge because they don't have an option for >>> controlling the volume of other >>> participants audio I have menu options in my confbridge configs that has increase and decrease conference volume. I'd still configure a small confbridge and test if you still have the issue, since meetme is no longer being developed. Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] audio problem with asterisk and meetme conference
>>> he problem is that there is some sort of distortion in the audio Has been been going on for a while or is this a new setup? Do you have a timing source? I bit the bullet around a year ago and moved to CONFBRIDGE; it wasn't as horrible as I thought it would be to setup. Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI not loading
>>> How do I do that? If you are using your package manager to install Asterisk & Dahdi, then I would not suggest that you compile. Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI not loading
>>> I saw something about needing to SIGN the dahdi modules. How do I do that ? >>> If that is the solution. Just a guess, Recompile Dadhi. Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No CID between Asterisk using IAX trunk
My Asterisk 13 IAX2 trunk posted below: type=friend trunk=yes allowcallerid=yes disallow=all allow=alaw allow=ulaw allow=gsm host=my.super.duper.host username=my.super.duper.username secret=my.super.duper.secret context=sip qualify=500 qualifysmoothing=yes requirecalltoken=no trunk=yes jitterbuffer=yes forcejitterbuffer=yes maxjitterbuffer=300 maxjitterinterps=100 resyncthreshold=1500 tos=ef cos=5 Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No CID between Asterisk using IAX trunk
>>> I am trying to troubleshoot two Asterisk servers that have an IAX2 >>> trunk between them. Carlos, Had caller-id ever worked between these two systems? Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] reviewboard.asterisk.org SSL Trust Failure
>>> Reviewboard is a legacy site and will likely be shutdown. Is there a reason >>> you are wanting to visit it? After seeing Olivier's post about his recent failures on compile and it referencing NBS (Network Broadcast Sound), which I had never heard of, I was googling to find out more and that was one of the Google hits Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] reviewboard.asterisk.org SSL Trust Failure
Under Firefox, browsing to https://reviewboard.asterisk.org I get Warning: Potential Security Risk Ahead Firefox detected a potential security threat and did not continue to reviewboard.asterisk.org. If you visit this site, attackers could try to steal information like your passwords, emails, or credit card details. Websites prove their identity via certificates, which are issued by certificate authorities. Most browsers no longer trust certificates issued by GeoTrust, RapidSSL, Symantec, Thawte, and VeriSign. reviewboard.asterisk.org uses a certificate from one of these authorities and so the website’s identity cannot be proven. I see that the cert is signed by RapidSSL Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call from an extension
I understood that part, I was hoping to understand why. In the past, I've used the PSTN lines to connect two Asterisk systems for extension to extension calls and was able to route source and destination extensions via the dial-plan, just by parsing the assigned CID. Was thinking that may be what you were also trying to accomplish. Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call from an extension
>>> I desire to make a call from my system looking like it comes from 4452 and >>> call the outside number If you have control over your CID with your provider, you can use Set(CALLERID(number)=4452) Otherwise, you cannot. If you would provide us with what you are trying to accomplish, maybe we can give you some options. Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call from an extension
>>> I can make calls over a SIP trunk as SIP//number
>>> I am trying to make calls over an extension thought using the same format
>>> SIP/4452/number - its not working
No,
Extension to extension calls would be:
Dial(SIP/${EXTEN])
My extension to extension dial line is
exten => s,n,Dial(SIP/${ARG1},${timeout},${dial.opts})
I'm currently still on chan_sip,
Doug
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Re: [asterisk-users] From the CLI, how can I hangup a channel name that includes a space character?
>>> Is there some control character(s) for the CLI to interpret everything in >>> between as a single argument? I think you can typically use tab completion when working with spaces or you can escape the space with a back slash For example Doug Lytle would be Doug\ Lytle Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] res_calendar & LetsEncrypt
On 12/24/19 10:34 AM, Sean Bright wrote: On 12/24/2019 9:02 AM, Doug Lytle wrote: [Dec 24 07:48:46] WARNING[10679] res_calendar_caldav.c: Unknown response to CalDAV calendar calendar.name.here, request REPORT to /dav/username/Calendar: Server certificate changed: connection intercepted? Would this be considered a bug, or do I have something setup incorrectly? This error message comes from neon and was removed in r1938 back in 2014[1]: src/ne_openssl.c (ne__negotiate_ssl): Don't fail hard for SSL cert change, invoke verify callback. For better or worse, Asterisk's verify callback allows all certificates, so this doesn't appear to be an Asterisk bug. You should probably try to find a newer version of neon for your distribution. Thanks guys for the input! Just another reason to upgrade that to Debian Buster. Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] res_calendar & LetsEncrypt
Everybody, For a while now, I've had a small home Asterisk setup to connect to my Zimbra mail server's calendar. Making an entry on the calendar would cause Asterisk to schedule a wakeup call at the time of the calendar entry. The Zimbra mail server uses LetsEncrypt for the SSL Certs and renews every 60 days. On the Asterisk side of things, if I do not restart the Asterisk process, the logs get spammed with the below and the wakeup call never occurs: [Dec 24 07:48:46] WARNING[10679] res_calendar_caldav.c: Unknown response to CalDAV calendar calendar.name.here, request REPORT to /dav/username/Calendar: Server certificate changed: connection intercepted? Would this be considered a bug, or do I have something setup incorrectly? Asterisk version: 13.29.2 OS: Debian GNU/Linux 7.11 (wheezy) Zimbra OSE 8.8.11 P4 Thanks! Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Block Spam Calls
On 12/13/19 11:48 AM, Julian Beach wrote:
Hello Doug,
Friday, December 13, 2019, 11:03:37 AM, you wrote:
This is exactly what I do - “press 1 for a human”
Works great
I do this as well, but I also do a database lookup to see if the number
is on our speeddial list and if so, pass the call directly on without
the IVR prompts.
For those that would like to see my code:
exten => 517xxx,1,Answer()
same => n,Gosub(check_blacklist,s,1)
same => n,Gosub(get_callerid,s,1)
same => n,Gosub(check_for_direct,s,1)
same => n,Set(CHANNEL(musicclass)=music)
same => n,Gosub(extension_timeouts,s,1)
same => n,Dial(SIP/3501,${timeout.timeout},TtKk)
same => n,NoOP(Dial Status: ${DIALSTATUS})
same => n,NoOP(Hangup Cause: ${HANGUPCAUSE})
same => n,Gosub(s-${DIALSTATUS},s,1)
[check_for_direct]
;**
;* Check if there is a match of the inbound call to the speed dial list
;* If not, make then go through the IVR menu
;***
exten =>
s,1,Set(ARRAY(speed.phone,speed.name)=${ODBC_MENU_DIRECT(drdos,${CALLERID(number)})})
;
;* If the contents of speed.phone is blank, assume that it
;* is not programmed and force the call to use the IVR to
;* prove they are not an automated call.
;
same => n,GotoIf($["${speed.phone}" != "" ]?3:ivr_menu,s,1)
same => n,NoOP(${speed.name} is on the approved list)
same => n,Return()
same => n,Hangup()
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Re: [asterisk-users] Block Spam Calls
On 12/12/19 6:55 PM, Adam Goldberg wrote: This is exactly what I do - “press 1 for a human” Works great I do this as well, but I also do a database lookup to see if the number is on our speeddial list and if so, pass the call directly on without the IVR prompts. Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Simple, fast single-word offline free speech recognition in Asterisk (or as an AGI)?
On 11/26/19 12:31 AM, Jonathan H wrote: Yes, I know I post similar back in January, but there was no response back then and I was hoping things might have changed :) I'm using IBM's Watson for voicemail transcriptions, they allow 500 minutes per month for speech to text on the Free/Lite plan. Maybe that could be used for a solution for you too. http://nerdvittles.com/?p=21703 Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] email notification on missed call
On 10/30/19 12:10 AM, Fourhundred Thecat wrote: Does asterisk not have some internal function to send email ? It does so for voicemail. Is there perhaps a better way to this than described above ? As far as I am aware, Asterisk has no built-in dialplan function to allow sending of email. The way that your currently programming this is the typical way that I would handle it. Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] clarification on gosub, macros and AEL
>>> Nobody has any information or opinions on any of this? Personally, I don't think MACROS are going anywhere any time soon, so I have not bothered looking into a substitution. As for ael; I've never used it. Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] setting up ODBC for cdr logging into MariaDB
On 10/12/19 8:15 AM, Fourhundred Thecat wrote: did you compile libmyodbc yourself ? No, If I recall correctly, after a lot of searching, I ran into the apt source below and created the myodbc.list and put it into /etc/apt/sources.list.d cat myodbc.list deb http://ftp.de.debian.org/debian jessie main I just ignored the complaints about not having a GPG key. Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] setting up ODBC for cdr logging into MariaDB
On 10/11/19 10:12 PM, Fourhundred Thecat wrote: Hello, I am trying to set up cdr logging into MariaDB through ODBC. I have installed unixodbc unixodbc-dev and now I am struggling with configuring /etc/odbcinst.ini All the examples online use non-existent libraries, ie: On my Debian Buster I have: dpkg -l|grep odbc ii libmyodbc:amd64 5.1.10-3 amd64 the MySQL ODBC driver ii libodbc1:amd64 2.3.6-0.1 amd64 ODBC library for Unix ii odbcinst 2.3.6-0.1 amd64 Helper program for accessing odbc ini files ii odbcinst1debian2:amd64 2.3.6-0.1 amd64 Support library for accessing odbc ini files ii unixodbc 2.3.6-0.1 amd64 Basic ODBC tools ii unixodbc-dev:amd64 2.3.6-0.1 amd64 ODBC libraries for UNIX (development files) cat /etc/odbcinst.ini [PostgreSQL] Description = ODBC for PostgreSQL Driver = /usr/lib/libodbcpsql.so.1 Setup = /usr/lib/libodbcpsqlS.so.1 FileUsage = 1 [MySQL] Description = ODBC for MySQL Driver = /usr/lib/x86_64-linux-gnu/odbc/libmyodbc.so Setup = /usr/lib/x86_64-linux-gnu/odbc/libodbcmyS.so FileUsage = 1 Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Amazon AWS question
Dan, I don't run Asterisk on AWS, but I do on ESXi. Are you running a version of Asterisk before 13? Newer versions Asterisk handle timing better that don't require a hardware timing source. I'm running Asterisk 13 on a small 60 phone system without issues under ESXi 6.0 Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Lightweight ODBC DB
On 8/1/19 5:08 PM, Dovid Bender wrote: Glenn, I can't use MySQL as each node currently has MySQL however there is a lot of data that is stored locally on each box. I may have to take this route if I can't find something else but that would mean syncing all sorts of data that does not need to be synced. If I recall correctly, you can exclude databases. Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] svnview.digium.com down?
>>> I have updated the wiki. The script can be found within the >>> contrib/scripts/sip_to_pjsip subdirectory of an unpacked download of >>> Asterisk 13 and forward. Got it! Thanks, Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] svnview.digium.com down?
I'm currently reviewing the Digium wiki on migrating from chan_sip to res_pjip and I'm trying to access the script that is provided to help with conversion. https://wiki.asterisk.org/wiki/display/AST/Migrating+from+chan_sip+to+res_pjsip It would appear that said server hosting the script is no responding or the link is no longer valid. Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Better audio in than just 8k
Maybe streaming will be helpful, https://www.agix.com.au/streaming-internet-music-for-asterisk-10-on-hold-music/ Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and pulseaudio Console/dsp
>>> I setup and extension to connect me with Console/Dsp. I am hearing the >>> audio but its warbly or does not sound right. Any thoughts on what I need >>> to do for that ? I had that issue at a previous employer and got around it by using ALSA instead. Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Linphone
My self-compiled Asterisk also shows that speex dependencies are not installed Speex Coder/Decoder Depends on: speex(E), speex_preprocess(E) Can use: speexdsp(E) You'll need to installed the dependencies and re-compile. Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] unsolved: Re: solved: how to create a working certificate for using TLS?
On 7/4/19 6:40 PM, hw wrote: This has again, and for no reason, ceased to work again after restarting asterisk. No matter what I try, I can't create a certificate asterisk would verify. Have you considered using LetsEncrypt for a valid certificate? Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Spectrum SIP trunks
>>> We've recently replaced an old Meridian phone system (Analog) with Asterisk >>> and signed up for Spectrum SIP trunks. Should have included that we're running Asterisk 13, under chan_sip Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Spectrum SIP trunks
We've recently replaced an old Meridian phone system (Analog) with Asterisk and signed up for Spectrum SIP trunks. The service gets installed on July 8th and I was hoping somebody that may have already gone through the process could give me some hints. I've only ever dealt with PRIs or IAX2 trunks when it came to Asterisk and this will be my first SIP trunk. They installed the Adtran fibre box yesterday. (We are in Michigan) Has anybody already setup a Spectrum SIP trunk? If so, could you provide me some input? Google provided the suggested setup: ;[spectrum] ;host=IP Address of Adtran ;type=peer ;disallow=all ;allow=ulaw ;allow=alaw ;context=spectrum ;trunk=yes ;insecure=port,invite ;qualify=500 ;qualifysmoothing=yes ;jitterbuffer=yes ;forcejitterbuffer=yes ;maxjitterbuffer=300 ;maxjitterinterps=100 ;resyncthreshold=1500 All comments or suggestions are welcome, Thanks! Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 302 moved temporally callerid behavior
core show version Asterisk 13.26.0 built by doug @ asterisk on a x86_64 running Linux on 2019-04-05 11:41:43 UTC Built from source, Douh -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 302 moved temporally callerid behavior
>>> Surely that is "call forwarding", which is quite different from either a >>> blind or attended transfer? That would be correct. The forward button on the polycom phones just do a redirect to the destination extension or external phone number. Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 302 moved temporally callerid behavior
We have Polycom phones (I'm using a VVX601, the destination is a VVX301). We're also on Asterisk 13. I forwarded my call to the VVX301 and then dialed my phones DID. The forwarded call showed my cell phone number, so I cannot reproduce. Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Res_Srtp
On 3/31/19 8:21 AM, Gokan Atmaca wrote: Hello The "res_srtp" module does not appear. How do I install it? Are you compiling or installing from packages? If compiling, you'll need to install the development library. Under Debian it is libsrtp0-dev Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Paging systems?
>>> Does anyone have an (overhead) paging system that they like that works with >>> SIP? Our old phone system back ends into a Bogen AMP. I'm in the process of replacing that system (Meridian) with Asterisk and found that the snom PA1 works very well. If an AMP is involved, this might work. http://wiki.snom.com/File:Snom_pa1.png Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IVR Loop
Your IVR should only play audio prompts and only attempt to dial once a selection has been made, Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] internal call record
On 3/9/19 9:56 AM, Gokan Atmaca wrote: a) work for recording incoming / outgoing calls b) do not work for recording internal calls then we might be able to give you a clue what's wrong. Hello For example: My phone number is 1000, the other's number is 1001. These numbers are in the same PBX (asterisk). I want 1000, 1001 Gokan, Since you've said that outside calls can be recorded, but not inside calls; Antony requested that you show us your dialplan code for recordings that work. This will give us an idea of what might be going wrong when trying to record inside calls. It would also be helpful to see your console output when things are not working. Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 16.2.1 inbound route
On 3/5/19 2:46 AM, Gokan Atmaca wrote:
Asterisk can send calls, but I don't get a call. What could be the problem?
[from-siptrunk]
exten => 13XXX,1,dial(${OPERATOR},20)
You are trying to match a pattern, so this needs to be
exten => _13XXX,1,dial(${OPERATOR},20)
Doug
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