Re: [asterisk-users] Voicemail notification by email is missing CallerID info

This is what comes with voicemail.conf.sample - works for me! ; Change the from, body and/or subject, variables: ; VM_NAME, VM_DUR, VM_MSGNUM, VM_MAILBOX, VM_CALLERID, VM_CIDNUM, ; VM_CIDNAME, VM_DATE ; Additionally, on forwarded messages, you have the variables: ; ORIG_VM_CALLERID,

Re: [asterisk-users] Soft SIP phones that support TLS - Asterisk version 13.13.1

Microsip (Windows) is free and small. 2.5Mb download, 10Mb RAM usage, does everything I need and configuring TLS is a doddle. http://www.microsip.org/ On 16 February 2017 at 13:04, Max Grobecker wrote: > Hello, > > I'm a big fan of PhonerLite. > It's more poplar

[asterisk-users] 14.3.0 download archive corrupt - cannot extract

Hi there; 2 linux boxes and Windows all report an error and the archive is not extractable. Wget reports the size as follows: 2017-02-14 08:36:21 (7.29 MB/s) - ‘asterisk-14-current.tar.gz’ saved [40653605/40653605] It starts un-tarring but then asterisk-14.3.0/bridges/bridge_native_rtp.c

Re: [asterisk-users] bash: asterisk: command not found

Did you actually do "make install" after doing "make"? On 7 December 2016 at 12:17, Tzafrir Cohen wrote: > On Wed, Dec 07, 2016 at 09:23:30AM +, k...@mayten.sch.bme.hu wrote: >> On 2016-12-07 09:13, Steve Howes wrote: >> >On 07/12/16 04:56, christopher kamutumwa

Re: [asterisk-users] Asterisk 14.2 CLI don't show debug/verbose data

I think it might be related to this? https://issues.asterisk.org/jira/browse/ASTERISK-26391 I think I remember having to edit logger.conf - this is what mine looks like now: console => notice,warning,error messages => notice,warning,error Try that, restart asterisk and see if it works :) On 30

Re: [asterisk-users] Any reason Asterisk won't start without a rebuild on a cloned VPS?

Thanks for the super-quick answer! Now I was able to find this: https://wiki.asterisk.org/wiki/display/AST/Building+and+Installing+Asterisk#BuildingandInstallingAsterisk-Buildingfornon-nativearchitectures I had just assumed a cloned vps would be identical. Out of interest, how unoptimized would

[asterisk-users] Any reason Asterisk won't start without a rebuild on a cloned VPS?

Any ideas why a VPS, cloned from another instance (DigitalOcean "droplets" if it matters), won't run on the new instance? Everything else is the same; region, memory, disk, hypervisor version etc. And everything else runs, just not Asterisk, unless I do a make distclean in the /usr/src/asterisk

Re: [asterisk-users] Non-global variable that follows channel?

Thanks. I did a while ago, but I couldn't make it "fit" what I wanted to do. Maybe with my increase Asterisk knowledge now I'll take another look. Thanks! On 27 November 2016 at 18:27, Richard Mudgett wrote: > Have you looked into ARI [1]? I think it would be a closer fit

Re: [asterisk-users] Non-global variable that follows channel?

Thanks, Richard - your code does indeed work reliably 100% of the time, and thank you for that explanation. I do think the docs at https://wiki.asterisk.org/wiki/display/AST/Asterisk+14+Function_SHARED could do with more clarification. BTW, there were a couple of typos in your code, so for

Re: [asterisk-users] Non-global variable that follows channel?

r you, you might use the SHARED() variables which > are kind of global accessible by the channel ID. > So you might call your Gosub with only the (unique) reference name of the > variables you wish to pass and then call it from your Gosub. > -> https://wiki.asterisk.org/wiki/display/AST/A

Re: [asterisk-users] ODBC locks warning in CLI - Asterisk 1.8.32.3

It might be worth pointing out that 1.8x was released 6 years ago, went into security fix only over 2 years ago, and reached "end of life/no further fixes" over a year ago. 11.x went into "security fix only" last month - 13 and 14 are the current versions - can you try with them? On 23 November

[asterisk-users] Non-global variable that follows channel?

Related to http://lists.digium.com/pipermail/asterisk-users/2016-November/290384.html, at the moment I'm passing one variable via DIAL. Now I'd like to pass a whole bunch, and my idea was to rather than having a great string of b(synctest3b^setVar^1(something)^2(more things)^3(etc)) and then

[asterisk-users] Anyway of simulating "hold" so that the moh announcement function works?

In the musiconhold.conf example, it says: announcement=queue-thankyou ;If this option is set for a class, then when callers get put on hold, the specified sound will be be played to them. I'm using the "m" option in Dial and was hoping to make use of this feature. Any dialplan way of getting

Re: [asterisk-users] Synchronous dialplan execution for feedback while processing speech recognition and voice synth, for example.

xten => setVar,1,Set(testVar=${ARG1}) ; setter same => n,Return() On 3 October 2016 at 23:48, Steve Edwards <asterisk@sedwards.com> wrote: > On Mon, 3 Oct 2016, Jonathan H wrote: > >> I've googled and I'm probably missing something pretty newbie 101 here, >

[asterisk-users] Any way of just killing ALL stray WHILE loops?

tl;dr Is there ANY way/hack of just telling Asterisk to destroy *all* WHILE loops it may be nested in at a certain time? Reason: you know the thing about WHILE loops not only having to have "seen" their endwhile to finish properly? If not, a reminder before it gives you 3am sleepless nights:

Re: [asterisk-users] What could be stopping "Disconnect Call" feature from working (set in features.txt)

. Thank you again. On 9 November 2016 at 12:32, Tony Mountifield <t...@softins.co.uk> wrote: > In article > <caeebynvuscicqvfyspvsgpnapbbubn_67txjsk5j7gm42+o...@mail.gmail.com>, > Jonathan H <lardconce...@gmail.com> wrote: >> Thank you - that makes sense. I

Re: [asterisk-users] What could be stopping "Disconnect Call" feature from working (set in features.txt)

y to do those to things, is there any way to force Asterisk to NOT "optimize" those channels? On 9 November 2016 at 00:09, Richard Mudgett <rmudg...@digium.com> wrote: > > > On Tue, Nov 8, 2016 at 5:19 PM, Jonathan H <lardconce...@gmail.com> wrote: >> >> Asterisk

[asterisk-users] What could be stopping "Disconnect Call" feature from working (set in features.txt)

Asterisk 14.1 Here's a bit of test dialplan, which works as expected and simulates exactly what I'm doing at the top of my large dialplan... [dial-pre-test] exten => s,1,NoOp() same => n,Set(LIMIT_PLAYAUDIO_CALLER=yes) same => n,Set(LIMIT_WARNING_FILE=time_limit_reached) same =>

[asterisk-users] en_GB extra sounds don't have ha and wx split out like en and fr do.

The extra sound packages for en and fr have ha (home automation) and wx (weather) broken out into seperate directories, but en_GB doesn't, although the files seems to exist in the main extra folder. There's no open ticket about this; I can just "ls" the wx and ha dirs to a text file and make a

Re: [asterisk-users] Suddenly getting lots of "Unable to send packet: Address Family mismatch between source/destination" but ONLY on 1 of 2 VPSs in same datacentre.

Just to say "thank you" on the list, and to confirm that the output of the command you suggested are as follows: # ip -6 addr show dev eth0 2: eth0: mtu 1500 state UP qlen 1000 inet6 fe80::601:ddff:fea2:dXX1/64 scope link valid_lft forever

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

erisk tips/gotchas! On 4 November 2016 at 23:02, John Kiniston <johnkinis...@gmail.com> wrote: > Could it be SELinux blocking you? > > If you change the path to /tmp does it work? > > > On Fri, Nov 4, 2016 at 3:14 PM, Jonathan H <lardconce...@gmail.com> wrote: >> >

[asterisk-users] Is this a reasonable way to store user prefs in ASTDB? And what's this Re: Is JSON a dialplan "thing" yet? (Asterisk 14)

(1,Current item is ${hashKey}:${HASH(userPrefs,${hashKey})}) same => n,EndWhile same => n,Verbose(1,Setting ${prefPairs} to DB) same => n,Set(DB(userPrefs/${CALLERID(num)})=${prefPairs}) same => n,Hangup() On 1 November 2016 at 23:29, Joshua Colp <jc...@digium.com> wrote: > &

Re: [asterisk-users] Suddenly getting lots of "Unable to send packet: Address Family mismatch between source/destination" but ONLY on 1 of 2 VPSs in same datacentre.

of course, the VPS host is set to V6 disabled. and as far as I am aware, and my ITSP doesn't have IPv6, so I just can't figure out why two IPv4 systems are getting IPv6 "pollution" as it were. And why now??! Anyway, that's what fixed it for me. Thanks! On 4 November 2016 at 21:31,

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

u)=Extension: s) > > On Fri, Nov 4, 2016 at 2:26 PM, Jonathan H <lardconce...@gmail.com> 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, without reverting to

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

2016 at 21:32, John Covici <cov...@ccs.covici.com> wrote: > Won't the system command do it? > > On Fri, 04 Nov 2016 17:26:13 -0400, > Jonathan H wrote: >> >> Seems I can write to an existing file, but is there really no way of >> creating a new file to log so

[asterisk-users] Suddenly getting lots of "Unable to send packet: Address Family mismatch between source/destination" but ONLY on 1 of 2 VPSs in same datacentre.

Two VPSs. Identical setups with the exception of the extension. Same version of everything, Asterisk 14.1, Ubuntu 16.10, same firewall rules and so on - box 2 was cloned from box 1. Both VPSs run in the same datacentre. Suddenly, after weeks of OK, I'm getting lots of this on ONE box only:

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

Seems I can write to an existing file, but is there really no way of creating a new file to log some data to, without reverting to AGI? (will be different for each caller ID) -- _ -- Bandwidth and Colocation Provided by

[asterisk-users] What's the smallest, lightest Asterisk you can build? Does size even matter?

All I need is PJSIP, ulaw, alaw, wav, astdb and all the dialplan functions. I don't need any other DB layer, I have no hardware, and I was wondering what the smallest build possible was. I experimented, but everything relied on other things. And then I wondered... is there actually any point? Is

[asterisk-users] Is JSON a dialplan "thing" yet? (Asterisk 14)

I need to store some basic caller data in ASTDB - certainly doesn't need full blown mySQL. There's 4 or 5 bits of info per caller, and I saw that there is a json entry in ASTDB for the endpoint. Does that mean that there are accessible functions to deal with json now? I couldn't find anything in

[asterisk-users] Variable pollution? Stack overflow? WTF is going on here or... how can I TOTALLY clear a variable?

The first time I run a loop, the AGI returns a list of files. I append a path from a variable, and play out the files using SHIFT to loop them. The FIRST time I enter the system, this is what the complete list to be looped looks like:

[asterisk-users] Got bitten by the 255 char variable limit - how best to work around it?

I loop through a list in Asterisk which is generated by a Python AGI and I've just been bitten by a variable limit I didn't realise existed before. The only way I can think of working around this is to get Python to write the list to file, then do a FILE_COUNT_LINE to get the number of items

Re: [asterisk-users] Problem with REMAINDER? 957%60 be 15 remainder 57 not 15 remainder -3 ?

Yes! That's the one. Thank you. That's a good workaround. The following test dialplan shows the bug (feature?) exten => 7,1,Verbose(Context: ${CONTEXT} Exten:${EXTEN}) same => n,Set(seconds=57) same => n,While($[${seconds} <= 400]); same => n,Set(minutes=$[FLOOR(${seconds} / 60)])

[asterisk-users] Is it possible that variables returned from AGI take a moment to "stick"?

I thought dialplan flow was that (normal!) agi was called, it did its thing (which include returning some dialplan variables/lists), and then when agi finished it returned to the dialplan which then reliably carried the product of agi. But I'm calling agi, scanning a path in python, and then

[asterisk-users] Problem with REMAINDER? 957%60 be 15 remainder 57 not 15 remainder -3 ?

I'm not mathematically gifted, but shouldn't 957%60 be 15 remainder 57? Google and my desktop calculator certainly think so. So where am I going wrong here? The following code exten => 7,1,Verbose(Context: ${CONTEXT} Exten:${EXTEN}) same => n,Set(myNum=957) same =>

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

turn() On 18 October 2016 at 17:56, John Kiniston <johnkinis...@gmail.com> wrote: > 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

[asterisk-users] Say duration of alaw file?

I can get the size of a ulaw file using STAT. And I can get the duration in seconds by doing filesize/8000. Your tea-break challenge is to help me find the shortest most Asterisk-like way of saying: "The following file is [ minutes and] seconds long". ...without referring to external

Re: [asterisk-users] Multiple readfile oddities, newlines etc

I'm going to go ahead and file a bug report, 'cos something definitely ain't right here! Bug filed: https://issues.asterisk.org/jira/browse/ASTERISK-26481 This bit of dialplan. exten => 5,1,Verbose(Context: ${CONTEXT} Exten:${EXTEN}) same => n,Set(featurefile=/home/test/feature-1.txt)

Re: [asterisk-users] Surfing the web via Asterisk.

On 17 October 2016 at 16:12, Matt Riddell wrote: > https://www.npmjs.com/package/speech-rule-engine Thanks. That and the tip about jackaudio look interesting, although that thing above is just a parser, not a renderer. I think, at this stage, it's an idea to go back in

[asterisk-users] Tips, tools and a question about debug level....

Lots of little bits in one email to save polluting the list too much today, time for me to try and give a little back, too! Someone posted about sngrep a couple of days ago. What a great tool! Is there a list of useful stuff like this that isn't hopelessly out of date? Talking about hopelessly

Re: [asterisk-users] Surfing the web via Asterisk.

te the object you create > and tts to describe current position. The hard part will be parsing the HTML > even though most HTML is broken :-) > > Kind regards, > > Matt Riddell > > On Oct 17, 2016, at 9:00 AM, Jonathan H <lardconce...@gmail.com> wrote: > > Has any

Re: [asterisk-users] Multiple readfile oddities, newlines etc

(feature2)} long) same => n,Set(unfilteredfeat=${FILE(${featurefile},0,1,l,u)}) same => n,Set(feature3=${SHIFT(unfilteredfeat)}) same => n,Verbose(1,Using a string with shift method, feature3 is set to ---${feature3}--- and is ${LEN(feature3)} long) Bug or... "feature"?

[asterisk-users] Surfing the web via Asterisk.

Has anyone attempted making the web phone accessible? I can only find one company which operated between 1996 and 2000. I was thinking, install Chrome with Chromevox, headless, on a server, and use something like an AGI to send basic keyboard commands to navigate a page, as a screenreader user

[asterisk-users] Multiple readfile oddities, newlines etc

I have a plain text file, ASCII, unix line breaks. 1 single line, and all that is in it is the word "radio". Here's some test dialplan: exten => 5,1,Verbose(Context: ${CONTEXT} Exten:${EXTEN}) same => n,Set(feature=${FILE(/home/test/feature-1.txt,0,1,l,u)}) same => n,Verbose(${feature})

Re: [asterisk-users] Registered successfully, but after a minute or so no SIP messages anymore

Hmmm, sorry, I can't think of anything except... why do you need the STUN server? And are you sure that all the ports in your router definitely match the ones Asterisk thinks it's using? Then there is always the SIP-ALG problem with some routers, which some people have been able to overcome by

Re: [asterisk-users] Registered successfully, but after a minute or so no SIP messages anymore

All other things aside, this stands out immediately: RTT: 434.393 msec That's almost half a second round trip for a packet. I'm amazed anything works at all. For SIP connections, mine are usually about 26ms max, anything above about 35 is bad. Looks like a serious config issue. Try pinging and

Re: [asterisk-users] Match one OR two digit extension not working as expected without using "dangerous" _. pattern (Ast 14)

On 13 October 2016 at 13:18, Tony Mountifield wrote: > exten => _X,1,NoOp(Matching single digit) > exten => _X.,1,NoOp(Matching multiple digits) > exten => _X!,2,SayNumber(${EXTEN}) > exten => _X!,3,Etc.. Thanks - I appreciate the idea, but it matches more than 2 digits.

Re: [asterisk-users] Match one OR two digit extension not working as expected without using "dangerous" _. pattern (Ast 14)

> > Best regards > > Jean Aunis > > > Le 13/10/2016 à 12:54, Jonathan H a écrit : >> >> Back to basics here. I want to match on one OR two digits. >> >> The following two both work, but ONLY for more than one digit, which >> is not as expect

[asterisk-users] Match one OR two digit extension not working as expected without using "dangerous" _. pattern (Ast 14)

Back to basics here. I want to match on one OR two digits. The following two both work, but ONLY for more than one digit, which is not as expected from the docs (see below). exten => _X.,1,SayNumber(${EXTEN}) exten => _[0-9].,1,SayNumber(${EXTEN}) This next one will ONLY match 2 digits, as

Re: [asterisk-users] Asterisk 12 error when installing

Are those numbers correct? Asterisk 12 stopped being supported almost 2 years ago and became "do not use" on 2015-12-20 https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions Ubuntu 14 may still be supported, if you're on 14.0.4.5 https://wiki.ubuntu.com/Releases You could try make

Re: [asterisk-users] AGI: How to break out of AGI when stream_file escape_digits are detected in middle of long sequence of files?

and the following code. Thank you again. escape_digits = str("*") pressed_digit=agi.stream_file(promptFile,escape_digits) if pressed_digit == "*": sys.exit(0) On 11 October 2016 at 09:31, Lefteris Zafiris <z...@fastmail.com> wrote: > On Mon, 10 Oct 2016, at 22:47, Jonathan

[asterisk-users] AGI: How to break out of AGI when stream_file escape_digits are detected in middle of long sequence of files?

For reasons best known to myself, I call a python agi (PYST2 - love it!) which streams a series of very short files in quick succession. Like this: escape_digits = str("0") agi.stream_file(promptFile,escape_digits) and this is what I see on the AGI debug: AGI Tx >> 200 result=0 endpos=6784 AGI

Re: [asterisk-users] Asterisk 14.0.2 Now Available

Just a minor thing: on http://www.asterisk.org/downloads/asterisk/all-asterisk-versions it still reports 14.0.1 as being the latest version, although the download itself contains 14.0.2 I'd have file a bug but there doesn't seem to be a "website" section in the tracker. On 30 September 2016 at

[asterisk-users] Synchronous dialplan execution for feedback while processing speech recognition and voice synth, for example.

I've got an agi that recognises speech (via Google) and another that turns text into speech (tts) (via Microsoft Translate). Both are web APIs, both called via seperate python AGIs. I've googled and I'm probably missing something pretty newbie 101 here, but is there any way, or fiddle, that I

Re: [asterisk-users] system metrics to see if Asterisk is getting overloaded

Funnily enough, I was just thinking the same this morning. I run two boxes, and my idea was the use a sys call to grab the loadav, multiply that by 1000 and then use that as the delay before answering. In other words, if box 1 had a loadav of 0.2 and box two have a loadav of 0.5, box 1 would

Re: [asterisk-users] cloud solution?

Something like this? https://github.com/lardconcepts/asterisk-digitalocean-voipfone-config/blob/master/Asterisk-13-on-Ubuntu.md On 27 September 2016 at 19:31, Ryan, Travis wrote: > So if someone has their own hardware and infrastructure but wants a software > (not FreePBX

Re: [asterisk-users] Asterisk 14.0.0-rc1 Now Available

Thanks, Marcelo. I just subscribed to that bug; you're right, I also noticed the lack of info when attacking to asterisk. I just forgot to mention it here! On 20 September 2016 at 13:43, Marcelo Terres <mhter...@gmail.com> wrote: > Hello Jonathan, > > https://issues.asterisk.

Re: [asterisk-users] Asterisk 14.0.0-rc1 Now Available

Great! Thanks, team, but just before I file a bug.. No matter how many v and d I put, when I now do "dialplan reload" in v14, it just says "Dialplan reloaded". Previously, it used to give some info, and I could scroll back and see if there were any obvious errors in the dialplan. Is this and

Re: [asterisk-users] Upgrading asterisk 13.7 to 13.11. Segfaults

All your libraries, kernel, headers and build tools up to date? The other thing that might be worth noting is the warning along the lines of "contains modules that were not installed by this version of Asterisk". Might be worth deleting anything that appears there, and then starting Asterisk.

Re: [asterisk-users] TLS problem

cert_file=/etc/letsencrypt/live/mysite.co.uk/fullchain.pem priv_key_file=/etc/letsencrypt/live/mysite.co.uk/privkey.pem method=tlsv1 But this won't be any good to you on sip. What version of Asterisk are you using? On 26 August 2016 at 11:36, hw <h...@gc-24.de> wrote: > Jonathan H schrieb: &

Re: [asterisk-users] TLS problem

Well, what immediately stands out is: "FILE * open failed!" Have you triple checked that the full filepath is correct and that the user that Asterisk is running as has full permissions to access your valid certificate file? I have it working with microsip and a free TLS cert from LetsEncrypt.

[asterisk-users] Asterisk 13.10.0 just randomly got pjsip endpoint amnesia.

Here's a weirdness - I got a call from someone who couldn't get to my info line earlier, I tried it and it was busy tone. Being on a layby beside a road on a mobile on a long journey, my only real option was a remote server reboot so I couldn't diagnose further. That fixed it, but here's the

Re: [asterisk-users] pjproject 2.5.5 + asterisk-certified-13.8-cert1 : many Error loading module...undefined symbol

On 17 August 2016 at 20:40, Jonas Kellens wrote: > When I compile "--without-pjproject" I loose all webrtc functionality. I get > errors about the lack of "ice-frag and ice-pwd in the SDP-body". > So I guess I DO need pjproject. But I do not want to use pjsip (I prefer

Re: [asterisk-users] Switching between Music on Hold streams. [13.8.2]

me know. The ones that > I have working is MP3 and MMS. > > On Mon, May 9, 2016 at 1:18 PM, Jonathan H <lardconce...@gmail.com> wrote: > >> Thanks Joshua and everyone, >> >> Joshua's solution seems a lot simpler and works well. Only one thing >> now - The r

Re: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ?

I'm genuinely fascinated why you are insisting on using a version of Asterisk almost 3 years old, for which EOL support ended last year. Is there any particular reason you cannot or will not use the current version as others have suggested? Also, I see you are using Doubango and WebRTC, but in

Re: [asterisk-users] Switching between Music on Hold streams. [13.8.2]

lication? Does that make sense? (It's getting late here!). Thanks! On 9 May 2016 at 18:22, Joshua Colp <jc...@digium.com> wrote: > Jonathan H wrote: >> >> Thanks Joshua and everyone, >> >> Joshua's solution seems a lot simpler and works well. Only one thing >&g

Re: [asterisk-users] Switching between Music on Hold streams. [13.8.2]

amdemo5] digit=5 mode=custom application=/usr/bin/mpg123 -q -r 8000 -f 32768 --mono -s http://206.225.87.121:8000/ On 9 May 2016 at 18:00, A J Stiles <asterisk_l...@earthshod.co.uk> wrote: > On Monday 09 May 2016, Jonathan H wrote: >> . {stuff deleted} . >> [streamdemo]

Re: [asterisk-users] Switching between Music on Hold streams. [13.8.2]

, continuing... On 8 May 2016 at 14:56, Dovid Bender <do...@telecurve.com> wrote: > Michael, > > What you do is you dial another context and then use the G option in the dial > string. So something like this. > > [radio-main] > Exten => s,1,answer > Exten => s,2,Ba

[asterisk-users] Switching between Music on Hold streams. [13.8.2]

I'd like multiple people to be able to dial in and listen to various live radio streams. I was told that the correct resource-friendly way would be to setup a MoH class, and then select that from the dialplan. This works well, but how do I switch between streams? Someone correct me if I'm

[asterisk-users] Tapping into an existing audio stream rather than starting a new mp3Player?

>From what I can tell from the Wiki page at https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_MP3Player, if someone dials in and starts playing a stream, mp3player will load up the URL and inject it into the current call. But what about if 20 or 30 people call in, and it's firing

Re: [asterisk-users] Nube question: where is chan_sip.so?

If it helps, here's a quick n easy guide I made to installing from scratch with pjsip on Ubuntu. https://github.com/lardconcepts/asterisk-digitalocean-voipfone-config/blob/master/Asterisk-13-on-Ubuntu.md There are some other scripts like firewall, wizard etc, but there are aimed at Voipfone users

Re: [asterisk-users] Asterisk 13.6 + pjsip: sip2sip registers but incoming calls get "No matching endpoint found".

uot;in sync" with the forum. Lesson learnt, and again, thank you. On 18 January 2016 at 11:57, Joshua Colp <jc...@digium.com> wrote: > Jonathan H wrote: >> >> Would greatly appreciate any input into this currently-unanswered >> question on the forum: >> >> htt

[asterisk-users] Asterisk 13.6 + pjsip: sip2sip registers but incoming calls get "No matching endpoint found".

Would greatly appreciate any input into this currently-unanswered question on the forum: http://forums.asterisk.org/viewtopic.php?f=1=96496 I posted it on Jan 6th, have tried so many things, so much forum/list searching and late nights since, but have had to admit defeat. Rather than duplicate

[asterisk-users] CDR and confbridge

Good evening all. I am having issues with CDR and confbridge. When the first call is placed into conference CDR stops tracking time. If I hang the call up the billsec reported is only up to the the time before the call enters the bridge. However if a second call joins the bridge the full amount

[asterisk-users] asterisknow-version

Does anyone know if it is possible to disable asterisknow-version from writing over my issues file? Alternatively is it really required to have it as a dependency in asterisk 13? Surly everyone has upgraded from the old repository file system and asterisk 13 wasn’t even on the old file system.

Re: [asterisk-users] Hold

Can you write the unique variable to astdb and then write it back to the variable? Not sure I have thought this through J On 11 Jun 2014 18:42, Kelly Opal ke...@ncwcom.com wrote: Hi I am trying to set up a hold system so that a call is always parked in the same spot no matter how

Re: [asterisk-users] 'restart when convenient'

I didn't know that feature existed. I'm doing a scripted restart by using the asterisk command line to tell me how many active calls are current. If 0 then restart. J On 28 May 2014 10:52, Sander Smeenk ssme...@freshdot.net wrote: Hi, I want to do a scripted 'restart when convenient' on a

Re: [asterisk-users] astdb delete all keys with the value of x

install that was not configured with ODBC support). -Josh On Mon, Apr 21, 2014 at 10:27 AM, Jonathan White j...@uvacity.com wrote: I’m trying to use the asterisk database but I think there is a limitation in deleting records I need to make my logic work. I understand that I can delete all

[asterisk-users] astdb delete all keys with the value of x

I’m trying to use the asterisk database but I think there is a limitation in deleting records I need to make my logic work. I understand that I can delete all family members with a specific key and that I can delete an entire family of keys but I would like to be able to delete specific keys

Re: [asterisk-users] ControlPlayback can not replay complicated file names

, 2014 8:02 PM To: Jonathan White ; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] ControlPlayback can not replay complicated file names This doesn't fix the issue, but a work around might be to try using file names without the any : in them -Original

[asterisk-users] ControlPlayback - Prompts

Is there a standard set of prompts to use with ControlPlayback as there are with many of the other applications like confbridge? I can’t find a prompt with instructs the user how to use it’s functions. Perhaps you have to construct a prompt out of multiple prompts. Does anyone have an idea of

[asterisk-users] ControlPlayback can not replay complicated file names

If not sure if I am looking at a bug or expected behaviour as I do not see anything in the documentation. ControlPlayback can not replay complicated file names For example it can replay 1005 but it can not replay 1005-2014-04-08_23:58:17 Playback can replay 1005-2014-04-08_23:58:17 I suspect

Re: [asterisk-users] ControlPlayback can not replay complicated file names

, Jonathan White wrote: If not sure if I am looking at a bug or expected behaviour as I do not see anything in the documentation. ControlPlayback can not replay complicated file names For example it can replay 1005 but it can not replay 1005-2014-04-08_23:58:17 On Thu, 10 Apr 2014, Eric Wieling

Re: [asterisk-users] asterisk 1.8.10.1 meetme

. r373242 comes to mind in particular. Other than that though, it would be helpful if you added some additional information, such as what arguments are are running meetme with and what kinds of devices you are connecting with (SIP phones presumably?) -- Jonathan R. Rose Digium, Inc. | Software

Re: [asterisk-users] asterisk 11 and no RTP

Jerry Geis wrote: I have a CentOS 6.3 machine I installed Asterisk 11, worked fine... I then tried to install on Cents 5.8, seemed to go fine... Then when I placed a call I got this: ast_rtp_instance_new: No RTP engine was found. Do you have one loaded? Did a search and found issues

Re: [asterisk-users] asterisk 11 and no RTP

machines... I really don't have a lot to go on. -- Jonathan R. Rose Digium, Inc. | Software Engineer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct +1 256 428 6139 Check us out at: http://digium.com http://asterisk.org

Re: [asterisk-users] Unable to create channel of type 'SIP' (cause 20 - Unknown)

. Alternatively you could just figure out how to get your devices to register to your Asterisk server. -- Jonathan R. Rose Digium, Inc. | Software Engineer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct +1 256 428 6139 Check us out at: http://digium.com http://asterisk.org

Re: [asterisk-users] Doubt regarding jabber

Harish Mandowara wrote: I have Asterisk server 1.8.19 with jabber enabled. On the other side i have openfire server with asterisk-im enabled. I have a doubt, whether my sip client connected with asterisk can send message to other sip client, which is connected to same asterisk server.

Re: [asterisk-users] Queue joinempty, even after AddQueueMember

I was poking around with the Add/Remove QueueMember code a while back. From the sound of what you are saying I might have just missed something critical. for your case. It'd be good to know what version you are using so that I can verify whether or not my changes could have affected you.

Re: [asterisk-users] Queue joinempty, even after AddQueueMember

Jonas Kellens wrote: Hello, using Asterisk 1.8.12.2 I think that was tagged before any of my recent app_queue patches. In that case, it might work if you just update to the latest 1.8 release. If it doesn't, go ahead and file an issue on JIRA. -- Jonathan R. Rose Digium, Inc. | Software

Re: [asterisk-users] Asterisk 1.8.16 Monitoring tools

. -- Jonathan R. Rose Digium, Inc. | Software Engineer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct +1 256 428 6139 Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http

Re: [asterisk-users] asterisk crashed on segmentation fault

on a test system to see if you can reproduce your problem and if you can, file a bug report against that and hope the patch either translates well to 1.6.2 without much intervention or you could attempt to backport it yourself if it doesn't. Good luck with your problem. -- Jonathan R. Rose

Re: [asterisk-users] monitor application, file name change on attended transfer

it's worth noting that if you aren't using 1.8 or higher there isn't really any point in filing a bug report since earlier versions aren't supported anymore. -- Jonathan R. Rose Digium, Inc. | Software Engineer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct +1 256 428 6139 Check us out

Re: [asterisk-users] is silk included in asterisk 11?

Jonathan Rose wrote: Sean Darcy wrote: I'm building asterisk 11 beta 2. I've been using silk a lot. I don't see silk listed in menuselect as a codec. But I also don't see an asterisk 11 silk codec on http://downloads.digium.com/pub/telephony/codec_silk. Do we use

Re: [asterisk-users] confbridge command not found

hope that helps. -- Jonathan R. Rose Digium, Inc. | Software Engineer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct +1 256 428 6139 Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth

Re: [asterisk-users] Asterisk 1.8 and 11

wiki. https://wiki.asterisk.org/wiki/display/AST/New+in+10 https://wiki.asterisk.org/wiki/display/AST/New+in+11 -- Jonathan R. Rose Digium, Inc. | Software Engineer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct +1 256 428 6139 Check us out at: http://digium.com http://asterisk.org

Re: [asterisk-users] confbridge

or something, you might consider sending them to some extension that verifies that they answered before sending them into the conference or something similar to that. -- Jonathan R. Rose Digium, Inc. | Software Engineer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct +1 256 428 6139 Check

Re: [asterisk-users] MixMonitor creating file on non-bridged calls with option b

file and remove it if you did. Are there any actual bug reports (in JIRA) you could reference though? If not, please create one and we'll look into it. -- Jonathan R. Rose Digium, Inc. | Software Engineer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct +1 256 428 6139 Check us

Re: [asterisk-users] MixMonitor creating file on non-bridged calls with option b

agreement for writing patches can't be verified if the patch isn't uploaded to JIRA (and maybe reviewboard I think). Without that we can't even legally use the patch. Thorben Jensen i...@thorben.dk wrote: Hi Jonathan, If I set the MixMonitor option on a queue, it will not create an zero length

Re: [asterisk-users] MixMonitor creating file on non-bridged calls with option b

Jonathan Rose wrote: Thorben Jensen i...@thorben.dk wrote: Hi Jonathan, If I set the MixMonitor option on a queue, it will not create an zero length file if the call is not bridged, and I just assumed it would be the case with option b. I have set the fileformat to raw

Re: [asterisk-users] MixMonitor creating file on non-bridged calls with option b

Thorben Jensen wrote: I was looking over Queue and I don't think there is actually an option for Queue that will automatically start a MixMonitor. I see a few options involving mixmonitor (x and X), but they appear to be more about allowing the parties involved with the call to start

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