Re: [asterisk-users] Alternative to Local channel
On Wed, 16 Aug 2023, Federico wrote: But now I upgraded to Asterisk18 and there is no longer a local channels Are app_originate.so and res_clioriginate.so loaded? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281 -- _ -- 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
Sorry, Joshua, I don't understand. One's a filename, one's an extension number. How are they the same? In other words, why would 'enter-ext-of-person' be considered a filename? I would think 'enter-ext-of-person' would be an extension number. On 6/17/2023 7:27 PM, Joshua C. Colp wrote: On Sat, Jun 17, 2023 at 7:48 PM Steve Matzura wrote: OK, this is how I thought it's supposed to work. It just confounded me why the book would say the Playback() and Background() syntax were the same, then in the very next paragraph give an example that belied that claim. The syntax is the same. They both take a filename. The example gave a filename of "enter-ext-of-person". You could pass that to Playback, though you would be unable to enter an extension. -- Joshua C. Colp Asterisk Project Lead Sangoma Technologies Check us out at www.sangoma.com <http://www.sangoma.com> and www.asterisk.org <http://www.asterisk.org> -- _ -- 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
OK, this is how I thought it's supposed to work. It just confounded me
why the book would say the Playback() and Background() syntax were the
same, then in the very next paragraph give an example that belied that
claim.
On 6/17/2023 1:46 PM, Doug Lytle wrote:
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
Doug, This is where the weeds start growing. On 6/17/2023 4:55 AM, Doug Lytle wrote: 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. Understood. From the book: The most common use of the Background() application is to create basic voice menus (often called auto attendants, IVRs , 9 or phone trees ). But now, the confusion: Background() has the same syntax as Playback() : [TestMenu] exten => start,1,Answer() same => n,Background(enter-ext-of-person) Stop right there. The syntax of Playback() is Playback(filename), there's no extension number. More book: 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: exten => *,VoicemailMain() -- _ -- 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] Expanding my answering-machine system
You all know the story--give the customer/client what they ask for, and if they like it, they'll be back for more. Such is just so with my one-trick-pony answering-machine project. Now the other two musicians in my virtual band want the following capabilities: 1. The ability to dial the main number from outside our three-party network and hit a button, like 1, 2 or 3, while the outgoing message is playing which will ring one of the parties now in the system. Alternatively, press a button and have everybody's phone ring and someone will eventually pick up in their studio. 2. Check voicemail remotely when not on the local network with a phone connected to the system. This is emminently doable and well described on how to do it in my book, but I have a question about how to invoke the VoicemailMain() function while the outgoing message is playing. Is it as easy as creating an extension whose number is "#" and send that to VoicemailMain()? As always, thanks in advance for a kick in the right direction. -- _ -- 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] Adding Voicemail to My System
I'm setting up voicemail on my answering-machine project. Since the directory for voicemail messages for an extension doesn't exist until there's a message to be saved therein, how can I create a custom greeting since it goes in that directory? That's what it sounds like the book is telling me anyway. Also, how do I tell the Voicemail() application to play a custom greeting? I don't mean one I can create with VoicemailMain; I mean to play a prepared file, or possibly have no greeting at all, with the greeting message actually being contained in the message played in the Play() application. It doesn't matter which way I do it, I'm just trying to figure out how to do one or the other, whichever is the right way. And one more, if I don't want voicemail messages to be sent out by email, how do I suppress this? It seems that the default is to send the email, but there's no option I could found that lets me say no, don't send. -- _ -- 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 5/28/2023 2:27 PM, Naveen Albert wrote: However, you can also pass audio without supervising (early media). You typically need to Progress() first to allow this, e.g. for SIP, or audio won't pass at all. ... If you want it to ring once and do something else, you could simply do: exten => s,1,Wait(6) ; 1 ring cycle is 6 seconds same => n,Answer(); answer, and do something else Just as you said at the top of this reply, no audio of any kind gets passed, so all the Wait(6) did was provide six seconds of dead-air silence before the outgoing message played. Oh well. Customers can't have everything. ;-) -- _ -- 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] A stupid problem with Playback
On Sun, 28 May 2023, Steve Matzura wrote: It's probably eight or nine years old now, an ASRock motherboard with I don't even know what on it in the way of processor speed or power. I should probably pick up another machine but I can't justify the expense because it's only for play, FTP, and running this Asterisk project, which is complete enough now that I don't have to mess with it any more. Who knows--it might even wind up on a spare Raspberry Pi 4, in which case this whole tower can just go away. A '4 is probably way more than you need. A Pi Zero W or a Pi Zero 2 W would probably do. ($15 plus case and spare USB wall wart.) You may be able to justify the expense just on power savings. (Electricity in San Diego is $0.51/kWh.) -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281 -- _ -- 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] Question on ring count on incoming circuits
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--the provider or Asterisk? If it's the DID provider, I'll work on that with them; if it's Asterisk, I didn't find anything anywhere that looks like it has anything to do with incoming ring count unless you set up a ring-no-answer system. For my purposes, that would mean defining a dummy extension that has no hardware attached to it that would fail over to my current call handling code after it rings once. Is this the proper method for handling this? You might wonder why I wouldn't want a call to a system that simply plays a message and then takes an optional voicemail message to pick up immediately. Short answer: Don't ask (groan). It's what the project supporter wants, presumably so that the person calling into the system will know their call went through and to be ready to hear the outgoing message, I don't know, it's a customer request so I feel duty-bound to figure it out and implement it. -- _ -- 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] A stupid problem with Playback
On 5/28/2023 6:19 AM, aster...@phreaknet.org wrote: A great reason to avoid Asterisk packages and compile from source instead. You'll save yourself a lot of headaches. That's how I started, by trying to build version 18 from source. It failed. Colossally. The compile of sources would run for a while, then the machine would crash spectacularly--I mean, not just hang or reboot. It actually turned itself off. I tried it several times, and each time it failed in the same way, but at a different spot in the compile process. If ever I could figure out a way to trace that one down, I would. It was the strangest thing. This sounds like your machine is defective in some major way. Granted, compiling software is pretty intensive, but your machine shouldn't just crash. I would try to figure that out. Is this a VM / bare metal? Have you tried this on another machine? It's probably eight or nine years old now, an ASRock motherboard with I don't even know what on it in the way of processor speed or power. I should probably pick up another machine but I can't justify the expense because it's only for play, FTP, and running this Asterisk project, which is complete enough now that I don't have to mess with it any more. Who knows--it might even wind up on a spare Raspberry Pi 4, in which case this whole tower can just go away. -- _ -- 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] A stupid problem with Playback
On 5/27/2023 11:40 AM, aster...@phreaknet.org wrote: Relative paths are relative to your language-specific directory. Ya know, that's the one thing I didn't do was test Playback before copying the sound files out of /usr/share/asterisk/sounds/en_us into /var/lib/asterisk/sounds--I don't even know for sure that hello-world was playing from the /var path and not the /usr path. Good idea to test that and see what's really going on. I think I set too much store by these books sometimes. But when that's all I have, I tend to go with what I know, and if the book is all I know ... well ... A great reason to avoid Asterisk packages and compile from source instead. You'll save yourself a lot of headaches. That's how I started, by trying to build version 18 from source. It failed. Colossally. The compile of sources would run for a while, then the machine would crash spectacularly--I mean, not just hang or reboot. It actually turned itself off. I tried it several times, and each time it failed in the same way, but at a different spot in the compile process. If ever I could figure out a way to trace that one down, I would. It was the strangest thing. So I gave up trying to build from source and went to the distro. Truth to tell, I'd rather have been able to build it from source because then I could follow my book more closely, and I enjoy and am familiar with working with SQL. I understood perfectly what the book was telling me to do and how it would all integrate with configuring Asterisk. Very strange indeed. Maybe I'll try a later version and see what happens. -- _ -- 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] A stupid problem with Playback
Acording to the book, I'm supposed to put things into what Asterisk thinks is its default audio file location, /var/lib/asterisk/sounds, and I'm supposed to be able to create a custom directory off of that path and use it in a relative-syntax way in the Playback directive, like so: ... same => n,Playback(mysounds/mygreeting) I'm here to tell ya, it doesn't work on my system. However, if I write: same => n,Playback(/var/lib/asterisk/sounds/mydir/mygreeting) it works fine. Where is the default directory defined? I search every configuration file and found no such definition. And then there was this: When I finally got my system working after all the connectivity and extension-not-found and endpoint-not-found nonsense straightened out, I of course tried the hello-world standard startup test. It didn't work. Why? Because when you install Asterisk version 16 from the Debian distro site, you don't get the core sounds, and when you do install the core sounds package, they don't get put into /var/lib/asterisk/sounds. Oh no--they get put into /usr/share/asterisk/sounds. In there, I found several directories such as 'en' and 'en_us'. I copied the files from the en_us directory into /var/lib/asterisk/sounds and hello-world worked fine. So then I created a custom directory and put my own things in it, changing the Playback statement to the first one above, and it failed. I have to specify the full path instead of using the relative syntax version thereof. This is technically not a problem, more just a curiosity as to why it didn't work the way I thought it's supposed to. -- _ -- 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
I wouldn't mind using SIP with voip.ms if it worked right. I'll try it again changing what needs changing, but for now, I'm not conducting any secure communications, so I'll worry about it when things break. On 5/27/2023 11:20 AM, aster...@phreaknet.org wrote: IAX2 tends to work really well for trunking. Unlike SIP, it usually just works, although it tends to be more a niche use case. For this reason, IAX2 has long been a controversial technology; most people seem to either love it or hate it. Obviously, you can guess what my bias is. The only downside in your case is voip.ms's IAX2 stack (whether Asterisk or something else) does not support encryption, and it does not appear they have plans to support it. If you don't mind that, it shouldn't be an issue. voip.ms is also the only major VoIP provider that supports IAX2, so if you do anything else you'll probably have to use SIP. On 5/27/2023 10:23 AM, Steve Matzura wrote: Sean, I'll take that under advisement, but Doug swears by IAX, I tried it, it worked, so until things break and break bad, I'll stick with that and try the recommended remedy, now recommended by two people. On 5/26/2023 8:08 PM, Sean Bright wrote: On 5/26/2023 5:41 PM, Steve Matzura wrote: Doug from this list got me to change my connectivity to my DID provider from SIP to IAX, and bingo, it all just worked instantly. Looking over your previous messages and the error you were receiving (the one referring to extension 's') it looks like you had your VoIP.ms account setting incorrectly configured. There is a "Device type" dropdown that needs to be set to "IP PBX Server, Asterisk, or Softswitch." If instead it is set to "ATA device, IP Phone or Softphone" (the default) then it will be sent to the 's' extension instead of the DID one. I captured a screenshot¹ from my account. I created a VoIP.ms account, acquired a DID, copy/pasted the VoIP.ms configuration samples², substituted my SIP Account User ID and passwords, restarted Asterisk, and everything worked as expected. I would never recommend new installs use IAX2, so if you envision this moving beyond the toy/PoC stage I suggest you giving PJSIP another go. Kind regards, Sean 1. https://seanbright.com/voipms.png 2. https://wiki.voip.ms/article/Asterisk_PJSIP -- _ -- 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
Thanks, Daryl. I fixed this before I saw this message by changing my connectivity from SIP to IVR/IAX on voip.ms's Manage DID Numbers page. I'll keep this one in my notes, though, should I ever do this again with SIP. On 5/26/2023 7:42 PM, Daryl Richards wrote: On 2023-05-23 7:22 p.m., Steve Matzura wrote: And I think they're both small. [May 23 18:34:12] NOTICE[46582]: res_pjsip_session.c:3968 new_invite: voipms: Call (UDP:208.100.60.12:5060) to extension 's' rejected because extension not found in context 'voipms-inbound'. Steve, In your voip.ms console, go to Account Settings -> Inbound Settings, and set Device Type to "IP PBX Server..." instead of "ATA device..." This will fix the 's' instead of the number. -- _ -- 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
Sean, I'll take that under advisement, but Doug swears by IAX, I tried it, it worked, so until things break and break bad, I'll stick with that and try the recommended remedy, now recommended by two people. On 5/26/2023 8:08 PM, Sean Bright wrote: On 5/26/2023 5:41 PM, Steve Matzura wrote: Doug from this list got me to change my connectivity to my DID provider from SIP to IAX, and bingo, it all just worked instantly. Looking over your previous messages and the error you were receiving (the one referring to extension 's') it looks like you had your VoIP.ms account setting incorrectly configured. There is a "Device type" dropdown that needs to be set to "IP PBX Server, Asterisk, or Softswitch." If instead it is set to "ATA device, IP Phone or Softphone" (the default) then it will be sent to the 's' extension instead of the DID one. I captured a screenshot¹ from my account. I created a VoIP.ms account, acquired a DID, copy/pasted the VoIP.ms configuration samples², substituted my SIP Account User ID and passwords, restarted Asterisk, and everything worked as expected. I would never recommend new installs use IAX2, so if you envision this moving beyond the toy/PoC stage I suggest you giving PJSIP another go. Kind regards, Sean 1. https://seanbright.com/voipms.png 2. https://wiki.voip.ms/article/Asterisk_PJSIP -- _ -- 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] Problems solved
Doug from this list got me to change my connectivity to my DID provider from SIP to IAX, and bingo, it all just worked instantly. For my next trick: setting up voicemail. The book does it all with smoke and mirrors (SQL), but I'm fresh outa those, so I'll be doing it the old-fashioned way, by editing the voicemail.conf and users.conf files with some hopefully helpful hints from our friends at ChatGPT. -- _ -- 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 Remaining
This was supposed to go to the list. I am now thoroughly confused. In the [voipms] stanza where endpoint is defined (type=endpoint), everything points to voipms. But in the [yealink] stanzas, I tried pointing everything to Steve, one item at a time, then both of them, and nothing changed. On 5/24/2023 10:00 AM, Stefan Tichy wrote: block quote Am Wed, May 24, 2023 at 09:40:18AM -0400 schrieb Steve Matzura: block quote On 5/24/2023 7:49 AM, Stefan Tichy wrote: block quote Am Tue, May 23, 2023 at 07:22:22PM -0400 schrieb Steve Matzura: block quote 1. Still can't register my phone The username and password are correct. I don't know what else to try. block quote end You can start a sip trace from the asterisk console. REGISTER sip:192.168.1.185:5060 SIP/2.0 block quote end block quote end block quote block quote Authorization: Digest username="Steve", realm="asterisk", [May 24 09:26:13] NOTICE[47903]: res_pjsip/pjsip_distributor.c:676 log_failed_request: Request 'REGISTER' from .' - No matching endpoint found block quote end block quote end In the endpoint section there is a parameter identify_by (default: "username,ip"). "username" means, the the from-user is used. Use "yealink" or "Steve" for both user names. Using different names for a phone make it just more complicated. You might have to change the phone configuration. block quote [yealink] type = aor contact = sip:Steve@192.168.1.185 block quote end There should be no "contact" parameter for a phone. The phone sends the required information with the register request. block quote end -- _ -- 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] Problems Solved, two left
And I think they're both small. Solved: tcpdump showed no packets coming in, so I went to my DID provider's Website to discover to my intense embarrassment that the DID number had been set up forwarded to their voicemail. I got egg on my face for this one. I changed that setting to SIP/IAX and packets now arrive and go where they should. Two problems remain. 1. Still can't register my phone The username and password are correct. I don't know what else to try. 2. Asterisk can't find the extension in my inbound context. [May 23 18:34:12] NOTICE[46582]: res_pjsip_session.c:3968 new_invite: voipms: Call (UDP:208.100.60.12:5060) to extension 's' rejected because extension not found in context 'voipms-inbound'. I changed the name of the context in pjsip's to 'voipms-inbound' and removed reference to '[mycontext]' from pjsip.conf and extensions.conf as they were superfluous. The endpoint section of pjsip.conf now reads: [voipms] type = endpoint transport = transport-udp context = voipms-inbound ... The bottom part of extensions.conf (with the phone number obfuscated) is now: [voipms-inbound] exten => 3115552368,1,Goto(hello,200,1) [phones] exten => 101,1,Dial(PJSIP/yealink) [hello] exten => 200,1,Answer() same => n,Playback(hello-world) same => n,Hangup() The idea was for any inbound call to the public network number to immediately go to extension 200, play the message and hang up, and you could still call extension 200 to here it from inside. -- _ -- 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 with inbound connection and registering phone
On Tue, 23 May 2023, Steve Matzura wrote: The "Definitive Guide" shows everything about adding phones as SQL statements... I'd look for another guide. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281 -- _ -- 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 with inbound connection and registering phone
On Tue, 23 May 2023, Steve Matzura wrote: ...when I dial my number from a phone on the Internet or any phone outside my LAN, Asterisk does not respond in any way, which means somehow my system is not picking up the fact that there's an incoming call to it. Or that you are not receiving any packets. Enabling SIP debugging in Asterisk can yield clues. `sudo tcpdump -i any -s 0 -v port 5060` can yield more clues. Note that tcpdump sees packets before iptables and Asterisk sees packets after iptables. If you're getting packets, sngrep and wireshark can yield even more clues. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281 -- _ -- 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] Problems with inbound connection and registering phone
I have two problems. The first is that when I dial my number from a phone on the Internet or any phone outside my LAN, Asterisk does not respond in any way, which means somehow my system is not picking up the fact that there's an incoming call to it. The second problem is that I thought I'd try an internal phone to see if I could get the hello-world stuff working at the least. I thought I'd try Zoiper, but none of the download buttons at https://www.zoiper.com/en/voip-softphone/download/current did anything when clicked, so I set up a spare line on a Yealink TA33 phone that is connected to another much much older Asterisk implementation running a piece of amateur radio gear called Allstar. The version of Asterisk used in the Allstar project is ancient--like 1.4 or 1.6--and the configuration syntax and options are quite different, so I didn't get lulled into thinking I'd just clone that configuration on my newer Asterisk implementation. The "Definitive Guide" shows everything about adding phones as SQL statements, so I made some educated guesses as to what to put into pjsip.conf. Something's obviously wrong because the phone won't authenticate (see below). Here's how I set it up in pjsip. [yealink] transport=udp type=auth auth_type=userpass username=Steve password=Steve [yealink] type = endpoint transport = transport-udp context = phones disallow = all allow = ulaw ; allow=g729 ; uncomment if you support g729 auth = yealink aors = yealink Here's how I set it up in extensions: [phones] exten => 101,1,Dial(PJSIP/yealink) Here's the error I get on the Asterisk console: [May 23 13:42:56] NOTICE[45189]: res_pjsip/pjsip_distributor.c:676 log_failed_request: Request 'REGISTER' from '"Steve" ' failed for '192.168.1.228:5060' (callid: 0_1554187534@192.168.1.228) - Failed to authenticate What did I omit? -- _ -- 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] Ready to throw up my hands in defeat
I haven't tried starting the daemon yet only because I wanted to verify
my pjsip and extensions stuff first before I started trying to debug
what I might not understand. I have a better handle on it all now. Will
post the results when I try it in just a few minutes.
Thankfully, the book does say SIP is deprecated in favor of PJSIP, so
I'm on board with understanding all that. Thanks for reminding me that
Google is my friend in this project, too.
handle
On 5/22/2023 12:50 PM, TTT wrote:
You don't say what happens when you start Asterisk, but I'll assume
your registration with your provider is failing. If you turn on SIP
debug from CLI you can watch your registration attempts, and see the
exact reason for failure. (eg: unreachable vs credentials). Post
that output into the list email if you aren’t sure what to make of it.
The numerous similar stanzas have to do with pjsip. Many of the older
examples/guides reference a simpler (deprecated) SIP stack, with
slightly different syntax. If you google PJSIP + Asterisk config
you'll send the purpose of all of those stanza's. PJSIP adds a lot of
complexity for the outlier use cases (99% of people don't need all of
PJSIP's capabilities) - but you have to fill it all in. There are
some nice diagrams here
(https://wiki.asterisk.org/wiki/display/AST/PJSIP+Configuration+Sections+and+Relationships)
which explains it fairly well.
Please reply to the list only (so anyone can jump in and benefit from
the discussion too)
-Original Message-
From: Steve Matzura [mailto:s...@noisynotes.com]
Sent: Monday, May 22, 2023 12:15 PM
To: TTT
Subject: Re: [asterisk-users] Ready to throw up my hands in defeat
Thanks. Further reading and digging did in fact prove out that the RTP
is a lot of what's been throwing me. I won't bother with that any longer.
I did make the "hello world" example from the Asterisk wiki work. It was
simple enough--lift the example right out of the book, paste it into the
appropriate files, install and configure Zoipr, restart Asterisk, and
it just worked. Good.
So now I'm branching out, ready to add my DID provider info and actually
be able to call in from outside.
The following file contents come from my DID provider, voip.ms. The only
thing I added was my specific DID registration info, which has been
redacted here:
pjsip.conf:
[transport-udp]
type = transport
protocol = udp
bind = 0.0.0.0
[voipms]
type = registration
transport = transport-udp
outbound_auth = voipms
client_uri = sip:**@newyork6.voip.ms:5060
server_uri = sip:newyork6.voip.ms:5060
[voipms]
type = auth
auth_type = userpass
username = **
password = **
[voipms]
type = aor
contact = sip:**@newyork6.voip.ms
[voipms]
type = endpoint
transport = transport-udp
context = mycontext
disallow = all
allow = ulaw
; allow=g729 ; uncomment if you support g729
from_user = **
auth = voipms
outbound_auth = voipms
aors = voipms
; NAT parameters:
rtp_symmetric = yes
rewrite_contact = yes
send_rpid = yes
*** NOTE: I left those lines in because I am after all behind a home
router so I thought I'd need it. ***
[voipms]
type = identify
endpoint = voipms
match = newyork6.voip.ms
Why so many stanzas all called 'voipms'? I see that they all have
different types, so why not have everything in the same stanza?
Here's extensions.conf:
[mycontext]
; 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 => 3115552368,1,Answer() ; fake DID number
Now, I have the block of text from the hello-world wiki:
[from-internal]
exten = 100,1,Answer()
same = n,Wait(1)
same = n,Playback(hello-world)
same = n,Hangup()
That's certainly clear enough.
My question is, how do I connect the two extensions.conf fragments?
i.e., where's the routing from the system answering the inbound connect
request to the actions in the extension 100 statements?
The book talks a lot about registering phones through SQL. Looking at
pjsip.conf is a little bewildering--all those '[6001]' examples. Which
ones do I absolutely need? I have two kinds of phones to register--one
on my own LAN and one remote, presumably coming in with NAT'ing and
definitely behind its own firewall on its own LAN.
On 5/22/2023 10:59 AM, TTT wrote:
> There are lots of little tweaks/adjustments overlooked in most
guides/books.
[asterisk-users] Ready to throw up my hands in defeat
I am not comfortable with admitting this on a public userlist [;-)] but after over forty years in software development and manual-reading and -interpretation, I've finally hit one that I can't get past. I've mention previously that I worked with Asterisk in older days--like in around 2003--and never had any trouble understanding what to do and how to do it in order to make it work. I am attempting to build what's probably the world's most basic system--one incoming trunk from a DID provider going to one internal extension that answers, plays a couple things, and possibly takes a message. I'd also like to add two extensions with real physical endpoints--phones--one local, one remote. I think I can manage that part. It's the initial SIP stuff that's making me dizzy. The book I am now reading--"Asterisk, the Definitive Guide" by Madsen, Bryant and Meggelin for Asterisk version 16-- assumes I have built an implementation from source, and that includes SQL. There are tons of references to SQL databases in the book which I understand, but having installed Asterisk from a distribution package, that component is not part of the installation, so I am presumably expected to supply the information by manually entering it into configuration files. I'm OK with doing that, too. The part I'm having trouble with is that the samples in the configuration files, particularly pjsip.conf, offer several choices for some of the stanzas, like all the things defining trunks and endpoints, and that's where I'm losing it. The book makes it sound and look so easy--add a couple records to a couple SQL tables according to your instruments and DID providers, and it probably works just that smoothly and easily. But how does one make these choices when one has to manually edit these configurations and choose the one that at least halfway looks like the SQL stuff in the book? I think I need a little hand-holding and am willing to buy some from someone who has the time and inclination to provide it. I'm a fast learner, I record all such sessions, and I'm sure I can get what I need in a couple hours, most likely less. if you're interested, or know someone who is, please contact me off-list, with my eternal thanks in advance. -- _ -- 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] Remote-Party-ID set to 0 on re-invite using pjsip in Asterisk 16.
We've found the problem. For anyone seeing this same problem in the future, the underlying problem was a miss-placed entry in our pjsip.conf callerid=0 If this is set, any changes to Remote-Party-ID are ignored, even when CONNECTEDLINE(num) is called in the dialplan. I don't know if this is desired behavior or not, but it's how pjsip seems to work. On 4/6/23 2:54 PM, Steve Sether wrote: We've been using Asterisk 16 for a while now, and tried turning on send_rpid = yes in my pjsip config for end points. This solves a problem we're having where attended transfers aren't updating the CallerID when the transfer is complete (it would show the callerID of the party attempting the transfer, and never update after the transfer happened). The side effect of this change is that now on all outgoing calls, the phone I placed the call on shows 0 as the dialed number, which is wrong of course. This shows up in a pcap to the phone that dialed as a re-invite sent back to it (before the call even connects), and shows something like: Remote-Party-ID: ;party=called;privacy=off;screen=no I'm not really certain why Asterisk is setting a Remote-Party-ID to 0, which I believe is the default. Clearly it knows who it's calling. I could fix this by explicitly setting the header, but that seems wrong, and Asterisk should be setting this correctly. Any help is appreciated. I've gone through a bunch of different attempts to fix this by changing pjsip settings (none worked), but in the end I don't know why Asterisk sets this header so wrong. We're using Asterisk 16.30, and of course pjsip. -- -- -- _ -- 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] TLS and NAT
Thanks, Michael. A few questions: Is [transport_name] a reserved word, or am I supposed to replace it with a name of my own, like '[did-transport]'? Some of the keywords I haven't seen before. Is ca_list_file supposed to be an aggregate of the public and private key? And what are the 'method,' 'tos' and 'cos' keywords, which are commented out in your instructions? Otherwise, the rest is quite clear. On 4/8/2023 12:35 PM, Michael Maier wrote: Hello Steve, use the following configuration for the transport and bind this transport to the trunk: [transport_name] type=transport protocol=tls bind=192.168.13.24 ; your bind IP ca_list_file=/etc/pki/tls/certs/ca-bundle.crt ; method=tlsv1_2 verify_server=yes allow_reload=no ;tos=0xb8 ;cos=3 external_media_address=your.ext.host.name ; hostname pointing to your ext. IP external_signaling_address=your.ext.host.name ; hostname pointing to your ext. IP local_net=192.168.0.0/24 # your local net Regards Michael On 07.04.23 at 17:25 Steve Matzura wrote: I want to configure communication with my phone provider using TLS for all the obvious reasons. Since I'm behind a firewall, I'll be needing to do it with NAT. There are examples of UDP plus NAT in pjsip.conf, but none for TLS plus NAT. Would it be correct to set up the TLS transport stanza to look like the [transport-udp-nat] stanza example, replacing UDP with TLS in lines like 'transport=tls' and 'protocol=tls', and including the lines for local_net, external_media_address and external_signaling_address? -- _ -- 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] TLS and NAT
I want to configure communication with my phone provider using TLS for all the obvious reasons. Since I'm behind a firewall, I'll be needing to do it with NAT. There are examples of UDP plus NAT in pjsip.conf, but none for TLS plus NAT. Would it be correct to set up the TLS transport stanza to look like the [transport-udp-nat] stanza example, replacing UDP with TLS in lines like 'transport=tls' and 'protocol=tls', and including the lines for local_net, external_media_address and external_signaling_address? -- _ -- 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] Intro and question
Sorry, meant version 16, like the book. Sure would prefer 20. On 4/6/2023 3:30 PM, Steve Matzura wrote: It appears I have bigger problems heretofore unknown. I've gone through this several times today since I last wrote, and the phreaknet-run build failed every time, but each time at a different point. The failure manifestation was always the same--during either a compile or link of some module, a different one each time, my entire system turned itself off, which to me means there is either a memory fault which is causing the whole thing to dump, or I have a hidden hardware problem I'll probably never find. I'm just going to have to go back to the original Debian-supplied version 18 installation and live with its limitations until such time as I can get another system to try building on, or something about the machine I have shows itself and can be fixed. On 4/6/2023 12:34 PM, Antony Stone wrote: On Thursday 06 April 2023 at 18:29:43, Jeff LaCoursiere wrote: If you just want something easy to use out of the box, install the FreePBX distro. Given that Steve originally said "I've been using Asterisk, including administering and maintaining it, in some aspect since 2003, but this is the first time I have attempted a from-scratch installation and setup on my own." I got the impression that he was not so much looking for something easy to use, but rather looking forward to learning about how to "do Asterisk" for himself. Antony. -- _ -- 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] Remote-Party-ID set to 0 on re-invite using pjsip in Asterisk 16.
We've been using Asterisk 16 for a while now, and tried turning on send_rpid = yes in my pjsip config for end points. This solves a problem we're having where attended transfers aren't updating the CallerID when the transfer is complete (it would show the callerID of the party attempting the transfer, and never update after the transfer happened). The side effect of this change is that now on all outgoing calls, the phone I placed the call on shows 0 as the dialed number, which is wrong of course. This shows up in a pcap to the phone that dialed as a re-invite sent back to it (before the call even connects), and shows something like: Remote-Party-ID: ;party=called;privacy=off;screen=no I'm not really certain why Asterisk is setting a Remote-Party-ID to 0, which I believe is the default. Clearly it knows who it's calling. I could fix this by explicitly setting the header, but that seems wrong, and Asterisk should be setting this correctly. Any help is appreciated. I've gone through a bunch of different attempts to fix this by changing pjsip settings (none worked), but in the end I don't know why Asterisk sets this header so wrong. We're using Asterisk 16.30, and of course pjsip. -- -- _ -- 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] Intro and question
It appears I have bigger problems heretofore unknown. I've gone through this several times today since I last wrote, and the phreaknet-run build failed every time, but each time at a different point. The failure manifestation was always the same--during either a compile or link of some module, a different one each time, my entire system turned itself off, which to me means there is either a memory fault which is causing the whole thing to dump, or I have a hidden hardware problem I'll probably never find. I'm just going to have to go back to the original Debian-supplied version 18 installation and live with its limitations until such time as I can get another system to try building on, or something about the machine I have shows itself and can be fixed. On 4/6/2023 12:34 PM, Antony Stone wrote: On Thursday 06 April 2023 at 18:29:43, Jeff LaCoursiere wrote: If you just want something easy to use out of the box, install the FreePBX distro. Given that Steve originally said "I've been using Asterisk, including administering and maintaining it, in some aspect since 2003, but this is the first time I have attempted a from-scratch installation and setup on my own." I got the impression that he was not so much looking for something easy to use, but rather looking forward to learning about how to "do Asterisk" for himself. Antony. -- _ -- 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] Intro and question
Anthony, No, I had no intention of doing any of those things, for I know not what they are or why I would need or want to be doing them. Maybe I should have just stuck with the original idea of installing from Debian distro. I'm exploring the phreaknet option now. If I come up with a running system, I'm just going to leave it that way and work with it as it is. On 4/6/2023 10:35 AM, Antony Stone wrote: On Thursday 06 April 2023 at 15:48:24, Steve Matzura wrote: this is the first time I have attempted a from-scratch installation and setup on my own. .. Then the weeds started to appear, and I was off into them. The first was the mention of Alembic. Reading on, I found this, regarding an SQL database: SQL? Database? Where ... what ... Thanks in advance for any assistance. Well, my first question would be "are you intending to use Asterisk Realtime features (ie: configurations in database tables instead of text files) in this installation?" If you are, then you do need to install a few more packages on your Debian system, but if not, then there is no reason to pay any attention at all to anything to do with Alembic, Realtime, SQL etc. Antony. -- _ -- 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] Intro and question
I've been using Asterisk, including administering and maintaining it, in some aspect since 2003, but this is the first time I have attempted a from-scratch installation and setup on my own. I'm following the instructions in the ePub edition of the book "Asterisk, the Definitive Guide, Fifth Edition," published by O’Reilly Media, Inc. in 2019, for Asterisk version 16 on a fresh install of Dedbian 11.6 (Bull's Eye). Backround: In Chapter 3, "Installing Asterisk," in the section "Asterisk Packages," the book says, and I quote from the ePub: There are Asterisk packages that can be installed using package management systems such as yum or apt-get . You are encouraged to use them once you are familiar with Asterisk. So, thinking I was familiar enough with the product, I indeed used aptp[get to install. # apt-get install asterisk I got lots of dependencies, plus the core, everything went swimmingly, no errors. In a minute or two, I had Asterisk 1:16.28.0~dfsg-0+deb11u2. Great. A little further along down the book, there's an "Initial Configuration" section detailing some changes to modules.conf and logger.conf, and a few ownership change commands to certain files and directories. All went exactly according to the documentation. Then the weeds started to appear, and I was off into them. The first was the mention of Alembic. This was not installed as part of the apt-get installation mentioned above, but I expected whatever Alembic could do, I could do manually, it'd just take longer and be more tedious. NO problem. I'd get through it. Reading on, I found this, regarding an SQL database: Log into the database now, and review all the tables that have been created: SQL? Database? Where ... what ... I got no SQL when I installed from the Debian package management system, nor was there any mention of it in the book with regard to complete package installation. Come to think of it, on some of the old implementations on which I'd worked in the past, I don't remember seeing SQL as a part thereof. Time to put the brakes on, find out what's going on, or what I did wrong. So, my question is, what is the correct approach to supplementing or correcting the standard Debian package installation, or if there isn't one, should I remove what I installed and stqart over, or even deeper, re-generate the Debian system (which I can do in less than fifteen minutes) and build from sources according to the book? Thanks in advance for any assistance. -- _ -- 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] 401 error
On Thu, 9 Mar 2023, Jerry Geis wrote: Trying to setup an incoming call with a DNIS When I dial the number - I see nothing on the CLI. Have you enabled [PJ]SIP debugging? Bumping up console debug and verbose levels may also yield clues. tcpdump+sngrep are my 'gotos' for packet analysis, but this may not need too much depth. The person says the server is returning 401 How do I debug that. Using asterisk 18.8.0 https://en.wikipedia.org/wiki/List_of_SIP_response_codes#:~:text=401%20Unauthorized,1%5D%3A%E2%80%8A%C2%A721.4.2 "401 Unauthorized The request requires user authentication. This response is issued by UASs and registrars.[1]: §21.4.2" My guess would be a user or password mismatch. Are you using SIP or PJSIP? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281-- _ -- 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_sqlite3
On Sat, 4 Mar 2023, Sean Bright wrote: On Sat, Mar 4, 2023 at 1:29 PM, Fourhundred Thecat <400the...@gmx.ch> wrote: /var/log/asterisk/master.db how can I change the location ? If this is not possible to change in the config file, where in the source code would I change that? cdr/cdr_sqlite3_custom.c line 311 Or... In the [directories] section of asterisk.conf, you can set astlogdir which is usually set to /var/log/asterisk/. If you want to change the actual file name, may the source be with you. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281 -- _ -- 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] Run asterisk -rx "command" and get plain text output
On Wed, 3 Aug 2022, Carlos Chavez wrote: ...running "asterisk -rx"... The '-r' is implied: -x command Connect to a running Asterisk process and execute a command on a command line, passing any output through to standard out and then terminating when the com‐ mand execution completes. Implies -r when -R is not explicitly supplied. Also, as already suggested, please try adding: [options] nocolor = yes in your asterisk.conf file. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281-- _ -- 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] GET DATA on AGI
On Sun, 27 Feb 2022, Dovid Bender wrote: When using GET DATA in an AGI it seems that the # key ends the input. So if say I want the user to input 123#456 the system will return 123. I did not see this in the documentation. Is this a bug, lack of documentation or do I have a bug in my AGI? AFAIK, # is it. I use 'wait for digit' in a loop to accumulate digits so I can terminate entry based on the number of digits or a specific key. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281 -- _ -- 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 16 voicemail app not playing wav49 files.
We're having a problem where Asterisk 16 refuses to play voicemail recordings and greetings stored in wav49 format. It throws an error similar to the following: 2022-01-27 11:31:37 format_wav.c: Not a supported wav file format (49). Only PCM encoded, 16 bit, mono, 8kHz/16kHz files are supported with a lowercase '.wav' extension. 2022-01-27 11:31:37 file.c: Unable to open format wav 2022-01-27 11:31:37 file.c: Unable to open /var/spool/asterisk/voicemail/user-VOICEMAIL/9202816/unavail (format (ulaw)): No such file or directory These greetings and voicemail were all created on an Asterisk 11 server, and they won't play on 16. The only thing we're doing slightly differently is we use the ODBC backend to store the recording as a binary blob. I'd make a stab based on the error message that the problem has something to do with there not really being a filename, since it's stored in a database. But this is speculation. It's worth noting that Asterisk 16 will record Asterisk voicemail and greetings as a PCM wav file, despite the default entry in voicemail.conf of: format=wav49|gsm|wav So it seems that Asterisk is ignoring what's in voicemail.conf, and just storing the file as a PCM wav. Changing this to format=wav49 changes nothing. Plaback of the PCM wav files works fine. Any help is appreciated. Thanks. -- -- _ -- 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 maybe a freepbx question
On Sat, 8 Jan 2022, John Covici wrote: How can both sip and pjsip be listening at port 5060 at the same time... They can't. One application per address/port pair. You can configure pjsip to bind to another address and/or port while you figure it out the configuration. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281 -- _ -- 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] Get context with hangup handler
On Wed, 5 Jan 2022, Dovid Bender wrote:
I thought of this but that would mean I would need to add this to the
beginning of every context which I can do, but I was trying to avoid.
Every extension in every context.
Or maybe get funky with a wildcard extension with priority = 1 and
starting all of your real extensions with priority = 3. Something like
this (which uses gosub() just for ease of testing):
; test wildcard extension
same = n, gosub(wildcard-extension,1234,1)
same = n, gosub(wildcard-extension,s,1)
same = n, gosub(wildcard-extension,testing,1)
same = n, hangup()
[wildcard-extension]
; save the current context so it can be used in the hangup handler
exten = _!.,1, verbose(1,[${EXTEN}@${CONTEXT}])
same = n, set(LAST-CONTEXT=${CONTEXT})
same = 4, return
; note all the 'real' extensions start with priority = 3
exten = 1234,3,
verbose(1,[${EXTEN}@${CONTEXT}!${PRIORITY}])
exten = s,3,
verbose(1,[${EXTEN}@${CONTEXT}!${PRIORITY}])
exten = testing,3,
verbose(1,[${EXTEN}@${CONTEXT}!${PRIORITY}])
; be explicit with 'h' so it doesn't get handled by the wildcard extension
exten = h,1,verbose(1,[${EXTEN}@${CONTEXT}])
same = n, hangup()
Hopefully somebody else has a more elegant solution.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244281
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Re: [asterisk-users] Get context with hangup handler
On Wed, 5 Jan 2022, Steve Edwards wrote:
same = n, set(LAST-CONTEXT=${context}
Double damn. I munged the case on ${CONTEXT}. I give up for today :)
--
Thanks in advance,
-
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Re: [asterisk-users] Get context with hangup handler
On Wed, 5 Jan 2022, Steve Edwards wrote:
same = n, set(LAST-CONTEXT=${context}
Damn. forgot the closing parentheses :)
--
Thanks in advance,
-
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https://www.linkedin.com/in/steve-edwards-4244281
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Re: [asterisk-users] Get context with hangup handler
On Wed, 5 Jan 2022, Dovid Bender wrote:
I have a hangup handler that's added at the beginning of a call. It logs
all the call details. Using the CONTEXT variable I am always going to
get the context where the code is being ran and not the last context
that the caller is in. Is there any creative way to get the last context
at the call was in?
At the start of each context/priority...
exten = ,1, verbose(1,[${EXTEN}@${CONTEXT}])
same = n, set(LAST-CONTEXT=${context}
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244281--
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Re: [asterisk-users] Exec two commands with ExecIf
On Thu, 23 Dec 2021, Dovid Bender wrote: Is there any way of using ExecIf to run two commands instead of 1? e.g. instead of Exten 123,1,ExecIf($["FOO" == "BAR"]?BackGround(you-owe)) Exten 123,1,ExecIf($["FOO" == "BAR"]?SayNUmber(100")) I would ideally like to do it in one line. 1) gotoif() 2) gosub() 3) AEL gosub() is probably 'cleaner' and more maintainable than gotoif(). AEL is good but sometimes fragile. -- Thanks in advance, ----- Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281-- _ -- 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] Arrays in Asterisk
On Wed, 22 Dec 2021, Steve Edwards wrote: same = n, set(ARRAY(foo1,foo2,foo3,foo4)=1,2,3,4) Just to be clear... The use of sequential ascending numbers in all of the examples should not be construed as having any meaning. You could just as easily have: same = n, set(ARRAY(foo,bar,baz,boo)=do,wop,be,bop) -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281 -- _ -- 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] Arrays in Asterisk
On Wed, 22 Dec 2021, Dovid Bender wrote:
I am experimenting with arrays in Asterisk. I am looking
at https://wiki.asterisk.org/wiki/display/AST/Function_SHIFT
and https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_ARRAY.
So for example I Do Set(FOO(1,2,3,4)=10,20,30,40)
What would be the correct way to get both the key and value into an
array? I want to say for instance OPT=1 and then OPT_VAL=10. Then on the
next interaction for OPT=2 and OPT_VAL=20 etc.
Is this possible or am I looking at this wrong?
In my mind, Asterisk does not really have arrays. You can set channel
variables and pretend they are an array, but they are not an array like in
real programming languages -- like 'how many elements are in this array'
or 'throw an exception if I try to access an invalid subscript' or
'iterate over every element in this array.'
For example, you can set channel variables like:
same = n, set(foo1=1)
same = n, set(foo2=2)
same = n, set(foo3=3)
same = n, set(foo4=4)
or
same = n, set(ARRAY(foo1,foo2,foo3,foo4)=1,2,3,4)
or
same = n, mset(foo1=1,foo2=2,foo3=3,foo4=4)
and 'dumpchan()' will show 4 discrete variables that have the same 3
letters and no other relationship.
(You could also 'set(ARRAY(foo1,bar2,baz3,boo4)=1,2,3,4)' to see that
there is no difference from setting channel variables as discrete 'set()'
statements or using the 'ARRAY()' function or using the 'mset()'
application)
You can 'pretend' a variable is an array by concatenating a 'subscript'
like:
same = n, set(foo${key}=value)
or
same = n, set(foo-${key}=value) ; a little bit more readable
but 'dumpchan()' will show each 'element' as a discrete channel variable.
Note that 'key' does not need to be numeric. It is just text that is
concatenated to form the channel variable name. This may be used to
pretend that Asterisk has associative arrays.
The 'SHIFT()' function just removes and returns the leading substring of a
variable up to a delimiter. It has even less to do with arrays than
'ARRAY()' :)
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
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Re: [asterisk-users] Python AGI's and hangups
On Wed, 8 Dec 2021, Dovid Bender wrote:
Could some change had been made to Asterisk that would alter how the
Python script dies when there is a hangup?
The setting of the '${AGISIGHUP}' channel variable should be checked.
The output of 'agi set debug on' may yield clues.
Do AGIs in other languages exhibit similar behavior?
I have no specific knowledge of Python/AGI. Sorry.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244281
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Re: [asterisk-users] Dial() after the h extension has been invoked?
On Fri, 12 Nov 2021, Steve Edwards wrote: I prefer to do database work in an AGI. I find quoting within the database to be obtuse and fragile. s/database/dialplan/g -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281 -- _ -- 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] Dial() after the h extension has been invoked?
On Fri, 12 Nov 2021, Antony Stone wrote: I've never used AGI, so what would your suggested solution involve? If all you need is to update/insert/delete some rows in a database, ODBC could be a solution. I prefer to do database work in an AGI. I find quoting within the database to be obtuse and fragile. Also, I find error handling better in an AGI with a real programming language. Also, also, things start with 'I just need to do x' and frequently grow to 2SLGBTQQIA+x and you will wish you started with a real programming language. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281 -- _ -- 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] Dial() after the h extension has been invoked?
On Fri, 12 Nov 2021, Antony Stone wrote: Can anyone suggest how I might be able to do this? I need to perform a Dial() command after an inbound channel has hung up. I do not expect the Dial() to bridge to anything (the context being dialled simply does some database manipulation and then hangs up without even bothering to answer). Any suggestions welcome :) How about creating a call file in the h extension? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281 -- _ -- 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] [FreePBX 15 and Asterisk 16] Changing/Migrating SIP Trunk Provider from DIDLogic to Hoiio in Singapore
On Thu, 11 Nov 2021, Turritopsis Dohrnii Teo En Ming wrote: Subject: [FreePBX 15 and Asterisk 16] Changing/Migrating SIP Trunk Provider from DIDLogic to Hoiio in Singapore This may be more useful if sent to a FreePBX mailing list. Redundant links: Agreed. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281 -- _ -- 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] Delay when dialing...
On Fri, 23 Jul 2021, Jeff LaCoursiere wrote: Are you sure the call has been sent? Some phones have odd dialplans installed, and may not send the call to the SIP relay until you meet the dialplan reqs, press #, or otherwise wait the inter-digit timeout before the call is actually placed. If you enable SIP debugging (and bump up debug and verbose), is the delay between when you dial and the INVITE is displayed or is the delay between the INVITE and subsequent steps in your dialplan. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281 -- _ -- 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] Patch to remove numbers from the logs
Please don't top-post. On Thu, 22 Jul 2021, Patrick Wakano wrote: If you need something quick you could create a batch script with sed or awk to remove the log lines you want and attach it to the prerotate script of logrotate (in case you use any of these in your env). Certainly this is not a final solution but it is already something that doesn't depend on an asterisk patch. On Thu, Jul 8, 2021 at 3:58 PM Dovid Bender wrote: We have a project where people will be making payments over the phone. I would like block Asterisk from logging any time the system is processing a card. So be it SayDigits(123456789), when the user enters DTMF or when I pass a card number as a variable to an AGI etc. I assume this affects others and I would like to have the patch created in a way that a. will be accepted by Sangoma and b. will work for anyone else that has this issue. I suspect the concern is having credit card numbers anywhere on disk, anytime. Your post suggests an alternative method that may be workable... rsyslog has a module, 'omprog' -- "This module permits to integrate arbitrary external programs into rsyslog's logging" I've never used it, but the description implies you could configure Asterisk to log to syslog, and then use rsyslog+omprog to pipe the messages through a script to filter out '16 digit numbers starting with 456' or '15 digit numbers starting with 3.' Way back in the day (before PCI), we used to keep the first 6 digits (the BIN) and the last 4 digits and replace the rest with x. We used to call the result a 'span.' I have no idea if this is current practice. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281-- _ -- 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] HELP! AGI AUTOHANGUP does not seem to hangup the channel.
On Wed, 26 May 2021, Jonathan H wrote:
AGI Rx << SET AUTOHANGUP 5
AGI Tx >> 200 result=0
AGI Tx >> HANGUP <<
This does raise a question in my mind...
The AGI protocol is: your AGI sends a request (the Rx line) and receives
a response (the Tx line). 1 line out, 1 line in.
If the 'HANGUP' text can arrive asynchronously, how are you supposed to
know it has arrived? Poll (or select) on the file pointer?
I cannot use other methods like setting the absolute channel timeout
variable
I don't understand why you can't use the absolute channel timeout.
Wherever you 'set autohangup x' just set 'TIMEOUT(absolute)=${EPOCH}+x.'
--
Thanks in advance,
-----
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244281
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Re: [asterisk-users] HELP! AGI AUTOHANGUP does not seem to hangup the channel.
On Wed, 26 May 2021, Jonathan H wrote: It just causes AGI to send "HANGUP" and any audio to stop playing. It does NOT hangup the channel, or even send any SIP event. The line just goes silent. I wouldn't expect the AGI() application to send a SIP event. The AGI() application does not care what technology you use. Receiving 'HANGUP' as text from Asterisk appears to be a FastAGI thing which kind of makes sense -- if your FastAGI server is not localhost, how could Asterisk send it a signal? Are you supposed to close your TCP connection and exit your AGI when you receive the HANGUP text? When I set autohangup in a 'normal' AGI, it looks like this: AGI Tx >> agi_request: null-agi.php AGI Tx >> agi_channel: SIP/poly-77a1-02a2 AGI Tx >> agi_language: en AGI Tx >> agi_type: SIP AGI Tx >> agi_uniqueid: 1622093977.1168 AGI Tx >> agi_version: 13.14.1~dfsg-2+deb9u4 AGI Tx >> agi_callerid: 55 AGI Tx >> agi_calleridname: Steve Edwards AGI Tx >> agi_callingpres: 0 AGI Tx >> agi_callingani2: 0 AGI Tx >> agi_callington: 0 AGI Tx >> agi_callingtns: 0 AGI Tx >> agi_dnid: * AGI Tx >> agi_rdnis: unknown AGI Tx >> agi_context: newline AGI Tx >> agi_extension: * AGI Tx >> agi_priority: 6 AGI Tx >> agi_enhanced: 0.0 AGI Tx >> agi_accountcode: AGI Tx >> agi_threadid: 1945654064 AGI Tx >> AGI Rx << set autohangup 5 AGI Tx >> 200 result=0 > 0x73c3dba0 -- Strict RTP learning complete - Locking on source address 192.168.0.139:2254 (and then after 5 seconds) -- AGI Script null-agi.php completed, returning 4 -- Thanks in advance, ----- Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281 -- _ -- 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] AGI: Why is stream file and wait for digit result ASCII, but get data is "normal"?
On Mon, 24 May 2021, Jonathan H wrote: any idea why it was done like this, and why someone would ever need the ascii result Maybe because ABCD are valid DTMF events? Maybe because 0 means playback completed, not a 0 was pressed? IIRC there are some inconsistencies in the AGI API that I stumbled across when I wrote my library back in '04. If you're not using a library, you may want to consider it. AGI Rx << STREAM FILE "hello-world" "1,2,3,4,5,6,7,8,9,*,0,#" 'Comma' is not a valid 'digit' so this the same as '#*0123456789' -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281 -- _ -- 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] S3 Bucket support for playing sound files
On Thu, 6 May 2021, Jonathan H wrote: "bumps up the outgoing volume to +7" I use 'normalize --amplitude=-22dB" to adjust volume levels to consistent levels. -- Thanks in advance, ----- Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281-- _ -- 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] Loading Json values into asterisk as variable values
On Fri, 26 Feb 2021, Dovid Bender wrote: Steve, What language are your AGI's written in? I have been using PHP for a long time and every time it's launched there seems to be a run on the CPU. I wonder if I would be better off using Python or something other than PHP. C. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281-- _ -- 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] Loading Json values into asterisk as variable values
On Thu, 25 Feb 2021, Dovid Bender wrote: Other than creating an AGI that opens a file to get a json object to set as variables is there any other easy way to set variables for a call when it starts? Regardless of if there is a way in dialplan, I'd vote for an AGI to avoid what I suspect will be a bunch of fragile, difficult to maintain dialplan with quoting issues. But, I am an AGI kind of guy :) Some may argue that dialplan MAY be more performant, but I have an AGI that sets over 2,000 channel variables from MySQL tables and nobody has ever complained about call startup time. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281-- _ -- 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] Calls sometimes ring through to paused agents on Asterisk 16.
We have an auto-pause feature where agents are paused after a call, and manually un-pause when they're finished with wrap-up. This worked perfectly in Asterisk 11. We've recently switched to Asterisk 16, and we now occasionally hear reports of users saying a call rang-through after the auto-pause. The initial thought that this happened during the brief time the call ended, and the time the pause was activated. So we added a custom state to the device that answered the call set to INUSE as soon as the call was answered. Once the call was completed, the agent gets paused, and only then does the state get set to UNAVAILABLE. Hints are set appropriately for the user. Testing manually by setting this custom state stops queue calls from getting to the user. We still occasionally get ring troughs though. It happens somewhat rarely, and likely works fine 99% of the time. We've tried testing this using sipp to create a lot of queue calls. Even with a high load of around 40 calls distributed on 2 queues we haven't been able to reproduce the behavior. We've tried answering more than 70 calls in a test environment, and none have reproduced the ring-through behavior. Has anyone seen any similar behavior in queues in Asterisk 16 where a call goes to an agent even though they're paused/INUSE? Any other suggestions are appreciated, though gathering more data is especially difficult since the event seems to be rare. -- -- _ -- 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] AGI Script Returning 4
On 1/30/21 1:18 PM, Alexander Perkins wrote:
the PHP-AGI script fails after it is executed and simply returns
'returning 4'.
On Sat, 30 Jan 2021, Michal Rybarik wrote:
I think this can happen by hanging up the call by one party, when SIGHUP
is sent to AGI script. PHP will exit on SIGHUP. It can be resolved by
initializing signal handler in PHP script (pcntl_signal) for SIGHUP and
doing nothing in it (return).
From res_agi.c:
enum agi_result {
AGI_RESULT_FAILURE = -1,
AGI_RESULT_SUCCESS,
AGI_RESULT_SUCCESS_FAST,
AGI_RESULT_SUCCESS_ASYNC,
AGI_RESULT_NOTFOUND,
AGI_RESULT_HANGUP,
};
so, AGI_RESULT_HANGUP == 4.
When Asterisk detects the hangup on the channel, it sends a SIGHUP to your
AGI.
I always set a signal handler on SIGHUP and do what makes sense to my
application: maybe some cleanup or syslog() before return() or exit().
Almost always exit().
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244281
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Re: [asterisk-users] Get a SHAKEN Identity Token (Alexander Perkins)
On Mon, 25 Jan 2021, Jeff LaCoursiere wrote: So how does this guy get around it? It sounds to me like he is offering to sign calls for whoever, which IMO totally defeats the purpose. IIRC, back when he first started hawking his solution, he accepted everything. Numbers from Vitelity, my old out of service copper number, 555-555-. I'm all for the discussion, but can you start a new thread so we don't keep associating the innocent party (the OP) with this spammer. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281-- _ -- 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] Get a SHAKEN Identity Token (Alexander Perkins)
On Sun, 24 Jan 2021, Saint Michael wrote: Please look at this https://issues.asterisk.org/jira/browse/ASTERISK-28924 I have a solution that works for any version of Asterisk, if interested contact me at venefax at the Google mail service. "I have a commercial solution that works for any version of Asterisk, if interested contact me at venefax at the Google mail service." Fixed. If you're going to post a commercial solution on a non-commercial forum, at least be up front about it. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281 -- _ -- 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 timing
On Wed, 6 Jan 2021, Dovid Bender wrote: The question is if it's using the card or the card or dahdi dummy (or whatever it's called) or if the card itself is being used. Does this yield a clue? pbx10:newline:13:47:53> module show like tim Module Description Use Count Status Support Level res_timing_pthread.so pthread Timing Interface 0 Running extended res_timing_timerfd.so Timerfd Timing Interface 1 Running core (I don't have any boxes using cards so I can't test.) -- Thanks in advance, ----- Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281 -- _ -- 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 timing
On Wed, 6 Jan 2021, Dovid Bender wrote: I have a box that I suspect had timing issues. I added a TE131 to see if that would help. Is there any way for me to verify that Dahdi is using the card for timing and not the kernel? Does this yield a clue: pbx10:newline:13:25:02> timing test Attempting to test a timer with 50 ticks per second. Using the 'timerfd' timing module for this test. It has been 1000 milliseconds, and we got 50 timer ticks -- Thanks in advance, ----- Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281-- _ -- 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] Detect if people is talking
On Wed, 30 Dec 2020, Valter Nogueira wrote: We have some agents that pick calls but say nothing, letting customers "alone". Is there any way to detect if an agent is speaking? I'm not sure I understand the situation. Are you saying agents are failing to do their job and just let the customer wait until they hang up in frustration? If you record the calls, could you analyze them after the call? I don't use agents or queues so I don't know if it is possible, but the 'monitor()' application records each leg in a separate file. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281 -- _ -- 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] faxdetect timeout configuration
On Wed, 30 Dec 2020, David Cunningham wrote: Would anyone be able to tell us how to configure this option for calls arriving via chan_sip? A 30,000 ft peek suggests you're out of luck unless you switch to pjsip: -ws10::sedwards:~$ rgrep -l faxdetect_timeout /usr/src/asterisk-17.4.0/ /usr/src/asterisk-17.4.0/CHANGES /usr/src/asterisk-17.4.0/ChangeLog /usr/src/asterisk-17.4.0/channels/chan_dahdi.c /usr/src/asterisk-17.4.0/channels/chan_dahdi.h /usr/src/asterisk-17.4.0/channels/chan_misdn.c /usr/src/asterisk-17.4.0/channels/chan_pjsip.c /usr/src/asterisk-17.4.0/channels/misdn/chan_misdn_config.h /usr/src/asterisk-17.4.0/channels/misdn_config.c /usr/src/asterisk-17.4.0/configs/samples/chan_dahdi.conf.sample /usr/src/asterisk-17.4.0/include/asterisk/res_fax.h /usr/src/asterisk-17.4.0/include/asterisk/res_pjsip.h /usr/src/asterisk-17.4.0/res/res_fax.c /usr/src/asterisk-17.4.0/res/res_pjsip/pjsip_configuration.c -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281 -- _ -- 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] HELP! I can't get my Cisco CP-7960G IP hardphone to register on my Asterisk VoIP IP PBX SIP Server with FreePBX GUI
On Thu, 24 Dec 2020, Turritopsis Dohrnii Teo En Ming wrote: 3. secret is 8 char only, must be numeric My my SIP.cnf file from 2007 contains: image_version: P0S3-8-12-00 line1_password: 346cc89a2526255839534c22ad7790c and my notes say my 9760 only allowed up to 31 character passwords. You may find it useful to use tcpdump with '-w' to write the packets to a file and then analyze with sngrep. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281 -- _ -- 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] Can I turn off logging of Invalid packets received by PJSIP?
We get some noise in our Asterisk error file generated by scanners sending invalid invites. Example below (details removed) [2020-12-0702:53:30]ERROR[23370]pjproject:sip_transport.c Error processing 559 bytes packet from UDP *** :PJSIPsyntaxerrorexceptionwhenparsing'Request Line'header on line 1 col 12:INVITEsip:@***SIP/2.0Via:SIP/2.0/UDP0.0.0.0:49990;branch=*Max-Forwards:70From:;tag=*To:Call-ID:***CSeq:1INVITEContact:Content-Type:application/sdpContent-Length:204v=0o=@**:50601626418299INIP40.0.0.0s=pplsipc=INIP40.0.0.0t=00m=audio25282RTP/AVP1006083185101a=rtpmap:0pcmu/8000a=rtpmap:101telephone-event/8000a=fmtp:1010-11 This seems to be a feature added in pjsip. While it's often very useful to log error packers, in our case it just winds up being noise 99.9% of the time. Is their some way to configure Asterisk or PJSIP to not log these packets as errors? Or maybe some way to just filter them out entirely before pjsip even gets it (other than using an IP, which of course changes). -- _ -- 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] some domains resolving issues
Joshua is lieing ASSHOLE Sent from my iPhone > On Sep 30, 2020, at 7:08 AM, Joshua C. Colp wrote: > > >> On Wed, Sep 30, 2020 at 9:06 AM sergio wrote: > >> On 30/09/2020 14:59, Joshua C. Colp wrote: >> >> > latest version of 16 on Ubuntu >> >> 16.12.0~dfsg-1 ? > > I don't use packages. I built a fresh copy of 16.13.0 as it is latest and > used that. > > -- > Joshua C. Colp > Asterisk Technical Lead > Sangoma Technologies > Check us out at www.sangoma.com and www.asterisk.org > -- > _ > -- 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 -- _ -- 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 do I run a command on "Failed to authenticate" ?
On Fri, 11 Sep 2020, sean darcy wrote: I'd like to get an alert if a call fails to authenticate: if "Failed to authenticate" then mail someone the source ip endif How about fail2ban? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281 -- _ -- 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 is upon us
On Sun, 12 Jul 2020, Steve Edwards wrote: So this is a provider issue, not an end user issue and 'June 30, 2021' doesn't sound like 'soon.' If this is legit, why haven't my providers said squat? Seems one of my providers, Vitelity (iax.cc to us old timers), when asked, is not panicking about the imminent end of the world: "Thank you for reaching out. We will not be doing any stir shaken changes until the end of the year. If changes are necessary client side, we will let you know." -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281 -- _ -- 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 Mon, 13 Jul 2020, Jeff LaCoursiere wrote:
Some of us may actually be interested in what you have to offer if you
changed the way you were presenting it. Who is going to base their
business on some list guy with a gmail address?
And can't follow directions and honor the mailing list rules. He got
spanked for this back in May.
I don't claim to understand much about this other than it is supposed to
help reduce spam by making providers accountable for sending calls with
CIDs that are not 'theirs.'
I also don't understand how the OP can sprinkle magic fairy dust on a call
and issue a token to any anonymous user for calls to and from CID/DIDs
they don't control as shown below:
mysql\
--batch\
--database=strshk\
--disable-column-names\
--disable-table\
--execute="call
strshk.stir_shaken_signature('7602588003','7602588003');"\
--host=208.73.232.47\
--password=\
--user=anonymous\
| cut --characters=1-30
eyJhbGciOiJFUzI1NiIsInR5cCI6In
I have no business relationship with the OP or 7602588003 so how does this
'token' add any value?
What am I missing?
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244281
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Re: [asterisk-users] Stir Shaken is upon us
On Sun, 12 Jul 2020, Saint Michael wrote: WORLDWIDE EMERGENCY Again? The code below needs to be executed before any SIP or PJSIP call destined to the US network, or soon no call will terminate. This is called Stir-Shaken, a new law from the FCC. If this is not working the whole Asterisk industry will crash, vanish, be gone. Seen any little chickens lately? According to 'https://www.fcc.gov/call-authentication': "In March 2020, the Commission adopted new rules requiring all originating and terminating voice service providers to implement caller ID authentication using STIR/SHAKEN technological standards in the Internet Protocol (IP) portions of their networks by June 30, 2021." So this is a provider issue, not an end user issue and 'June 30, 2021' doesn't sound like 'soon.' If this is legit, why haven't my providers said squat? Server = 208.73.232.47 So why do you want everybody to send you their call metadata? What's your endgame? Generate leads to call to pitch your service? Poach clients? Sorry if I sound cynical. It's 2020 and I'm fresh out of "F's." -- Thanks in advance, ----- Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281 -- _ -- 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] Redis in place of astdb
On Wed, 8 Jul 2020, Dovid Bender wrote:
we need to use an AGI to connect to redis...
I can execute about 400 AGIs (written in C, only parsing the AGI
environment) per second on a Linode Nanode:
verbose(1,${EPOCH});
agi(null-agi);
...
agi(null-agi);
verbose(1,${EPOCH});
Is the Redis startup (or script startup if you're using a scripting
language) that expensive or are you running very high calls per second?
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244281
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Re: [asterisk-users] Redis in place of astdb
On Wed, 8 Jul 2020, Dovid Bender wrote: Does anyone know of any projects that would allow you to use Redis in place of AstDB? https://langiac.blogspot.com/2018/04/asterisk-dialplan-and-redis-integration.html covers func_redis and Perl Redis. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281 -- _ -- 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] Forbidden call
On Fri, 12 Jun 2020, Jerry Geis wrote: Any chance you can configure the speaker to syslog to your host so you may get a clue why the speaker is rejecting? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281 -- _ -- 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] Any api (agi/ari/ami) equivalent of "core show calls"?
) not null , mode varchar(80) not null default '' , directory varchar(255) not null default '' , application varchar(255) not null default '' , digit char(1) not null default '' , sort varchar(16) not null default '' , formatvarchar(16) not null default '' , stamp timestamp ) ; insert into musiconhold set name = 'default' , directory = '/var/lib/asterisk/moh' , application = '' , mode = 'files' , digit = '' , sort = 'random' , format= '' ; insert into musiconhold set application = '/usr/bin/mpg123 --mono -b 0 -f 8192 -q -r 8000 -s -@ http://streaming.radionomy.com/80sFunkDanceMusic' , mode = 'custom' , name = 'foobar' ; Hope this helps rather than hinders :) -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281 -- _ -- 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] Any api (agi/ari/ami) equivalent of "core show calls"?
On Sat, 13 Jun 2020, Jonathan H wrote: I need to ensure that a MusicOnHold stream is only running when there's a caller on hold and listening.To do that, I need to rewrite and reload the moh.conf file when the caller hangs up IF there are no other callers (ie there's just 1 active call as the caller hangs up), and then rewrite and reload again when there's a new caller. How about ARA to configure MOH and then just update the database. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281-- _ -- 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] Forbidden call
On Thu, 11 Jun 2020, Jerry Geis wrote:
I have a call from a call file:
This looks a lot more like an AMI event than a call file. In any case, it
doesn't matter.
Action: Originate
Async: yes
Channel: SIP/2012
Codecs: ulaw,alaw,gsm
Context: dialout
Exten: callprogress
Priority: 1
Timeout: 2
Variable: SIPADDHEADER="Alert-Info: Ring Answer"
ActionID: 100014
CallerID: Axis < 525 >
The SIP/2012 is a IP Speaker on the computer. The error is:
[Jun 11 15:44:45] WARNING[8132]: chan_sip.c:24191 handle_response_invite: Received
response: "Forbidden"
Why am I getting "Forbidden" ? Its a call file on my server
It's not a call file permissions thing. That would be a different error
and reported by something before chan_sip.
the speaker is directly connected to my server.
How is an IP speaker 'directly connected?' Do you mean directly from the
Ethernet on the speaker to a NIC on the computer? It doesn't matter, just
curious :)
The only thing that will tell you what is going on is the packets. Crank
up 'sip set debug on' and see if that yields a clue.
--
Thanks in advance,
-----
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Re: [asterisk-users] problem with logger: syslog vs. file
On Wed, 3 Jun 2020, Fourhundred Thecat wrote: On 2020-06-03 17:21, Steve Edwards wrote: How about: syslog.local0 = error,verbose,warning no debugging detail. syslog.local0 = debug,error,verbose,warning include debugging detail. currently, the above has no effect on logging. Sorry. I guess I wasn't clear. I wasn't implying that the feature had already been implemented. I was replying to Tony's question 'should it be a configuration option in logger.conf whether they include or omit? if so, what should the default be, if not specified in logger.conf?' I'm suggesting that if the 'debug' log level is specified, the debug detail (function name and line) should be included. Otherwise, not. The 'debug' log level is 'already there' and it seems reasonable to me that if I'm deep enough into logging that I want debug level log messages that I'm probably also interested in logging debug details. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281-- _ -- 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] problem with logger: syslog vs. file
On Wed, 3 Jun 2020, Fourhundred Thecat wrote: On 2020-06-03 12:18, Tony Mountifield wrote: In article <88f96e46-e6bb-a7ef-bebb-5588ef6cd...@gmx.ch>, However, the conversation would then be: should both logging types include line number and function? should both logging types omit them? should it be a configuration option in logger.conf whether they include or omit? if so, what should the default be, if not specified in logger.conf? that's easy! log level should be configurable in config file, not hardcoded. Logging debugging info in production environment is madness. How about: syslog.local0 = error,verbose,warning no debugging detail. syslog.local0 = debug,error,verbose,warning include debugging detail. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281 -- _ -- 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 Thu, 28 May 2020, Saint Michael wrote: My company is one if the six service providers approved. Which part of 'Non-Commercial' do you not understand? The topic may be of general interest. Hawking your wares is not. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281 -- _ -- 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 for asterisk
On Wed, 27 May 2020, Saint Michael wrote: We are in the business of... Then this probably should have been posted on -biz. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281 -- _ -- 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] rotatestrategy = none not working
On Wed, 20 May 2020, David Cunningham wrote:
Thanks for the answer. Since that's what we already have configured, any
idea why it wouldn't work? As I said, when "asterisk -rx 'logger reload'"
is run it still rotates the log file.
Sorry. No clues.
Here's a clue from asterisk-11.3.0-rc1/main/logger.c:
(line 94)
static enum rotatestrategy {
SEQUENTIAL = 1 << 0, /* Original method - create a new file, in
order */
ROTATE = 1 << 1, /* Rotate all files, such that the oldest
file has the highest suffix */
TIMESTAMP = 1 << 2, /* Append the epoch timestamp onto the end of
the archived file */
} rotatestrategy = SEQUENTIAL;
So the default strategy is SEQUENTIAL.
(line 423)
if ((s = ast_variable_retrieve(cfg, "general", "rotatestrategy"))) {
if (strcasecmp(s, "timestamp") == 0) {
rotatestrategy = TIMESTAMP;
} else if (strcasecmp(s, "rotate") == 0) {
rotatestrategy = ROTATE;
} else if (strcasecmp(s, "sequential") == 0) {
rotatestrategy = SEQUENTIAL;
} else {
fprintf(stderr, "Unknown rotatestrategy: %s\n", s);
}
So, since 'none' is not a valid option, the default remains set.
Since the code casually appears the same in 11.17.1, I'll have to
backtrack on my assessment that 11.17.1 doesn't rotate without a more in
depth analysis.
I don't know when 'none' became a valid option, but 17.4.0 has these as
the respective snippets:
static enum rotatestrategy {
NONE = 0,/* Do not rotate log files at all, instead
rely on external mechanisms */
SEQUENTIAL = 1 << 0, /* Original method - create a new file, in
order */
ROTATE = 1 << 1, /* Rotate all files, such that the oldest
file has the highest suffix */
TIMESTAMP = 1 << 2, /* Append the epoch timestamp onto the end of
the archived file */
} rotatestrategy = SEQUENTIAL;
if ((s = ast_variable_retrieve(cfg, "general", "rotatestrategy"))) {
if (strcasecmp(s, "timestamp") == 0) {
rotatestrategy = TIMESTAMP;
} else if (strcasecmp(s, "rotate") == 0) {
rotatestrategy = ROTATE;
} else if (strcasecmp(s, "sequential") == 0) {
rotatestrategy = SEQUENTIAL;
} else if (strcasecmp(s, "none") == 0) {
rotatestrategy = NONE;
} else {
fprintf(stderr, "Unknown rotatestrategy: %s\n", s);
}
So, backport or upgrade?
Also, inquiring minds want to know why the enum is in powers of 2? It's
not like we can set sequential AND timestamp.
--
Thanks in advance,
-----
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https://www.linkedin.com/in/steve-edwards-4244281
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244281
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Re: [asterisk-users] rotatestrategy = none not working
On Wed, 20 May 2020, David Cunningham wrote: Thanks for the answer. Since that's what we already have configured, any idea why it wouldn't work? As I said, when "asterisk -rx 'logger reload'" is run it still rotates the log file. Sorry. No clues. I always use 'syslog' for logging everything. I just did a quickie test to see if I could replicate the behavior. There's about 600 lines of 'diff' between asterisk-11.3.0-rc1/main/logger.c and asterisk-11.17.1/main/logger.c. Maybe 'upgrading' to 11.17 wouldn't be too painful if it would resolve your issue? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281 -- _ -- 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] rotatestrategy = none not working
On Wed, 20 May 2020, David Cunningham wrote: We have an Asterisk 11.3 server where we want log rotation handled purely by Linux's logrotate, and not by Asterisk. To this end we've configured the [general] action of /etc/asterisk/logger.conf with: rotatestrategy = none However, an "asterisk -rx 'logger reload'" still rotates the log files. Does anyone know why? I had to hunt, but I found an 11.17.1 system :) 'none' does not rotate a log file on this host. Here's my logger.conf: ; Created by makefile on 2020-05-19 at 23:05:08 ; from /source/src/obl-server/logger.conf.pre [general] rotatestrategy = none [logfiles] /tmp/ast-log-test = debug,dtmf,error,event,notice,verbose,warning ; (end of /etc/asterisk/obl/logger.conf) -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281 -- _ -- 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] Perl AGI: read variable with quotes
On Fri, 24 Jan 2020, Steve Edwards wrote:
2) How about doing 'GET FULL VARIABLE' in your Perl script?
Sorry. After a couple more cups of tea I think this was a bit vague.
Try whatever call/method in your library that does 'GET FULL VARIABLE' on
'${PJSIP_HEADER(read,P-Asserted-Identity)}' in your AGI.
--
Thanks in advance,
-
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Re: [asterisk-users] Perl AGI: read variable with quotes
On Fri, 24 Jan 2020, Benoit Panizzon wrote:
I have stumbled of this problem.
I need the P-Asserted-Identity header in an AGI scrip.
In the Dial-Plan I do:
same => n,Set(PAI=${PJSIP_HEADER(read,P-Asserted-Identity)})
In the AGI I do:
my $pai = $AGI->get_variable(PAI);
This works fine, unless the PAI contains quotes:
P-Asserted-Identity:
I get "" in the variable $pai.
P-Asserted-Identity: "John Doe"
Is getting me $pai containing just "John".
Anyone a clue how I could get the whole header?
1) Does the PAI channel variable contain the full header? Try 'verbose(PAI
= ${PAI})' or something similar.
2) How about doing 'GET FULL VARIABLE' in your Perl script? You can set
the channel variable PAI in the AGI if needed back in the dialplan.
--
Thanks in advance,
-----
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https://www.linkedin.com/in/steve-edwards-4244281
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Re: [asterisk-users] USB dahdi fxo ?
On Fri, 13 Dec 2019, sean darcy wrote: I'm moving asterisk to a laptop, so can't use the dahdi board. Is there any supported USB dahdi device ? I see the Sangoma USBfxo device, but the dahdi driver no longer supports it. Anything else ? How about something like the ancient Ethernet based Sipura 3000? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Two sip extensions
On Fri, 19 Jul 2019, Jerry Geis wrote:
I was not aware of the (+) format... basically "add" to the general
section.
How far back does that go? T o 1.4.X ?
I don't know, but I checked a sip.conf from 1.2 (2012ish?) and I was using
it then.
Is there a documentation piece on that ?
I'm sure there is, I just don't know where :)
Another cool configuration file feature is templates (an exclamation mark
instead of a plus sign). It lets you define common 'snippets' once and
include them in each context as needed.
Here's an example from that same (1.2 based) project:
; templates
[digit-timeout](!)
exten = t,1,verbose(1,[${EXTEN}@${CONTEXT}])
exten = t,n,goto(${CONTEXT},s,1)
[h](!)
exten = h,1,verbose(1,[${EXTEN}@${CONTEXT}])
exten = h,n,goto(settle-card,s,1)
[i](!)
exten = i,1,verbose(1,[${EXTEN}@${CONTEXT}])
exten = i,n,goto(${CONTEXT},s,1)
[s](!)
exten = s,1,verbose(1,[${EXTEN}@${CONTEXT}])
[max-timeout](!)
exten = T,1,verbose(1,[${EXTEN}@${CONTEXT}])
exten = T,n,goto(max-time,s,1)
[x](!)
exten = _x.,1, verbose(1,[${EXTEN}@${CONTEXT}])
; authorized the card
[auth-card](h,i,s,max-timeout,digit-timeout)
exten = s,2,agi(write-cdr)
exten = s,n,set(PRODUCT=${CONTEXT})
exten = s,n,set(PER-MINUTE=0)
exten = s,n,set(PREAMBLE=${CUSTOMER}/menu/m1101)
exten = s,n,
agi(auth-card,${AUTH-FLAGS},${DEBUG-MODE},${VERBOSE-MODE})
exten = s,n,goto(theme,s,1)
The templates are inserted into the auth-card context when the file is
parsed. I don't have a 1.2 host running anymore, but a 'show dialplan
auth-card' (1.2) would look something like:
[auth-card](h,i,s,max-timeout,digit-timeout)
exten = T,1,verbose(1,[${EXTEN}@${CONTEXT}])
exten = T,n,goto(max-time,s,1)
exten = h,1,verbose(1,[${EXTEN}@${CONTEXT}])
exten = h,n,goto(settle-card,s,1)
exten = i,1,verbose(1,[${EXTEN}@${CONTEXT}])
exten = i,n,goto(${CONTEXT},s,1)
exten = s,1,verbose(1,[${EXTEN}@${CONTEXT}])
exten = s,2,agi(write-cdr)
exten = s,n,set(PRODUCT=${CONTEXT})
exten = s,n,set(PER-MINUTE=0)
exten = s,n,set(PREAMBLE=${CUSTOMER}/menu/m1101)
exten = s,n,
agi(auth-card,${AUTH-FLAGS},${DEBUG-MODE},${VERBOSE-MODE})
exten = s,n,goto(theme,s,1)
exten = t,1,verbose(1,[${EXTEN}@${CONTEXT}])
exten = t,n,goto(${CONTEXT},s,1)
Templates are also useful in other configuration files like sip.conf to
define 'classes' of parameters like 'dial-in-agent' or 'supervisor' that
can be included in endpoint definitions to reduce clutter, increase
consistency, and reduce maintenance.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244281
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Re: [asterisk-users] Two sip extensions
On Thu, 18 Jul 2019, Joshua C. Colp wrote: On Thu, Jul 18, 2019, at 10:10 AM, Jerry Geis wrote: I have two SIP extensions defined in sip.conf register => 4450@10.20.1.1/4450 [4450] type=friend username=4450 host=10.20.1.1 allow=all dtmfmode=inband context=incoming register => 4451@10.20.1.1/4451 [4451] type=friend username=4451 host=10.20.1.1 allow=all dtmfmode=inband context=incoming "register" lines have to be under the general section. They can't be within a friend/peer/user. I format my entries in sip.conf like below to keep everything related to the endpoint together. ; 4450 [general](+) register= 4450@10.20.1.1/4450 [4450] allow = all context = incoming dtmfmode= inband host= 10.20.1.1 type= friend username= 4450 ; 4451 [general](+) register= 4451@10.20.1.1/4451 [4451] allow = all context = incoming dtmfmode= inband host= 10.20.1.1 type= friend username= 4451 I like to keep the parameters in each stanza sorted and 'tabbed out' to make it easier to compare stanzas and because I'm just that kind of guy :) -- Thanks in advance, ----- Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281 -- _ -- 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?
hw-- I see this kind of behavior when the certificate expires... you've probably checked this, but sometimes we miss little details like that. murf On Fri, Jul 5, 2019 at 1:14 PM hw wrote: > On 7/5/19 10:50 AM, Doug Lytle wrote: > > 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 > > > > > > What would be the point in making this even more complicated? > > Today all of a sudden the certificate couldn't be verified anymore even > without restarting asterisk. How is it possible that a certificate > which was fine for 10 hours and 18 minutes suddenly can not be used > anymore? > > -- > _ > -- 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 -- Steve Murphy ParseTree Corporation 57 Lane 17 Cody, WY 82414 ✉ murf at parsetree dot com ☎ 307-899-0510 -- _ -- 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] Find out which key ended recording?
On Fri, 7 Jun 2019, David Cunningham wrote:
We're using Perl and so far I haven't found an equivalent there.
On Thu, 6 Jun 2019, Steve Edwards wrote:
I'm not much of a Perl programmer...
But you should never turn down an opportunity to develop your skills :)
Try something like:
my $result = $AGI->record_file(
'/tmp/foo'# filename
, 'wav' # format
, '#*0123456789'# escape digits
, '5000'# timeout
);
$AGI->verbose('result = ' . $result, 0);
Which results in:
AGI Rx << RECORD FILE /tmp/foo wav #*0123456789 5000
AGI Tx >> 200 result=50 (dtmf) endpos=0
AGI Rx << VERBOSE "result = 50"
when '2' is pressed.
--
Thanks in advance,
-
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Re: [asterisk-users] Find out which key ended recording?
On Fri, 7 Jun 2019, David Cunningham wrote: What language is that please? C. We're using Perl and so far I haven't found an equivalent there. I'm not much of a Perl programmer, but I'd guess something like: $AGI->result or $AGI->lastresult might yield clues. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281 -- _ -- 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] Find out which key ended recording?
On Fri, 7 Jun 2019, David Cunningham wrote:
We're using Perl and so far I haven't found an equivalent there.
On Thu, 6 Jun 2019, Steve Edwards wrote:
I'm not much of a Perl programmer...
But you should never turn down an opportunity to develop your skills :)
Try something like:
my $result = $AGI->record_file(
'/tmp/foo'# filename
, 'wav' # format
, '#*0123456789'# escape digits
, '5000'# timeout
);
$AGI->verbose('result = ' . $result, 0);
Which results in:
AGI Rx << RECORD FILE /tmp/foo wav #*0123456789 5000
AGI Tx >> 200 result=50 (dtmf) endpos=0
AGI Rx << VERBOSE "result = 50"
when '2' is pressed.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244281
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Re: [asterisk-users] Find out which key ended recording?
On Fri, 7 Jun 2019, David Cunningham wrote:
We have a need to record audio and allow the user to press any DTMF key
to end the recording. Currently we're using the AGI command "record
file" which does allow us to specify which DTMF keys can end the
recording.
However we also need to know which key actually ended the recording.
Note that only allowing # or * to end the recording won't work for us.
Does anyone know how we can tell which key ended the recording? Thanks
in advance for any help.
Here's a snippet from one of my AGIs:
// record the voice
exec_agi("RECORD FILE"
" %s" // filename
" wav"// format
" #*1234567890"
// escape digits
" %d000" // timeout in ms
" BEEP" // BEEP
, recorded_path
, recording_limit
);
// should we abort?
if ('*' == agi_environment.result)
{
agi_set_variable("STATUS", "*");
exit(EXIT_SUCCESS);
}
// are we finished?
if ('#' == agi_environment.result)
{
break;
}
Looks like agi_environment.result is your Huckleberry.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244281
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Re: [asterisk-users] Play Music While Processing AGI Script
On Tue, 14 May 2019, Alexander Perkins wrote:
Hi All. I have a question - I have an AGI script that may run for 10 seconds,
or it may run for 60 seconds while an agent becomes available (agents are
geographically
dispersed). Is there a way to have the music play in the background while the
AGI scripts executes? When the AGI script finishes, then the music should also
finish.
I tried this, but the music needs to finish before moving on to step 4 and
execute the script.
exten => _NXZNXX,1,Answer()
exten => _NXZNXX,2,MusicOnHold()
exten => _NXZNXX,3,AGI(SetRecordingID.php,${UNIQUEID})
Create a separate thread in your AGI to play a small segment (5 to 10
seconds) of music in a loop. At the end of each 'play' check to see if the
AGI is ready to exit.
Off topic, but you can make maintenance of your dialplan easier if you
write it like:
exten = _nxznxx,1, answer()
same = n, musiconhold()
same = n, agi(SetRecordingID.php,${UNIQUEID})
The 'whitespace' and 'lowcasing' is just my personal preference. I would
also use 'getopt/longopts' to parse the command line so you can have
meaningful (long) options and are not dependent upon passing arguments in
a particular order.
--
Thanks in advance,
-----
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244281--
_
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Re: [asterisk-users] Sending SMS and SIM card
On Tue, 23 Apr 2019, bilal ghayyad wrote: Is it possible to send SMS from asterisk? Using DAHDI or using what is possible? You can use an SMS provider like Twillio. And, is there a card that can be fixed in the machine and insert the SIM card in this card to be used for GSM calls and sending SMS through asterisk? Through which channel? Is it DAHDI or something else? I've never used an internal card, but what you're looking for is a GSM gateway. I used a Goip32 (32 SIMs, 32 channels) a couple of years ago. It is an external box you hang on your network via Ethernet. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281 -- _ -- 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] Forking AGI or GoSub
On Wed, 10 Apr 2019, Dovid Bender wrote: I have an AGI that can sometimes take time complete. I don't want the dialplan to be held up by the agi. Is there any way to call it and have Asterisk continue with the dialplan? On Wed, 10 Apr 2019, Dovid Bender wrote: I have an AGI that can sometimes take time complete. I don't want the dialplan to be held up by the agi. Is there any way to call it and have Asterisk continue with the dialplan? I had a situation that required this functionality -- processing a credit card could take a second or two and we didn't want 'dead air' for our user experience. I created a pthread to play 'Please hold on while we process your card and get ready for a good time...' while the main program continued with the card authorization. Most of the time the auth completed before the audio finished so it appeared to be instantaneous to the caller. The only caveat is to not interact (stdin/stdout) with Asterisk until 'stream file' in the thread completed. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281 -- _ -- 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] AMI mulitple calls quickly
On Mon, 11 Mar 2019, Jerry Geis wrote: If I use the AMI interface to originate a call, close the connection, open another connection etc...This works. but is slow... Would opening multiple AMI connections be an option? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281 -- _ -- 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] Outbound caller ID ignored
On Sun, 13 Jan 2019, Mitch Claborn wrote: Setting the outbound caller ID works fine on our PRI (T1) lines, but does not work on our local POTS lines. No errors in the logs, the new caller ID just seems to be ignored. Should I expect it to work on the analog lines? Nope. Setting caller ID is not a POTS feature. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281 -- _ -- 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
