| 00:00.04 | Cyrillax | acovrig: And you could ALSO use AGI to play a message to the Caller saying, 'we are now transfering you...' so the caller isn't sitting there while the queue member listens to the message from the Macro. |
| 00:00.35 | Cyrillax | acovrig: the AGI parameter to Queue runs on the Caller's channel, while macro/gosub run on the Called party (queue member)'s channel. |
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| 00:14.58 | acovrig | wow, I shoulda read my logs, I did Queue(macro-name) when it shoulda been Queue(name) (with [macro-name]) - thanks Cyrillax and [TK]D-Fender |
| 00:15.26 | [TK]D-Fender | You're welcome |
| 00:17.20 | acovrig | My queue has a few extensions listed in the queue.conf file and it doesn't seem to ring those, but if I remove it and `queue add member` then it works; I'm guessing there's a flag in the queue.conf I need to change? |
| 00:20.23 | [TK]D-Fender | We'd have to look at it |
| 00:20.27 | [TK]D-Fender | can't judge blind |
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| 01:10.50 | Kobaz | oh man, oh man |
| 01:10.50 | Kobaz | so close |
| 01:11.28 | Kobaz | It's almost like it actually works |
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| 01:55.24 | acovrig | How do I pass an arg to a macro in a Queue? Queue(801,,,,,,queue_support^${reason},,,)? I get No such context 'macro-queue_support^test reason' for macro 'queue_support^test reason'. |
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| 03:02.49 | Cyrillax | acovrig: Try queue_support(${reason}) |
| 03:05.45 | acovrig | Cyrillax: it does the same thing, but with parenthesis "No such context macro-queue_support(test reason)" |
| 03:06.40 | Cyrillax | is your macro context named "macro-queue_support" ? |
| 03:07.35 | Cyrillax | What version of asterisk are you on? |
| 03:08.52 | acovrig | I have "[macro-queue_support]" defined above this in the dialplan; I'm on 13.22.0 |
| 03:22.12 | [TK]D-Fender | Show the full config and attempt |
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| 03:42.32 | Cyrillax | acovrig: Just use GoSub instead (the next argument in Queue) and use channel variable inheritance (Set(_TESTVAR=test)) to get the arguments |
| 03:42.59 | Cyrillax | Macro is deprecated anyway |
| 03:52.59 | [TK]D-Fender | Gosu is what you want |
| 03:53.02 | [TK]D-Fender | gosub |
| 03:53.16 | [TK]D-Fender | certainly the way... |
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| 10:04.39 | DonAlex | Uggg.. having a brain fart moment. Years ago I remember I used to dial an external number to an extension form the console using orginate. I know that has been moved to channel originate.. but I am still having trouble with the flags.. Can someone explain to me what is wrong ? It cannot seem to find the trunk name though that is the name it identifies as.. |
| 10:06.22 | DonAlex | <PROTECTED> |
| 10:06.45 | DonAlex | what is wrong with that .. when the trunk name is correct ? |
| 10:07.20 | Cyrillax | what error are you getting |
| 10:07.32 | Cyrillax | show us the logs please, aterisk -rvvvvvvvvvvv |
| 10:07.35 | sibiria | channel format looks a bit off for being pjsip, or maybe it's just me being off |
| 10:07.45 | sibiria | PJSIP/123456@trunk |
| 10:07.58 | Cyrillax | yea that looks right ^ |
| 10:08.09 | DonAlex | hmmm |
| 10:08.22 | DonAlex | lemme try it the other way around then |
| 10:08.25 | sibiria | chan_sip uses the SIP/trunk/12345 format |
| 10:08.34 | Cyrillax | ^ |
| 10:09.28 | DonAlex | Could be.. I am dragging it up from a long time ago ;) |
| 10:12.31 | Cyrillax | i dont even think pjsip was out yet when the switch from originate to channel originate happened XD |
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| 12:00.55 | DonAlex | Sorry to taken so long been doing something else.. but it does not complain but it does not set up the call either ? All I get is this |
| 12:00.58 | DonAlex | PJSIP/VOIPFONE200-00000052 is making progress |
| 12:01.44 | DonAlex | But no other errors.. not bad number not cannot connect and the extension does not ring ? |
| 12:02.21 | DonAlex | hannel originate PJSIP/xxxxxxxxxxxx@VOIPFONE200 extension 302@from-internal |
| 12:02.37 | DonAlex | Called xxxxxxxxxxxxx@VOIPFONE200 |
| 12:02.47 | DonAlex | <PROTECTED> |
| 12:36.48 | Cyrillax | DonAlex: Make sure you're running asterisk with # asterisk -rvvvvvvvvvvvvv |
| 12:37.05 | DonAlex | Well I was.. |
| 12:37.11 | Cyrillax | DonAlex: Then when you're in the * CLI, run:> pjsip set logger on |
| 12:37.17 | DonAlex | ok. |
| 12:37.30 | Cyrillax | and then try to place the call, and then paste the FULL log to pastebin and send us the link |
| 12:37.48 | DonAlex | I shall with the number redacted |
| 12:37.51 | DonAlex | give me 2 |
| 12:37.56 | Cyrillax | yes of course |
| 12:42.50 | DonAlex | Cyrillax: Here you go. https://pastebin.com/5xs2nu0m |
| 12:43.25 | DonAlex | Of course you are getting the keep alive messages from the extensions as well. but I don't see any errors being thrown? It just sort of quietly does. |
| 12:43.29 | DonAlex | dies |
| 12:45.31 | file | "SIP/2.0 603 No Funds" your call is going out with anonymous callerid, depending on the provider they may not associate it with your account |
| 12:47.48 | DonAlex | hmmm |
| 12:48.25 | file | you may need to set "from_user" and "from_domain" on the endpoint, but I have no experience with said VoIP provider |
| 12:48.35 | DonAlex | Well the trunks are set up and registered.. do I some how need to pass on a user id as well? That does not seem right as extensions can dial out normally ok? |
| 12:49.05 | DonAlex | Granted as I am originating from the console maybe it is filling out some default values ? |
| 12:49.08 | file | extensions dialing out may be presenting callerid they like? |
| 12:49.19 | file | from the CLI has anonymous callerid, you can see it in the "From" header in the INVITE |
| 12:49.42 | DonAlex | Not sure really.. I was told not to pass on any caller id to the VOIP provider they do that for me. |
| 12:50.06 | DonAlex | is there any way to insert an id from the CLI ? |
| 12:50.20 | file | no, not without going through the dialplan |
| 12:50.43 | DonAlex | So bottom line is I cannot originate calls like I used to maybe 15 years ago from the console? ;) |
| 12:50.57 | sibiria | :P |
| 12:51.10 | sibiria | stop lingering behind! why do you hate society's progress?! |
| 12:51.40 | file | well, you could examine a working case and non-working case and identify the difference |
| 12:52.12 | file | the From header is required, so something has to be there - you can't just have it be empty and contain no callerid |
| 12:52.33 | file | like I said, your provider may need something explicitly in there |
| 12:52.51 | file | and based on your own configuration elsewhere as a side effect it may end up there when dialing out normally |
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| 13:04.18 | DonAlex | Fairy nuff |
| 13:07.36 | sibiria | is that a controlled substance? i've never heard about fairy nuff abuse |
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| 13:39.18 | Cyrillax | no She gives you Nuff's certificate |
| 13:40.36 | Cyrillax | DonAlex: sorry fell asleep for a little bit, yea saw the 603 also, if you don't want to connect an endpoint, you could make another extension in the dialplan that sets the CallerID and any other vars, and then Dial the PJSIP channel from that extension. Then you could do channel origiante Local/newexten@context exten 302@from-internal |
| 13:41.23 | Cyrillax | extension* |
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| 15:41.14 | _ganapathi_ | https://paste.linux.community/view/e9c92a86 |
| 15:41.29 | _ganapathi_ | am trying to play gsm file through asterisk 16 version |
| 15:42.01 | _ganapathi_ | but second file (1111200864.gsm) is not playing |
| 15:42.23 | _ganapathi_ | getting error as " gsm_read: Short read of gsm data (expected 33 bytes, read 24): No such file or directory" |
| 15:42.31 | _ganapathi_ | Invalid GSM data |
| 15:43.13 | _ganapathi_ | anybody help me what's the pbm |
| 15:43.45 | Cyrillax | sounds like the gsm file is corrupt/empty |
| 15:43.48 | igcewieling | I don't see the ACTUAL command. |
| 15:44.05 | [TK]D-Fender | ^ |
| 15:44.09 | [TK]D-Fender | as well |
| 15:44.14 | Samot | Or you know, not the same format as the other. |
| 15:44.27 | Samot | As soxi kicks out different output for each |
| 15:44.33 | Cyrillax | yea that |
| 15:44.42 | igcewieling | For example are you using Playback(1111200864.gsm) which won't work or are you using Playback(1111200864) which should work if the file is in the default location. |
| 15:45.29 | _ganapathi_ | gsm file is playing on player and after converting using sox then also playing. |
| 15:45.48 | Samot | Asterisk likes its files encoded a certain way |
| 15:46.02 | Samot | Just because it plays on an external player doesn't mean it will play in Asterisk. |
| 15:46.21 | Samot | As the external player can support encoded formats Asterisk doesn't. |
| 15:46.21 | _ganapathi_ | igcewieling: am playing without extension. asterisk gathered file and trying to play as well |
| 15:46.26 | _ganapathi_ | am getting some noise |
| 15:46.33 | Samot | Asterisk likes its files encoded a certain way |
| 15:46.50 | seanbright | try renaming 1111200864.gsm to 1111200864.WAV |
| 15:47.38 | _ganapathi_ | in that case am getting error as "Not a supported wav file format (49). Only PCM encoded, 16 bit, mono, 8kHz/16kHz files are supported with a lowercase '.wav' extension." |
| 15:48.36 | _ganapathi_ | small wav or CAPITAL wav ? |
| 15:49.10 | seanbright | capital WAV |
| 15:49.31 | igcewieling | .wav is ULAW with MS header. .WAV is GSM with a MS header. |
| 15:49.42 | seanbright | eh |
| 15:51.26 | seanbright | wave is a container format and can contain audio encoded in a variety of formats |
| 15:51.45 | seanbright | but asterisk makes decisions based on extension, so it's limited to PCM (wav) and GSM (WAV) |
| 15:52.06 | seanbright | i think saying "with MS header" is overly simplistic and not technically correct |
| 15:53.09 | Samot | How was the working .gsm file generated versus the non working .gsm file? |
| 15:53.11 | igcewieling | It is overly simplistic |
| 15:53.48 | _ganapathi_ | actually two different file which i got it from client. |
| 15:55.13 | _ganapathi_ | after renaming to CAPITAL WAV then it's working.. |
| 15:55.53 | Samot | So they are encoded differently. |
| 15:56.46 | seanbright | one is raw GSM data |
| 15:56.57 | Samot | Correct. |
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| 15:57.01 | seanbright | the other is a WAVE that contains GSM audio |
| 15:57.12 | seanbright | so you can either convert it with asterisk |
| 15:57.32 | seanbright | file convert /path/to/1111200864.WAV /path/to/1111200864.gsm |
| 15:57.40 | seanbright | or just leave it as .WAV and move on with your life |
| 16:00.11 | _ganapathi_ | thanks. |
| 16:00.15 | igcewieling | Hmm..T-Mobile completed its purchase of Sprint. Time to switch carriers. |
| 16:03.45 | Cyrillax | nooooooooooooo |
| 16:06.42 | igcewieling | Sprint has a habit of pulling defeat from the jaws of victory. |
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| 16:13.02 | Samot | Yeah but you're missing the other key factor of all this. |
| 16:13.17 | Samot | The FCC wants at least 4 nation wide wireless providers. |
| 16:13.25 | Samot | This would make 3. |
| 16:13.30 | igcewieling | Apparently not, as they approved the merver. |
| 16:13.50 | Samot | Because DISH bought Boost Mobile from T-Mobile. |
| 16:13.56 | Samot | And will be the 4th provider. |
| 16:14.11 | Samot | They have a few years to ramp up. |
| 16:14.20 | Samot | Until then they are using T-Mobile's network. |
| 16:16.38 | igcewieling | I guess we'll see in 5 - 10 years. |
| 16:17.02 | igcewieling | Regardless, Sprint is a shit company with shit management. |
| 16:17.30 | igcewieling | And has been for 25 years. |
| 16:22.57 | Samot | Just 25 years? |
| 16:30.09 | igcewieling | At least 8-) |
| 16:30.35 | Samot | Well they've been around way long. |
| 16:30.53 | Samot | They where one of the results of the Bell Breakup. |
| 16:31.08 | Samot | AT&T, MCI, Sprint the three interexchanges. |
| 16:31.10 | igcewieling | *nod* I never dealt with them when they only did Long Distance, only once they started offering T-1s. |
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| 17:15.41 | igcewieling | Looks like I might be doing FreePBX in virtual machines soon. Our current hosted software provider Bicom says the product we have is no longer supported. |
| 17:15.58 | igcewieling | Bicom is a GUI for Asterisk, similar to FreePBX, but it is commercial only. |
| 17:17.13 | igcewieling | Bicom's PBXware runs on Gentoo Linux only. I really hate it. |
| 17:17.44 | file | I always forget Bicom exists |
| 17:19.21 | igcewieling | I wish I could. Back in 2010 it was the best multi-tennant Asterisk based system, but FreePBX and virtualization technology have come a long way since then. |
| 17:24.01 | Samot | That's rather funny. |
| 17:24.52 | igcewieling | Now they want to sell us something which sounds exactly what I'm considering, but with PBXware virtualized instead of FreePBX. |
| 17:25.29 | Samot | Wouldn't it be multi-tenant? |
| 17:26.23 | igcewieling | there would be no point in a multi-tennant version if you setup individual virtual machines for each customer. |
| 17:26.33 | Samot | Seriously? |
| 17:26.47 | igcewieling | As least for us. |
| 17:26.54 | Samot | So I have 40+ tenants on a single system that I manage... |
| 17:27.16 | Samot | But it would be better to have 40 individual sysetsms to manage? |
| 17:27.21 | Samot | systems* |
| 17:29.12 | igcewieling | yes and no. Yes, it would be better because you don't have any concern over some little bug giving tennants access to other tennants in whatever multi-tennant support you added. No, becuase you have more OSs to keep updated and more FreePBXs to be updated. |
| 17:29.56 | Samot | OK. |
| 17:30.25 | Samot | So it's better because it stops a bug that could break something. A bug you would need to update/patch on system for. |
| 17:30.34 | igcewieling | Also yes, because we would not have some stupid Gentoo system I'd be expected to managed. |
| 17:30.49 | Samot | Versus a bug that is on 40 systems that breaks something and you have to update 40 systems. |
| 17:32.55 | igcewieling | I simply cannot see how adding a bunch of code to FreePBX to make the current system be multi-tennant is better than using standard FreePBX. |
| 17:33.14 | Samot | Well.. |
| 17:33.22 | Samot | I didn't add a bunch of code to FreePBX. |
| 17:33.59 | igcewieling | How did you add support to allow individual customers to log into the GUI without allowing them access to other tennants? |
| 17:34.21 | Samot | By having a different system. |
| 17:34.32 | Samot | Because they need access to other things. |
| 17:34.47 | igcewieling | So you created a GUI for them? |
| 17:34.50 | Samot | Like SMS, XMPP and other services. |
| 17:34.53 | Samot | Yes. |
| 17:35.02 | igcewieling | I'm not going to do that. |
| 17:35.07 | Samot | I know. |
| 17:35.08 | igcewieling | I did it once. never again. |
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| 17:37.12 | Samot | I run my own XMPP servers. |
| 17:37.34 | Samot | I setup my own SMS "gateway" to handle HTTP API calls. |
| 17:40.53 | Samot | However, I do pony up for a commercial fax platform because it's way better than using Asterisk as a solution. |
| 17:42.05 | igcewieling | With g729 support included with the FreePBX distro I also don't need to use trancoding cards for our hosted customers. |
| 17:42.43 | Samot | Well I also don't do g729. |
| 17:43.28 | igcewieling | 10 years ago we had a lot of people on T-1 or even lower so we needed a highly compressed codec. These days almost everyone has several Mbps to spare for voice. |
| 17:43.58 | Samot | Right. |
| 17:44.12 | Samot | 10 years ago T1 customers may have been on g729. |
| 17:46.13 | igcewieling | When we upgraded our Verizon circuit a few years ago and increased our max usage rate, I switched everyone with faster than t-1 service to ulaw or (where possible) g722. |
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| 19:12.47 | seanbright | are there any decent softphones for windows anymore that don't have ads or require me to give over my e-mail address? |
| 19:12.53 | seanbright | free softphones |
| 19:13.13 | igcewieling | All softphones suck. |
| 19:13.39 | seanbright | that's what i figured |
| 19:13.39 | MLC | zoiper is decent on other platforms. (not great, but decent). Haven't tried it on Windows. |
| 19:13.54 | seanbright | ads |
| 19:13.56 | seanbright | pretty sure |
| 19:15.07 | seanbright | microsip looks interesting and really ugly |
| 19:17.12 | seanbright | oh yeah, now that is an open source GUI if i have ever seen one |
| 19:17.18 | seanbright | https://www.microsip.org/ |
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| 21:03.24 | drmessano | "All softphones suck" <--- ok boomer |
| 21:28.13 | Samot | LOL |
| 21:28.41 | Samot | Bria was a dream for this whole WFH thing we gots going on. |
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| 21:38.38 | MLC | is there a dial plan function or variable that will indicate if the call has been answered or not? |
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| 21:50.27 | Cyrillax | MLC: Do you mean like ${DIALSTATUS} |
| 21:51.19 | Cyrillax | seanbright: Yes microsip is the only client i use on windows now, so far from the jitsi/linphone era of a few years ago :D |
| 21:52.23 | MLC | no, when a call comes in to a context in asterisk, has that call been "answered" or not, e.g. with Answer() |
| 21:54.08 | MLC | ${CDR(answer)} seems to work, but not sure if that is reliable or not |
| 21:57.01 | file | ${CHANNEL(state)} is probably a thing |
| 21:59.20 | MLC | file: nice, thanks |
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