| 00:01.28 | DannyA | which one does? |
| 00:02.28 | drmessano | heh |
| 00:02.34 | drmessano | It's right there on the page |
| 00:12.58 | sibiria | having the 'h' extension defined suffices, too |
| 00:13.31 | sibiria | caveat: needed for each context |
| 00:23.48 | ritztech | i havent touched asterisk in quite a few years . which GUI nowadays is easier or ppl tend to go to (i liked PIAF) Elastix cause of possible custom plugins... is there one with API and possible call flow designer API inbound (i like how twilio studio is im not sure if theres something like that in the open source PBX world) |
| 00:42.05 | DannyA | @file @sibiria thank you! |
| 00:42.18 | DannyA | @drmessano sometimes the language is confusing, esp for beginners (shrug) |
| 01:30.35 | velix | Can I call people via "originate" to connect them to a conference? |
| 01:30.48 | [TK]D-Fender | .... |
| 01:30.55 | [TK]D-Fender | What is unclear about Originate? |
| 01:31.00 | [TK]D-Fender | You dump them into the dialplan. |
| 01:31.03 | [TK]D-Fender | that's it. |
| 01:31.10 | [TK]D-Fender | what that dialplan does is up to you |
| 01:31.24 | velix | Ok ;) |
| 01:31.26 | [TK]D-Fender | You want to dump them into a conference.. then dump them into a conference |
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| 01:44.33 | velix | Asterisk is really nice. I've done that in a week what our IT department couldn't do in a year. |
| 01:44.42 | velix | Thanks for not kicking me, guys. |
| 01:45.56 | [TK]D-Fender | Perhaps they were just lacking motivation... |
| 01:47.25 | velix | of kicking me? |
| 01:48.53 | [TK]D-Fender | Sure ... that too :p |
| 02:25.59 | velix | It works. |
| 02:26.02 | velix | Yipee. |
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| 02:41.23 | drmessano | Yippee ki-yay |
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| 05:42.03 | Testeree | I am looking for a way to get multi-channel audio stream in a conference of three people, where one person can listen to the audio of the other two person without being mixed to a single stream(stereo audio where left channel is one person and the right channel is the other). Is it possible in asterisk. I have found |
| 05:42.03 | Testeree | binaural_active(https://wiki.asterisk.org/wiki/display/AST/Asterisk+15+Configuration_app_confbridge#Asterisk15Config...) not sure if this is what i am looking for. There is also sfu but it seems to be supported only for video. Thanks. |
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| 11:10.44 | velix | Anyone with an idea, why my call is anonymous internally? It's working for AMI and Callfiles, but actually the extension should handle it even via channel originate? https://bpaste.net/raw/IEYA |
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| 11:41.18 | jamesc | I have a sip user telling me that the contact header in the 200OK is missing the user portion, it only contains IP address no +44...@IP address. From what I read in the rfc the 200OK does not have to have a user portion. Is this true? |
| 11:41.34 | file | yes it is true |
| 11:41.34 | jamesc | https://tools.ietf.org/html/rfc3261#section-8.1.1.8 |
| 11:41.49 | file | but some people, for whatever reason, put meaning into the Contact header when there shouldn't be |
| 11:43.30 | jamesc | @file Thanks. |
| 11:44.42 | file | there is https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Configuration_res_pjsip#Asterisk13Configuration_res_pjsip-endpoint_contact_user though for configuring a user, and also https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Configuration_res_pjsip#Asterisk16Configuration_res_pjsip-global_use_callerid_contact for newer versions |
| 11:47.09 | DanFromUK | Where I can find the description of the AVAILSTATUS values set by ChanIsAvail() ? |
| 11:47.54 | DanFromUK | Specifically, I'm trying to make use of PJSIP_DIAL_CONTACTS, but need to figure out how to combine that with ChanIsAvail, and not doing to well at it. |
| 11:48.52 | file | it's the device state, so the list at https://wiki.asterisk.org/wiki/display/AST/Asterisk+17+Function_DEVICE_STATE would be the list |
| 11:49.42 | DanFromUK | It's returning an integer. I assume the device states listed on that link are constants? |
| 11:50.28 | file | ah if integer then the device state header file would have the list |
| 11:50.43 | DanFromUK | ok. thanks. i'll take a look |
| 11:51.00 | file | https://github.com/asterisk/asterisk/blob/master/include/asterisk/devicestate.h#L52 unknown would be 0, and then incrementing for each |
| 11:51.30 | DanFromUK | Whats the right way to ChanIsAvail() with a PJSIP endpoint with multiple contacts? |
| 11:51.55 | file | never done it, 'nor seen it done |
| 11:52.21 | file | but really you just need to call ChanIsAvail on the endpoint... without specifying contacts... |
| 11:52.37 | file | if you are just wanting to see if it is possible to call or get the device state or something |
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| 12:04.46 | DanFromUK | perfect. thanks for replying so quickly. the world's gone nuts. it's nice to see that #asterisk is still as friendly/reliable as always! stay safe and healthy! |
| 12:05.21 | file | has worked from home for 15+ years and avoided going out in winter, so it's very much business as usual for him |
| 12:11.55 | DanFromUK | :) |
| 12:12.33 | velix | Anyone with an idea, why my call is anonymous internally? It's working for AMI and Callfiles, but actually the extension should handle it even via channel originate? https://bpaste.net/raw/IEYA |
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| 12:33.46 | sibiria | i'm also 100% remote, since 5 years. no real change for me either now that my employer sent everyone home a few weeks ago |
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| 13:07.23 | velix | bangs his head to the table |
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| 14:02.09 | velix | kharwell: Hi, never seeen you before here. |
| 14:02.48 | kharwell | velix: greetings |
| 14:03.32 | velix | kharwell: Be aware, I'm a support vampire, as Samot called me. |
| 14:03.55 | Samot | Stop with that. |
| 14:04.13 | Samot | That is now the second time you have either put words in my mouth and skewed what I have said. |
| 14:04.42 | velix | hm? Sorry, wasn't bad. I didn't take notes, who said what. |
| 14:05.30 | velix | It was "seanbright", sorry! Not you. |
| 14:05.33 | sibiria | guys, don't fight. please hug and make up |
| 14:05.38 | sibiria | on second thought, keep your distances |
| 14:05.40 | velix | Samot: Sorry! |
| 14:05.48 | sibiria | wait until the pandemic has blown over |
| 14:12.26 | Testeree | is it possible to create confbridge using ARI yet? |
| 14:13.32 | file | yes? |
| 14:17.06 | Testeree | what should be the type for creating confbridge with ARI, i can see mixing,holding, etc not confbridge, is it a speacial type? https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Bridges+REST+API#Asterisk16BridgesRESTAPI-create |
| 14:17.35 | file | define "create confbridge"... |
| 14:17.49 | file | you can create a mixing bridge which is a multi-party conference bridge, but it has no ConfBridge functionality or features |
| 14:17.52 | file | those are up to you to implement |
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| 14:29.59 | velix | Samot: I'm _really_ sorry. |
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| 14:54.42 | velix | Interesting. 7 oder 10 students now have a beautiful sound quality. |
| 14:54.53 | velix | 2 students sound like having a wrong samplerate. |
| 14:55.27 | velix | Can this be a problem with the windows soundcard? |
| 14:55.32 | velix | And the SIP Software? |
| 14:56.26 | sibiria | could be. if it were me, i would dump the RTP stream from one of those calls and see what's going on there |
| 14:56.43 | sibiria | that is, see if the stream itself is fine but the audio in it is botched, or if packets are missing in the stream |
| 14:57.41 | sibiria | but if you can hear that the stream is uninterrupted but for example wrong pitch, then likely you're not having a network issue |
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| 15:00.24 | velix | Good idea. |
| 15:00.29 | velix | They're using the same university laptop. |
| 15:00.43 | velix | I bet, they've got a fixed kHz on the mic. |
| 15:00.47 | velix | Since then can understand me. |
| 15:01.48 | velix | Sorry for hammering on this question, but it drives me nuts: Anyone with an idea, why my call is anonymous internally? It's working for AMI and Callfiles, but actually the extension should handle it even via channel originate? https://bpaste.net/raw/IEYA |
| 15:02.25 | Kobaz | velix: not trying to offend, but this is where the 'support vampire' comment came from |
| 15:02.33 | Kobaz | you need to start providing dumps and examples |
| 15:02.50 | Kobaz | like, full examples |
| 15:02.54 | igcewieling | I try to ignore the vampires. |
| 15:02.56 | Kobaz | the snippit of paste is not enough |
| 15:03.09 | velix | Kobaz: Actually, the file isn't much more. |
| 15:03.12 | Kobaz | and there's no context. what are you trying to do, what did you expect.. |
| 15:03.23 | Kobaz | "this doesn't work |
| 15:03.27 | Kobaz | isn't helpful... |
| 15:03.35 | Kobaz | much more explanation is needed |
| 15:04.07 | velix | From what I've read from the docs, "channel originate" seems to act different. Perhaps that's all the magic. |
| 15:04.27 | velix | It has its own timeout (waiting time) etc. |
| 15:04.42 | Kobaz | channel originate and Originate() are operationally equivalent |
| 15:04.50 | Kobaz | manager Originate has some more control |
| 15:05.13 | Kobaz | if you want callerid, use manager Originate |
| 15:05.41 | igcewieling | The thing is, the paste you provided does not have any reference to originate. It also does not show any evidence of annoymous callerid. |
| 15:05.47 | Kobaz | that's been a limiting factor for a long time... although you *can* use Originate()/channel originate with Local channels to accomplish callerid changes |
| 15:07.28 | velix | igcewieling: I've just left out the sip trunk part for outgoing connections, since I was playing with an internal conference. |
| 15:07.37 | velix | So that's pretty much the real setup. |
| 15:08.06 | velix | I mean, it works with Callfiles and via AMI... so I'm fine at the end. |
| 15:08.07 | Kobaz | don't leave things out! |
| 15:09.05 | velix | Kobaz: Why is my emergancy extension important here? |
| 15:09.08 | igcewieling | velix: that might be the case, but part of the reason to pastebin things so we can see the issue. Your reported issue does not show in the logs you pasted. |
| 15:09.27 | Kobaz | velix: i'm saying in general |
| 15:09.32 | velix | The issue is on the phone only. |
| 15:09.40 | velix | From the logs, it's totally correct. |
| 15:09.51 | velix | The phone shoes "anonymous call". |
| 15:09.56 | Kobaz | you are not experienced enough to know all thats important in the debug output |
| 15:10.03 | Kobaz | so include everything |
| 15:10.07 | igcewieling | then turn on sip debug to see the packet being sent to the phohne. |
| 15:10.22 | velix | okay. let me record it |
| 15:10.29 | igcewieling | Personally, I suspect the extra quotes on the CallerID is causing a problem, but I've not seen that bug in years. |
| 15:10.29 | Kobaz | and dont wait for someone to ask for this |
| 15:11.12 | Kobaz | prior to asking your question, it's your job to provide as much debug as possible... if it's a SIP call, provide SIP dump. asterisk logs, sip configuration |
| 15:11.13 | Kobaz | all of it |
| 15:11.23 | igcewieling | You are also incorrectly using the CALLERID() function. |
| 15:11.45 | igcewieling | I see more than just a number: Set("SIP/5020-00000001", "CALLERID(number)="0123456" <0123456>") |
| 15:11.57 | Kobaz | haha yeah that's not right |
| 15:12.06 | Kobaz | You meant to use (all) there |
| 15:12.24 | igcewieling | of course, a sip debug of the invite packet should have shown that. |
| 15:41.27 | velix | Ah, yeah. Anonymous" <sip:anonymous@anonymous.invalid> |
| 15:41.31 | velix | igcewieling: Here it is: https://bpaste.net/raw/CPYA |
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| 15:53.49 | igcewieling | velix: did you correct your callerid error? |
| 15:54.31 | velix | igcewieling: setvar=CID="0123456" or without quotes? |
| 15:55.30 | igcewieling | without the quotes. But did you change your CallerID lines in the dialplan too? |
| 15:56.20 | igcewieling | CALLERID(all) John Doe <1234> or CALLERID(name)=John Doe and CALLERID(num)=1234 |
| 15:57.01 | igcewieling | NEVER CALLERID(num)=John Doe <1234> and NEVER CALLERID(name)=John Doe <1234> |
| 15:58.25 | velix | yes, I changed them. |
| 15:58.32 | igcewieling | Also in your sample Executing [1701@conference:1] Verbose("SIP/5020-00000007", "1, "User "" <> has created or joined a conference."") in new stack |
| 15:59.32 | igcewieling | and STILL wrong -- Executing [1701@conference:3] Set("SIP/5020-00000007", "CALLERID(name)="0123456" <0123456>") in new stack |
| 16:00.18 | igcewieling | And it is CALLERID(num) not CALLERID(number) |
| 16:00.30 | velix | igcewieling: Yeah, I will correct his. |
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| 16:03.38 | Samot | 12:00:18 PM <igcewieling> And it is CALLERID(num) not CALLERID(number) <-- What's the difference? Considering that CALLERID(num) is listed as the datatype in the Wiki and docs. |
| 16:05.21 | igcewieling | core show function CALLERID does not list (number), it only lists (num) |
| 16:06.01 | sibiria | it's num, name and all (primarily) |
| 16:07.10 | Samot | Yeah, I read that wrong. I reversed it. |
| 16:07.14 | Samot | Sorry. |
| 16:11.33 | igcewieling | heh, ordered a fiber line on Oct 10th and it still isn't installed. |
| 16:18.21 | sibiria | "the only guy who can be bothered to fulfil what you paid us for is in the ICU. let's reconnect in 6 months" |
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| 17:19.13 | igcewieling | 6 months from order to install is not all that unusual for us. |
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| 17:21.53 | BiggTimber | I'm trying to setup a SIP trunk from my Asterisk 11 IVR box to my 3CX main pbx. When I configure the sip.conf to register to the 3cx box there are errors in the logs that say "chan_sip.......: operation not permitted" |
| 17:22.17 | BiggTimber | I'm not sure what is failing, as I'm not seeing traffic in my tcpdump on the 3cx box. |
| 17:22.33 | BiggTimber | iptables isn't blocking it on the 3cx box, that I can tell. |
| 17:22.55 | BiggTimber | They are on seperate LANs, but reachable via 'internal' routers. |
| 17:23.00 | BiggTimber | and can ping each other |
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| 17:58.07 | Kobaz | igcewieling: i would love a 6 month turnaround for installs... we've come to be known locally as the 'emergency phone system vendor' |
| 17:58.30 | Kobaz | We spin up fully configured systems in like 2-3 days for 75+ users |
| 17:58.50 | Kobaz | Although we have a lot of tools to automate that |
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| 18:01.08 | igcewieling | Kobaz: Once the circuit is installed, we can do installs in 1 - 2 days. I also have tools to automate that sort of thing. |
| 18:01.22 | igcewieling | PBX installs in 1 - 2 days |
| 18:07.07 | Kobaz | oh, well yeah waiting for the circuit can take a while |
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| 18:25.46 | theGoat | so i want to implement asterisk cdr into a mysql database via res_odbc. all the schema i am finding is saying that the schema is depricated in 1.8, is there a place i can get the latest schema? |
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| 18:43.02 | igcewieling | ./contrib/realtime/mysql/mysql_cdr.sql |
| 18:43.10 | igcewieling | I agree it is a stupid place for it. |
| 18:43.52 | velix | Can I choose an alias extension, f.e. for [student01] to be reachable as SIP/5010, but he needs to login as student01? Or is this a task to be done in the dialplan? |
| 18:53.11 | seanbright | dialplan |
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| 19:04.11 | Kobaz | does zoiper on iphone have 'background' issues. Where if you put the phone into the background does it become unreachable/unavailable? |
| 19:04.23 | Kobaz | curious if anyone'e used this on iphone (I'm on Android) |
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| 19:05.04 | theGoat | igcewieling: cool, thanks much |
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| 19:16.17 | theGoat | is there a way i can build res_odbc manually. i don't want to have to recompile my whole asterisk instance |
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| 19:47.38 | velix | Kobaz: here, iPhone 6S with iOS 13 it works. |
| 19:47.52 | velix | Kobaz: but when WiFi is down in standby, it's gone of course ;) |
| 19:48.32 | velix | seanbright: Oh, that's sad. I thought one could seperate the auth. |
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| 20:04.07 | seanbright | velix: i don't understand your question |
| 20:04.26 | seanbright | velix: the username defined in sip.conf _can_ be the same as the extension, but it doesn't have to be |
| 20:06.11 | Alblasco1702 | Kobaz, on iphone and newer versions of android a softphone goes byebye after some time a app canĀ“t run long times, as it will be killed by rhe os. you need push message to wakeup the phone, |
| 20:06.38 | Alblasco1702 | i didn't get it to work with zoiper or linphone. |
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| 20:12.59 | velix | seanbright: I've tried to set [1234] username=blah and [blah] username=1234, but it seems like username is ignored. |
| 20:13.26 | velix | wasn't there also auth_username? let me look it up |
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| 22:13.50 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.32.0 (2020/03/12) 16.9.0 (2020/03/12) Standard: 17.3.0 (2020/03/12); DAHDI: 3.0.0 (2018/11/15); libpri 1.6.0 (2017/01/27) -=- Wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 |
| 22:46.48 | *** join/#asterisk pchero_work (~pchero@2a02:a210:2241:6480:930:c5c8:8d76:ddf1) |
| 23:02.23 | sawgood | hey: what is the best way (to determine) which channel driver (PJSIP vs chan_SIP) is currently bind to 5060? |
| 23:24.42 | *** join/#asterisk m4rcu5 (nobody@84-106-248-133.cable.dynamic.v4.ziggo.nl) |
| 23:41.02 | *** join/#asterisk pchero_work (~pchero@2a02:a210:2241:6480:930:c5c8:8d76:ddf1) |