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| 02:53.01 | SaxonManangan | hi |
| 02:54.03 | SaxonManangan | i hav a 2 gateways and need to be load balanced |
| 02:54.28 | SaxonManangan | is this right? |
| 02:54.29 | SaxonManangan | exten => _X.,1,Set(CALLERID(all)="321" <321>) |
| 02:54.29 | SaxonManangan | exten => _X.,n,GotoIf($[${RoundCount}>=3]?skip2) |
| 02:54.29 | SaxonManangan | exten => _X.,n,GotoIf($[${RoundCount}<=2]?skip1) |
| 02:54.29 | SaxonManangan | exten => _X.,n,SetGlobalVar(RoundCount=0) |
| 02:54.29 | SaxonManangan | exten => _X.,n,Wait(1) |
| 02:54.30 | SaxonManangan | exten => _X.,n(skip2),SetGlobalVar(RoundCount=0) |
| 02:54.30 | SaxonManangan | exten => _X.,n(skip1),SetGlobalVar(RoundCount=$[{${RoundCount}+1}]) |
| 02:54.31 | SaxonManangan | exten => _X.,n,Dial(SIP/${EXTEN}@GW${RoundCount},45,r) |
| 02:54.31 | SaxonManangan | exten => _X.,n,Hangup |
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| 05:16.42 | SaxonManangan | hi |
| 05:16.57 | SaxonManangan | need help |
| 05:17.20 | SaxonManangan | 2 gateways to be load balanced |
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| 06:59.45 | avb | SaxonManangan: it could work. why not try it? :) |
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| 11:46.33 | file | falls over |
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| 11:56.56 | davlefou | hi, is it possible to have statistic about quality of call? |
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| 12:59.41 | Kobaz | whatwhatwhat |
| 12:59.49 | Kobaz | polycom is getting bought by mitel |
| 13:02.57 | file | yes. |
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| 13:08.12 | Kobaz | nut diddely uddlered |
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| 13:31.44 | avb | Kobaz: wow :) fresh news |
| 13:32.05 | avb | Kobaz: are they going to continue 'polycom way' or 'mitel way'? |
| 13:32.25 | avb | if its a mitel way, thats the worst day in voip industry |
| 13:34.53 | utrack | Hey! I'm using Originate() to create calls in the app now - is there any way to notify the first called extension when the second one had picked up the phone? |
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| 13:41.42 | IPalmer | hi all, I'm having a bit of trouble with curl realtime. Not sure if this is the right place but here goes. I've written an app in c# which services curl realtime, the trouble is it seems to block Asterisk, when I watch the console it can take up to a minute for a response to a pjsip show endpoints. The c# code returns almost instantly so it doesn;t seem to be here. When i stop my app the asterisk cli then sends through loads of requests at once, so i |
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| 13:44.20 | wdoekes | IPalmer: your msg was truncated at "requests at once, so i ..." |
| 13:46.17 | IPalmer | wdoekes: Apologies I'd nearly finished. Was just saying that when I stop my app, the automated requests Asterisk has made all seem to run at once, almost like the initial request had blocked them |
| 13:52.58 | wdoekes | sounds like a clue |
| 13:53.48 | wdoekes | my first guess would be that your app doesn't properly indicate the end of the response |
| 13:54.07 | wdoekes | (through content-length or chunked-len stuff) |
| 13:54.40 | wdoekes | or that it gets a Connection: close from Asterisk, but doesn't honor it |
| 13:55.14 | IPalmer | wdoekes: yep I think you are correct, I killed the httplistener after every request and it returns instantaneously. Looks like it's my bad code :( |
| 13:56.00 | wdoekes | creating a real world http server is not trivial |
| 13:56.28 | wdoekes | I suggest you use a well tested lib, or -- even better -- a real http daemon (nginx) in front of your app |
| 13:56.39 | Kobaz | avb: that's a good question |
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| 14:35.15 | ice9 | I defined peer in iax.conf and created the required extension, now when I try to dial a number from jitsi, I just get "service unavailable", I'm not seeing anything related in the log file |
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| 14:48.10 | [TK]D-Fender | Confirm where you're connecting from, your networking betweem the two, and show us you have iax2 debug enabled at CLI |
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| 14:51.56 | ice9 | how can I monitor/debug the dialed numbers in real time? |
| 14:52.05 | [TK]D-Fender | iax2 set debug on |
| 14:52.18 | [TK]D-Fender | and my Jitsi doesnt' show IAX2 support at all... |
| 14:53.11 | ice9 | which Linux clients supports TLS client cert? |
| 14:59.42 | [TK]D-Fender | https://www.google.ca/#q=jitsi+tls+configuration |
| 14:59.49 | [TK]D-Fender | Sure seems to say you can with jitsi |
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| 16:29.41 | ice9 | I enabled encryption in the sip.conf for the users but when I try to dial another user I get this " Matched device setup to use SRTP, but request was not" |
| 16:31.59 | [TK]D-Fender | then the other side does not agree |
| 16:35.50 | ice9 | clients are jitsi and csipsimple, both supports SRTP I think |
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| 16:37.38 | [TK]D-Fender | Saying they support doesn't mean they're configured right to use it |
| 16:37.52 | ice9 | I see |
| 16:37.55 | ice9 | I will check that |
| 16:38.07 | [TK]D-Fender | One side says it expects it, the other says they aren't interested. You need to fix the mismatch |
| 16:39.24 | pzn | Hi! I'm asterisk newbie. How can I make asterisk to log into another pbx as a sip client (so be used for incoming calls)? can someone point me to some documentation about that? what is the name of this feature that I need to configure, thanks in advance |
| 16:40.00 | ice9 | so asterisk supports zrtp or srtp? |
| 16:40.01 | [TK]D-Fender | There is no "feature". |
| 16:40.10 | [TK]D-Fender | SIP is SIP. |
| 16:40.20 | [TK]D-Fender | Set up a peer like you would any other, like for a phone or ITSP, etc |
| 16:40.44 | [TK]D-Fender | ice9, Yes. Whether you configured it to allow it is another matter |
| 16:41.24 | ice9 | [TK]D-Fender: I mean when I add "encryption=yes" which encryption it use? |
| 16:41.40 | [TK]D-Fender | Should use SRTP. |
| 16:41.52 | [TK]D-Fender | And tthere are more settings like the keys to use, etc |
| 16:41.56 | [TK]D-Fender | You can't jsut say "yes" |
| 16:44.02 | pzn | [TK]D-Fender, sorry, I have problems with english terms, "feature" wrongly used. my common usage is to create an entry in sip.conf for sip-phones connect to asterisk. now I need asterisk to be as a sip-client / like a phone. which is the name of this, for me to search asterisk documentation? |
| 16:47.39 | [TK]D-Fender | its the same thing |
| 16:47.43 | [TK]D-Fender | there is no difference |
| 16:47.59 | [TK]D-Fender | make a peer. Set the host. set the user & secret |
| 16:48.01 | [TK]D-Fender | set your codecws |
| 16:48.23 | [TK]D-Fender | the comm to a phone is no different |
| 16:48.50 | ice9 | [TK]D-Fender: I'm following that tutorial https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial can you point me to a correct one |
| 16:49.09 | [TK]D-Fender | ice9, Show us what you've actually done along with the actual failure. |
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| 16:56.11 | pzn | [TK]D-Fender, ok, I think I got the point, tks! |
| 17:07.25 | *** join/#asterisk eitan (~eitan@cpe-172-91-119-47.socal.res.rr.com) |
| 17:08.14 | eitan | hey guys! I am trying to create an internal office video conferencing. We grew to a bigger office, and people are up stairs and stuff. I want them to easily with 1 button click on a device and be able to buzz and talk to someone elsewhere.... I was thinking asterisk? am I crazy? |
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| 17:24.34 | MaliutaLap | In general? or because of that idea? |
| 17:24.46 | MaliutaLap | all * people are a bit crazy :) |
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| 18:43.21 | ski7777 | hi |
| 18:44.04 | ski7777 | can I use an old Smartphone via US on a Raspberry pi to use it as an Gateway to the public phone net? |
| 18:48.42 | [TK]D-Fender | If your smartphone can talk to * clearly that end will do what it does |
| 18:48.54 | [TK]D-Fender | How you conenct to "public phone net" is another matter" |
| 18:49.05 | [TK]D-Fender | There are certainly ways & options |
| 18:54.04 | MaliutaLap | some string and a couple of tin cans for example |
| 18:56.54 | ski7777 | can you give me further information or a link |
| 18:57.39 | [TK]D-Fender | Dep[ends HOW you want to connect to it |
| 18:58.11 | [TK]D-Fender | Using an ITSP... that will depend on the ITSP (and they will likely have samples),if you intend to use a physical line, then that'll depend on the interface you choose for thatr |
| 18:58.35 | Samot | So I have something strange happening. I have a couple PBX systems that use AMI to originate calls. Generally these calls are one to one but I'm testing using this to originate and then send people into a conference room. That's working great but for some reason about 3 minutes or so into the call, all the calls go out again. |
| 18:59.52 | [TK]D-Fender | Either you're resending them, or you hit a timeout and the call was never answered |
| 19:00.20 | ski7777 | I will only connec a USB-Cable and finish |
| 19:00.52 | Samot | Well I know I'm not re-sending them because the testing is manual and my call is answered because I'm in the conference room but it recalls me. |
| 19:01.00 | [TK]D-Fender | "and finish"? |
| 19:01.13 | Samot | Money shot? |
| 19:01.30 | [TK]D-Fender | Samot, We'll need to see debug..... |
| 19:02.08 | MaliutaLap | Samot: I wouldn't call that setup "money" anything :) |
| 19:02.17 | Samot | Give me a sec. I'll get the call file setup and move it into /var/spool/asterisk/outgoing... |
| 19:02.39 | MaliutaLap | Samot: it'll probably leave him as unsatisfied as his partner ;) |
| 19:12.00 | Samot | It's always fun being in a conference room with yourself twice. |
| 19:16.11 | MaliutaLap | echoes and feedback ... my kind of "fun" ;) |
| 19:17.59 | Samot | Doh! I see the issue. |
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| 19:19.46 | Samot | MaxRetries will happen even if the call is connected? |
| 19:21.18 | Samot | I mean it's set to 0 so it shouldn't retry on a connected call. |
| 19:21.31 | Samot | But after 300 seconds it retires the call. |
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| 19:22.21 | Samot | Shouldn't a retry only be attempted if the call failed? |
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| 19:25.26 | MaliutaLap | that was my impression ... I've never used that setting though |
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| 19:25.42 | MaliutaLap | and I generally use call files rather an AMI |
| 19:25.58 | Samot | It's the same thing. |
| 19:26.18 | Samot | Right now I'm not using AMI to connect the calls for testing. |
| 19:26.42 | Samot | I have callfiles that I use for testing and I just move them to /var/spool/asterisk/outgoing |
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| 19:28.05 | Samot | MaxRetries=0 (default setting) and I didn't specify RetryTime because it shouldn't be needed but that has a default setting of 300 seconds. |
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| 19:30.21 | MaliutaLap | Tried setting retrytime to 0 (or to something really big, like a week) |
| 19:30.24 | MaliutaLap | ? |
| 19:32.58 | Samot | Not yet but I don't want that either. |
| 19:33.11 | Samot | I originated a call to two of my DIDs. |
| 19:33.24 | Samot | Both were answered and after 300 seconds it retried them both. |
| 19:33.34 | Samot | While I was still on the calls. |
| 19:33.50 | Samot | I don't want a "ghost" call to happen 7 days later on a retry. |
| 19:36.25 | MaliutaLap | odd. According to documentation MaxRetries=0 means only one attempt |
| 19:36.34 | Samot | Yup. |
| 19:36.54 | Samot | And retry only if it fails. |
| 19:37.36 | MaliutaLap | by that definition there should be no retry, but if maxretries > 0 then, yes it should only retry on a failure |
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| 19:58.21 | twanny796 | what should I use for an ISDN card, mISDN or Dahdi? |
| 19:59.07 | Samot | MaliutaLap: I agree. |
| 20:01.02 | rmudgett | twanny796: Use DAHDI with libpri. The version of mISDN used with Asterisk does not compile on newer OS versions. |
| 20:03.41 | lorsungcu | Samot: show the call file and the CLI capture. |
| 20:04.22 | Samot | Give me a moment on that |
| 20:05.03 | Samot | - Executing [3213003577@from-trunk:1] [1;36mNoOp[0m("[1;35mSIP/MegaPath2-000001ad[0m", "[1;35mCatch-All DID Match - Found 3213003577 - You probably want a DID for this.[0m") in new stack |
| 20:05.03 | twanny796 | rmudgett: ok |
| 20:05.07 | Samot | Wrong channel |
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| 20:08.14 | Samot | https://www.irccloud.com/pastebin/bg1DI3PX/ |
| 20:08.18 | Samot | There's the call file. |
| 20:09.51 | Samot | I'll need to redo the trace. The scroll buffer wasn't set big enough. |
| 20:10.06 | Samot | So that's going to be about another 5 minutes or so. |
| 20:11.47 | Samot | Let's see if it will do it with just the one call. |
| 20:18.07 | MaliutaLap | Samot: that file looks fine to me |
| 20:18.25 | MaliutaLap | just checked a couple of things ... it shouldn't be getting the retries you're seeing |
| 20:19.21 | Samot | http://pastebin.com/Qt15tNHM |
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| 20:25.21 | MaliutaLap | Samot: is that a loop? |
| 20:25.38 | Samot | What do you mean a loop? |
| 20:25.52 | Samot | that was the original call followed by it recalling while I was still on the original call. |
| 20:26.35 | MaliutaLap | Samot: ok, so not the file being picked up again? |
| 20:26.50 | Samot | As far as I can tell. |
| 20:26.52 | Samot | One sec. |
| 20:27.18 | MaliutaLap | put the call file in place - let it get picked up by the spool, the delete it |
| 20:27.38 | Samot | It deletes on it's own. |
| 20:27.46 | Samot | It deletes on its own. |
| 20:27.48 | MaliutaLap | it is being deleted? |
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| 20:28.46 | MaliutaLap | because the "[2016-04-15 20:15:55] WARNING[20915]: pbx_spool.c:318 safe_append: Unable to set utime on /var/spool/asterisk/outgoing/test: Operation not permitted" message might indicate that * doesn't have the correct permission to remove the file |
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| 20:29.49 | cousin_luigi | Greetings. |
| 20:29.56 | Samot | yup, I just caught that. |
| 20:29.59 | Samot | Let me try again |
| 20:30.18 | MaliutaLap | Samot: might have been a simple permissions issue :) |
| 20:30.24 | cousin_luigi | Can you recommend a commercial embedded device for residential use that runs asterisk? |
| 20:30.30 | MaliutaLap | Samot: wouldn't that suck? :) |
| 20:30.32 | Samot | Yup. Stupid script. |
| 20:30.36 | MaliutaLap | cousin_luigi: RPI |
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| 20:31.20 | lorsungcu | wtf is an embedded device. |
| 20:31.21 | MaliutaLap | Samot: well, it's the same with anything daemony under *nix ... if the daemon can't remove the spool file it will keep picking it up :) |
| 20:31.23 | cousin_luigi | MaliutaLap: I forgot to mention I also need an analog port. |
| 20:31.40 | MaliutaLap | cousin_luigi: RPI+ATA |
| 20:31.50 | Samot | Well I'm testing that now |
| 20:32.02 | Samot | But the callfile doesn't delete until the call is done I believe. |
| 20:32.07 | cousin_luigi | MaliutaLap: I'm aware of that option. I'd very much like to have a single device. |
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| 20:33.02 | MaliutaLap | cousin_luigi: good luck with that |
| 20:36.03 | Samot | Well sombitch |
| 20:36.53 | MaliutaLap | Samot: let me guess ... I was right? |
| 20:37.09 | MaliutaLap | is shocked! Shocked I tells you! ;) |
| 20:37.20 | Samot | Blah, yeah. The script wasn't executing the ownership changes. |
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| 20:37.38 | Samot | I wrote a script to move the call files because there was like 20 during the testing. |
| 20:37.56 | MaliutaLap | gotta love those hours of SIP debugging for a file permission error :) |
| 20:38.37 | afallison | Hey everyone I'm still banging my head on a Header issue and could use some help. I posted on stackoverflow hoping to hear from some Twilio devs, but please chime in on here or on the post if you can help thanks. http://bit.ly/1VqCvym |
| 20:40.03 | MaliutaLap | afallison: Freepbx?? go ask in #freepbx - they have their own channel |
| 20:40.11 | MaliutaLap | and they do nasty things to * |
| 20:41.02 | MaliutaLap | Samot: I have a script around here that does something similar ... used for prank calling work mates with tt-monkeys |
| 20:41.39 | MaliutaLap | Samot: I _think_ I used cp with a file that was 0666 |
| 20:41.56 | Samot | It was ownership issues not permissions issues. |
| 20:42.02 | Samot | It was owned by root not asterisk. |
| 20:42.12 | Samot | It needs to be owned by asterisk for what I'm doing. |
| 20:42.20 | MaliutaLap | ok |
| 20:42.34 | MaliutaLap | normally if * has write perms it's sufficient |
| 20:42.35 | cousin_luigi | bbl |
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| 20:47.09 | Samot | Well this is a FreePBX system, so it all needs to be owned by Asterisk. |
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