| 00:09.39 | *** join/#asterisk cryptic_ (~cryptic@67-8-35-31.res.bhn.net) |
| 00:17.12 | Samot | Then you shouldn't use Kamailio. |
| 00:18.20 | drmessano | 3CX runs on Linux now |
| 00:18.26 | drmessano | Might be a good time to switch |
| 00:18.27 | Samot | No crap? |
| 00:18.51 | Samot | I wonder if that brings the license cost down. |
| 00:19.06 | drmessano | IncrediblePBX 5 is 3CX based, apparently |
| 00:20.47 | Samot | Hrm. |
| 00:21.01 | Samot | "Free" is 8 sim calls, 5 web conferences, 1 sip trunk. |
| 00:21.06 | drmessano | Yeah |
| 00:21.09 | Samot | Awesome |
| 00:21.12 | Samot | Woooo. |
| 00:21.25 | Samot | For low it's good. |
| 00:21.39 | Samot | As long as you have one provider. |
| 00:22.08 | drmessano | I assume they are still using some proprietary calling engine |
| 00:22.58 | Samot | Hrm. I wonder if my partner account is still good... |
| 00:23.31 | *** join/#asterisk [NC] (~nc@rv1.sabius.net) |
| 00:26.37 | Samot | Was looking at 3CX a few months ago for the hotels. |
| 00:26.58 | Samot | Since they have the PMS integration with most of the PMS system they use. |
| 00:31.17 | drmessano | I wonder how it runs on a NUC or Pi |
| 00:32.48 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 00:36.14 | *** join/#asterisk shootbird (~quassel@beepbeep.serverpit.com) |
| 00:37.54 | drmessano | Some forum posts states it runs the "same technology" |
| 00:42.11 | Akari | They offer an ARM build? |
| 00:43.42 | Akari | I think I've heard of their softphone app |
| 00:46.40 | drmessano | They do not |
| 00:47.57 | drmessano | Apparently its a Debian ISO for amd64 |
| 00:48.06 | drmessano | Netinstall even |
| 00:50.14 | WIMPy | Nice |
| 00:52.19 | drmessano | Hell freezing over and all that |
| 00:52.29 | drmessano | Maybe Counterpath will update BRIA for Linux |
| 00:53.16 | Akari | What do you guys like to use for softphones? |
| 00:55.37 | drmessano | Which OS? |
| 00:56.28 | Akari | Windows/macOS/iPhone/Android |
| 00:56.39 | Akari | Well really any of them |
| 00:56.48 | Akari | I guess some people use linux on a desktop. |
| 01:02.28 | drmessano | I run BRIA on iOS.. Blink on Windows and Linux |
| 01:03.11 | drmessano | Some people do run Linux on a desktop. Lots of us |
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| 01:31.12 | *** join/#asterisk Tamo (~t@5ec0f3cb.skybroadband.com) |
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| 02:08.16 | voipmonk | Acrobitz Softphone |
| 02:08.16 | voipmonk | Bria , Zoiper, and Jitsi |
| 02:08.41 | voipmonk | sometimes Csimple |
| 02:14.12 | snadge | im just going to do some simple user agent filtering.. since most of the time its our friendly sip scanner |
| 02:14.18 | *** join/#asterisk rpifan (~rpi@73.106.75.8) |
| 02:14.20 | snadge | until i can think of something more robust :P |
| 02:23.30 | *** join/#asterisk celer (~celer@pc1.robertwaltersjp-unet.ocn.ne.jp) |
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| 02:41.50 | Samot | Kamailio would be more robust, however, you said you were looking for something just to turn up and make small tweaks. That's not possible with Kamailio. You need to write the configuration, load the modules you want to use, configure the settings for the modules, etc. |
| 02:48.25 | nate_b | Does anyone know where I might be able to discover what variables and functions are available in the HINT priority? |
| 02:48.39 | voipmonk | um... |
| 02:48.44 | nate_b | EXTEN seems to be there, but CHANNEL does not.... |
| 02:49.19 | *** join/#asterisk cryptic (~cryptic@67-8-35-31.res.bhn.net) |
| 02:49.26 | nate_b | Trying to take advantage of the dynamic hints (or even RealTime if at all possible...) |
| 03:11.14 | kunwon1 | realtime hints is doable- :exten => _1XX,hint,SIP/${ODBC_GETHINT(${EXTEN})} |
| 03:11.19 | kunwon1 | minus colon |
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| 03:39.30 | nate_b | @kunwon1: Thanks!! I'll dig into that! |
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| 10:01.11 | *** join/#asterisk chl_ (~chl@unaffiliated/chl/x-9330839) |
| 10:01.49 | chl_ | hello, got a question regarding cdr-csv in asterisk, is it possible to rotate the file daily without it having any effect on asterisk? |
| 10:02.03 | chl_ | rotating Master.csv daily* |
| 10:06.29 | *** join/#asterisk matrix1233 (~matrix123@41.230.26.59) |
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| 10:59.17 | bittis | just log to a db |
| 10:59.20 | bittis | better |
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| 11:26.40 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 12:03.38 | *** join/#asterisk MajesticFudgie (~MajesticF@178-32-65-132.eur1.stack.ilkotech.co.uk) |
| 12:04.58 | MajesticFudgie | Any outgoing calls I make through sipgate when using the originate command in CLI or via outgoing spool show as private numbers, how can I resolve this? |
| 12:05.11 | MajesticFudgie | If I set callerid to random crap in the .call file the correct number shows |
| 12:17.13 | *** join/#asterisk M2K_ (uid194711@gateway/web/irccloud.com/x-kfowtrnzdjnvxazr) |
| 12:17.27 | M2K_ | hello everyone |
| 12:22.32 | M2K_ | having some interest on the askterisk to fulfil my communications need. but ive ran into a bit of a wall at the moment, think anyone could help me? id be glad to giving a description of the system and possably a diagram of the callflow :D |
| 12:22.52 | M2K_ | its more of an architectural setup question |
| 12:25.59 | *** join/#asterisk [TK]D-Fender (~joe@216-191-106-165.dedicated.allstream.net) |
| 12:26.23 | [TK]D-Fender | M2K_, just ask |
| 12:28.31 | MajesticFudgie | Hi [TK]D-Fender |
| 12:30.17 | M2K_ | so im wanting to change the cli of a call as it is going out, however i want people to call in or "request a call" thus there will be two calls generated by the pbx one calling the client the other calling the end point, |
| 12:30.53 | [TK]D-Fender | clarify that |
| 12:30.54 | M2K_ | i believe that the calls can be set with call files, however how do i make these two calls merge into a conference like link |
| 12:31.39 | [TK]D-Fender | How does 1 in lead tto 2 out? |
| 12:31.58 | M2K_ | pbx calls 0123456 and then calls 03548954 how do i set these to merge with o user input |
| 12:32.54 | M2K_ | placing the calls can be achieved by call files, to ring both parties, however how do i go by merging these two seperate calls into something such a s a conference call or any other method. |
| 12:33.11 | [TK]D-Fender | dump them into a conference |
| 12:33.53 | M2K_ | thus they are two out, however thorugh the placing of a call fine, am i able to place them in a conferenc room ? |
| 12:34.15 | M2K_ | using the pbx as a middle man handling both calls |
| 12:34.15 | [TK]D-Fender | yes |
| 12:34.35 | M2K_ | nice :p |
| 12:34.40 | [TK]D-Fender | Call file gets dumped into the dialplan wherever you tell it to |
| 12:34.50 | M2K_ | know of any material that would help on this? |
| 12:35.05 | *** join/#asterisk miralin (~Thunderbi@195.19.212.23) |
| 12:35.47 | M2K_ | ive looked at the dialing using the call files however im unsure of how to merge these two outgoing calls |
| 12:37.00 | [TK]D-Fender | <[TK]D-Fender> dump them into a conference <- |
| 12:37.27 | [TK]D-Fender | Or use Bridge() |
| 12:37.56 | M2K_ | okay thank you sdo much d-fender |
| 12:38.12 | M2K_ | i shall pop back if i need further help by the way ^.^ |
| 12:45.55 | MajesticFudgie | [TK]D-Fender, Any idea how to set the outgoing ID for sipgate in sip.conf |
| 12:46.26 | [TK]D-Fender | huh? |
| 12:46.38 | [TK]D-Fender | What "ID"? |
| 12:47.04 | MajesticFudgie | callerid |
| 12:47.13 | MajesticFudgie | Outgoing calls show as anonymous |
| 12:48.01 | [TK]D-Fender | so go set the callerid |
| 12:48.19 | MajesticFudgie | I tried, but it makes no difference |
| 12:48.24 | [TK]D-Fender | Show us |
| 12:48.28 | MajesticFudgie | sec |
| 12:48.35 | [TK]D-Fender | the full call, beginning to end |
| 12:50.31 | *** join/#asterisk jjrh (~jjrh@ppp-199-246-169-1.storm.ca) |
| 12:51.03 | MajesticFudgie | okie give me a tick gotta remember how to do that |
| 12:57.45 | *** join/#asterisk clopez (~tau@neutrino.es) |
| 12:58.09 | MajesticFudgie | [TK]D-Fender I don't remember how to. There was an application that does this but I dont remember the name |
| 12:58.24 | [TK]D-Fender | huh? |
| 12:58.26 | [TK]D-Fender | Application? |
| 12:58.30 | [TK]D-Fender | I said to show the CALL |
| 12:58.31 | [TK]D-Fender | * CLI |
| 12:58.40 | [TK]D-Fender | There is no "Application" |
| 12:59.53 | *** join/#asterisk friedrich (~friedrich@aextron.de) |
| 13:01.23 | MajesticFudgie | Well the CLI shows this when using originate command http://pastebin.com/wicnTwrM |
| 13:01.48 | MajesticFudgie | Someone installed an interactive program on my server to view the raw call |
| 13:02.19 | MajesticFudgie | voipmonk you there? |
| 13:03.53 | [TK]D-Fender | no |
| 13:03.59 | [TK]D-Fender | You look with * |
| 13:04.05 | [TK]D-Fender | You do not need external garbage |
| 13:04.11 | [TK]D-Fender | You* is generating the call |
| 13:04.42 | [TK]D-Fender | <PROTECTED> |
| 13:04.50 | [TK]D-Fender | that means the calling leg has NO CALLERID |
| 13:05.04 | [TK]D-Fender | Three IS no identity for the calling end |
| 13:05.08 | [TK]D-Fender | there* |
| 13:05.31 | [TK]D-Fender | because it's not like the call sttarted from another channel issuing a Dial() that had a CID to begin with |
| 13:05.33 | MajesticFudgie | Yes, But I were under the understanding I could set this in the sip.conf so any calls automatically assume the CID |
| 13:05.43 | [TK]D-Fender | So go show what you set |
| 13:06.09 | [TK]D-Fender | And "show the call" means EVERYTHING. ---> SIP DEBUG |
| 13:06.13 | [TK]D-Fender | "sip set debug on" |
| 13:06.26 | MajesticFudgie | Thankyou for the comand, I said I had no idea how to get it to do that |
| 13:06.28 | [TK]D-Fender | show a new call and the config to match masking only the secret |
| 13:12.02 | MajesticFudgie | Call Log: http://pastebin.com/f4ycznPm |
| 13:13.41 | [TK]D-Fender | From: "Anonymous" <sip:2399808e0@sipgate.co.uk>;tag=as4c1c322d |
| 13:13.53 | [TK]D-Fender | You know seeing LETTERS in there doesn't install a lot of faith |
| 13:14.01 | MajesticFudgie | Letters? |
| 13:14.26 | [TK]D-Fender | ----------> 2399808e0 <-------------------- |
| 13:14.26 | MajesticFudgie | http://paste.ubuntu.com/23416036/ thats my sip.conf |
| 13:15.04 | [TK]D-Fender | fromuser=2399808e0 <- Overrides callerid is that's what gets used for presentation |
| 13:15.05 | MajesticFudgie | I might add, I didnt setup sipgate, voipmonk helped me out as idk how to do it. |
| 13:15.09 | MajesticFudgie | Ah |
| 13:15.28 | MajesticFudgie | Can fromuser be a convetional callerid format? "Blah" <123> ? |
| 13:15.37 | [TK]D-Fender | Set : "sendrpid=yes", "trustrpid=yes" |
| 13:15.39 | [TK]D-Fender | ^ |
| 13:15.44 | [TK]D-Fender | Do both, place a new call |
| 13:16.33 | [TK]D-Fender | <MajesticFudgie> Can fromuser be a convetional callerid format? "Blah" <123> ? <- no. it literally takes that and sets it as the user portion of the From:. It is sometimes required to be the exact username which would not be the CID |
| 13:16.46 | MajesticFudgie | Ah |
| 13:16.47 | [TK]D-Fender | Apply the changes I suggested and try again |
| 13:16.54 | MajesticFudgie | and I did both and it shows anonymous still |
| 13:17.03 | MajesticFudgie | The call shows no different in console |
| 13:17.44 | MajesticFudgie | I put them in [sipgate], werent too sure if you wanted me to put them in [general] or wether it doesnt matter |
| 13:19.21 | MajesticFudgie | btw if I put a .call file in /var/spool/asterisk/outgoing and set the callerid to some random crap it will show the correct number |
| 13:19.51 | Samot | Are you setting Caller ID at all? |
| 13:21.17 | MajesticFudgie | No, although I were under the impression If I dont sipgate will use my sip number they assigned to me |
| 13:21.42 | MajesticFudgie | I've tried to set the caller id in sip.conf which has no effect on outgoing and automated calls only assume the correct callerid if I set it to something at random |
| 13:21.52 | MajesticFudgie | I'm entirely confused and have no idea how to do this |
| 13:22.41 | MajesticFudgie | I'm currently testing with the playback application rather than pushing it through a context |
| 13:22.43 | Samot | Well you could set the callerid= in the device peer |
| 13:22.48 | MajesticFudgie | I did |
| 13:22.55 | Samot | Show the device peer. |
| 13:22.59 | Samot | Mask the secret. |
| 13:23.02 | MajesticFudgie | http://paste.ubuntu.com/23416036/ |
| 13:23.16 | MajesticFudgie | I added sendrpid and trustrpid as suggested above |
| 13:23.28 | MajesticFudgie | under [sipgate], thats a paste from before I added them |
| 13:24.48 | Samot | Is this also a paste before you updated fromuser=? |
| 13:25.04 | Samot | Because that's still the same from when TK said to change it. |
| 13:27.25 | MajesticFudgie | idk what to change it to |
| 13:31.16 | Samot | The number you want to use? |
| 13:31.20 | Samot | Or comment it out. |
| 13:33.50 | *** join/#asterisk Ahmadi (bcd26b24@gateway/web/freenode/ip.188.210.107.36) |
| 13:33.54 | Ahmadi | Hi |
| 13:33.58 | Ahmadi | Good day to all |
| 13:35.05 | Ahmadi | I want playback a "hello-world" |
| 13:35.11 | Ahmadi | but not in extensions.conf |
| 13:35.42 | Ahmadi | i want play it at now that i'm in asterisk CLI |
| 13:35.44 | Ahmadi | is it possible? |
| 13:35.45 | [TK]D-Fender | MajesticFudgie, I don't see those lines in the new config |
| 13:36.14 | [TK]D-Fender | Ahmadi, You can't just playback things from nowhere |
| 13:36.18 | [TK]D-Fender | Ahmadi, You need a call channel; |
| 13:36.29 | [TK]D-Fender | Ahmadi, As well as dialplan to process. Playback = dialplan app |
| 13:36.41 | [TK]D-Fender | Ahmadi, What is your final goal there? |
| 13:36.46 | MajesticFudgie | Samot if I put the number as fromuser or comment it out I just get not authorized |
| 13:37.20 | Samot | Then you should talk to you provider as to how they expect Caller ID to be passed. |
| 13:37.29 | [TK]D-Fender | MajesticFudgie, Where is the updated config with only the changes I told you to do? |
| 13:37.35 | MajesticFudgie | Ill paste it, sec |
| 13:37.42 | [TK]D-Fender | undo everything else |
| 13:37.49 | Ahmadi | [TK]D-Fender, I have a room with 2 parties, I want playback a disconnect alram for them, because their credit is near to end. |
| 13:38.36 | *** join/#asterisk cresl1n (Adium@asterisk/libpri-and-libss7-expert/Cresl1n) |
| 13:38.37 | *** mode/#asterisk [+o cresl1n] by ChanServ |
| 13:38.52 | [TK]D-Fender | Ahmadi, clarify what this "room" is, when & how you intend for that playback to occur |
| 13:39.13 | MajesticFudgie | http://paste.ubuntu.com/23416143/ is latest |
| 13:39.53 | [TK]D-Fender | And the call I asked for 25 minutes ago |
| 13:40.18 | MajesticFudgie | When? |
| 13:41.04 | [TK]D-Fender | <[TK]D-Fender> Apply the changes I suggested and try again |
| 13:41.06 | [TK]D-Fender | ^^^^^^^^^^^^^^^^ |
| 13:41.12 | MajesticFudgie | You never asked for the call |
| 13:41.14 | [TK]D-Fender | ^^^^^^^^^^^^^^^^ |
| 13:41.38 | [TK]D-Fender | TRY AGAIN <- PLACE THE DAMN CALL AND OF COURSE I EXPECT TO SEE IT. |
| 13:41.54 | MajesticFudgie | Sorry, I weren't aware I should read your damn mind |
| 13:42.30 | MajesticFudgie | http://pastebin.com/iD7K3zcP Latest call |
| 13:42.39 | [TK]D-Fender | And you only just did the change I told you to do |
| 13:43.18 | [TK]D-Fender | I don't see RPID in that debug.... |
| 13:43.40 | [TK]D-Fender | then again I think that was supposed to come from the originating channel which has none because you['re using CLI originate |
| 13:43.46 | [TK]D-Fender | Why are you using this method? |
| 13:43.56 | [TK]D-Fender | You have no control because of it |
| 13:44.00 | MajesticFudgie | For testing it quickly |
| 13:44.08 | MajesticFudgie | But mainly I use a .call file |
| 13:44.11 | [TK]D-Fender | That is specifically what screws you |
| 13:44.14 | [TK]D-Fender | your test method is bad |
| 13:44.24 | [TK]D-Fender | Call files let you set callerid <--- |
| 13:44.36 | MajesticFudgie | I know, however I didnt want to keep setting it in the call file |
| 13:44.53 | MajesticFudgie | I were wanting to find a way to set it automatically without specifying it in the call file |
| 13:45.09 | [TK]D-Fender | Who cares if it's going to be the same anyway? |
| 13:45.22 | [TK]D-Fender | That was what would let you set it |
| 13:45.24 | [TK]D-Fender | So use it |
| 13:45.45 | MajesticFudgie | Fine |
| 13:46.25 | [TK]D-Fender | Reliably Transmitting (NAT) to 217.10.79.23:5060: <- sipgate is not behind NAT. Your peer is wrong. |
| 13:46.55 | MajesticFudgie | What would setting nat=no actually change btw |
| 13:47.02 | [TK]D-Fender | Contact: <sip:2399808e0@192.168.2.204:5060> <- and you're passing them your PRIVATE IP for the contact. Your NAT settings under [general] as also wrong/incomplete |
| 13:47.29 | [TK]D-Fender | MajesticFudgie> What would setting nat=no actually change btw <- if they propse a media server that is different from your signalling IP then you get NO AUDIO |
| 13:47.34 | [TK]D-Fender | You are setting yourself up for failure |
| 13:47.39 | MajesticFudgie | ah |
| 13:47.45 | MajesticFudgie | I let voipmonk set it up for me |
| 13:47.56 | MajesticFudgie | I weren't aware of this |
| 13:52.44 | MajesticFudgie | Do calls in the spool dir queue one after the other or does asterisk attempt to process them at once |
| 13:53.13 | *** join/#asterisk Akuma (~Akuma@aylmpq0104w-lp130-01-76-71-222-97.dsl.bell.ca) |
| 13:53.13 | *** join/#asterisk nanoha-sama (~nanoha-sa@van-mig-svr.ad.v10networks.ca) |
| 13:57.51 | Samot | As soon as it sees a callfile it processes it. |
| 13:58.05 | Samot | The outgoing spool is checked about every 1 second. |
| 13:58.47 | MajesticFudgie | yeah I put two files in for the same number and both seemed to have been processed at once rather than one after the other |
| 13:58.47 | masked | hi, i'm using asterisk on loopback with opensips as a sip proxy and have set media_address to the external ip, but when trying echo test i get no audio |
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| 14:11.13 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 14:15.58 | *** join/#asterisk tirej (~tirej@unaffiliated/tirej) |
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| 14:17.53 | *** mode/#asterisk [+o kharwell] by ChanServ |
| 14:19.17 | rwsq1 | Hi, I think I've enountered this issue described here: http://www.spinics.net/lists/asterisk/msg166202.html Simplest scenario is one client allows only ulaw and the other only alaw. directmedia=no for both. I would expect asterisk to transcode but instead it native bridges and there is no audio. Workaround is to enable monitor app. Is there a better solution? |
| 14:20.02 | rwsq1 | (BTW before anyone asks there is no NAT involved) |
| 14:21.10 | rwsq1 | (Asterisk 13, and seems to be the same with sip and pjsip) |
| 14:22.16 | file | I'd suggest trying branch 13 or waiting for 13.13.0 as I did media changes which should resolve those problems |
| 14:24.28 | *** join/#asterisk jeffspeff (~Jeff@12.49.160.131) |
| 14:24.35 | rwsq1 | ahh cool. I'm glad I wasn't just imagining it |
| 14:25.17 | rwsq1 | file - in the interim is there a best way to put asterisk in the media stream without incurring too much extra overhead? |
| 14:26.09 | file | no, but my media changes were only applicable to pjsip fyi |
| 14:26.19 | file | if it's happening for both then you might have a different problem |
| 14:27.28 | rwsq1 | you can definitely see the native bridge occurring in the log when it shouldn't be |
| 14:27.33 | rwsq1 | both for pjsip and chan_sip |
| 14:27.46 | rwsq1 | however if it's fixed in pjsip then that's good :) |
| 14:28.03 | file | it might be fixed, if it's the same sort of problem as reported by others |
| 14:30.18 | *** join/#asterisk Penguin (~xwQ5kwYl6@our.systems.are.full.of.penguins.at.penguinsystems.net) |
| 14:31.51 | Samot | You could show a call so we can see what is actually happening... |
| 14:32.43 | rwsq1 | I will try out branch 13 and see if that has fixed it |
| 14:33.09 | rwsq1 | may also grab a call for you although I do not have the logs immediately to hand |
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| 14:41.23 | Samot | live debugs are better than logs. |
| 14:44.35 | rwsq1 | is there a preferred pastebin for this channel? |
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| 14:49.32 | *** mode/#asterisk [+o rmudgett] by ChanServ |
| 14:51.18 | masked | how do i make the equivelant of a sip friend with pjsip? |
| 14:51.29 | masked | i wanto to allow by ip with no user/pass auth |
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| 15:14.24 | masked | hmm |
| 15:17.24 | *** join/#asterisk WIMPy (~wimpy@85.183.95.26) |
| 15:26.14 | masked | ok |
| 15:26.18 | masked | so i sorted tht out |
| 15:26.30 | masked | but when i bind the rtp/media to a different IP |
| 15:26.35 | masked | when i go to make a call |
| 15:26.42 | masked | i get 401 unauthorized |
| 15:28.37 | *** join/#asterisk Penguin (~xwQ5kwYl6@our.systems.are.full.of.penguins.at.penguinsystems.net) |
| 15:29.59 | Samot | masked: Show the results. |
| 15:45.50 | masked | Samot: ok will in a sec |
| 15:46.16 | masked | i think the problem is in the www auth it's trying to auth against a realm named "asterisk" |
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| 15:55.57 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 15:56.22 | masked | oh |
| 15:56.24 | masked | hang on |
| 15:56.26 | masked | with pjsip |
| 15:56.35 | masked | is extensions.conf still used as normal? |
| 15:57.20 | rmudgett | Yes. Dialplan hasn't changed. |
| 15:59.57 | masked | Samot: pm'd u a pastebin of the siptrace |
| 16:00.05 | masked | rmudgett: mind if i do u the same? |
| 16:03.05 | Samot | There's all sorts of wrong with that |
| 16:03.29 | Samot | You're mangling the hell out of those packets at the opensips level I'm guessing. |
| 16:03.29 | masked | uhm yeah? where do you wanna start? haha |
| 16:06.07 | masked | ok i'll do another one without binding the media address |
| 16:08.58 | masked | hmm well it would appear im still getting 401's now |
| 16:16.27 | *** join/#asterisk Jesterboxboy (~Thunderbi@80-109-194-26.cable.dynamic.surfer.at) |
| 16:25.42 | Samot | masked: Are you going to show the new attempts? |
| 16:28.33 | masked | Samot: yeh ok sec |
| 16:28.39 | masked | i've got the calls going through now with pjsip |
| 16:28.41 | masked | just no audio |
| 16:28.56 | masked | gimme a sec to continue to debug and if i still cant work it out i'll send them through |
| 16:31.42 | *** join/#asterisk Jesterboxboy (~Thunderbi@80-109-194-26.cable.dynamic.surfer.at) |
| 16:32.58 | masked | Samot: pm |
| 16:35.14 | *** join/#asterisk miralin (~Thunderbi@194.8.128.48) |
| 16:35.20 | Samot | masked: Please post these in channel so others can look at it. |
| 16:35.44 | Samot | Also, when you do this, provide the whole call. Not just three or four packets. |
| 16:35.53 | Samot | That being said.... |
| 16:35.57 | Samot | Contact: <sip:127.0.0.1:5060> <-- That's not right. |
| 16:36.37 | Samot | Because you end up with: c=IN IP4 127.0.0.1 <-- IP for the RTP |
| 16:41.47 | masked | hmm ok looking into it |
| 16:46.16 | masked | Samot: ive tried a few things, but i'm not sure how to change that contact field |
| 16:46.44 | Samot | So show a whole call. |
| 16:46.50 | Samot | I need to see it from the beginning. |
| 16:46.52 | Samot | To the end. |
| 16:48.42 | masked | thats all that hows up in asterisk console |
| 16:48.47 | masked | shows* |
| 16:50.20 | Samot | Show it again, that paste expired. |
| 16:51.42 | masked | http://pastebin.com/zCZERiYr |
| 16:52.07 | Samot | INVITE sip:AN_echo@127.0.0.1:5060 SIP/2.0 |
| 16:52.34 | Samot | This is coming from OpenSIPS? |
| 16:56.01 | masked | yeah |
| 16:56.09 | Samot | So fix OpenSIPs. |
| 16:56.25 | masked | i changed it to sip:AN_echo@faith.ski:5060 |
| 16:56.25 | Samot | Because your RURI is the 127.0.0.1 |
| 16:56.33 | masked | and the contact still remains 127.0.0.1 |
| 16:56.48 | Samot | So you made a new test call? |
| 16:59.10 | masked | yeah |
| 16:59.31 | masked | uhm and it seems rewrite_contact=on is simply not enough either.. |
| 16:59.44 | Samot | So if you made a change and made a new call.. |
| 16:59.44 | *** join/#asterisk u0m3 (~u0m3@86.120.85.26) |
| 16:59.48 | Samot | Please show it. |
| 17:00.01 | masked | sec |
| 17:02.54 | masked | actually |
| 17:02.59 | masked | im going to leave it for now |
| 17:03.00 | masked | it's 4am |
| 17:03.10 | masked | thanks for your help |
| 17:03.25 | masked | and thanks to whoever is DoSing me |
| 17:09.51 | Samot | are you implying that someone here is DoSing you? |
| 17:10.03 | Samot | Or just making a thank you note to the universe? |
| 17:20.39 | *** join/#asterisk pchero (~pchero@109.70.54.56) |
| 17:24.05 | M2K | use the power of the cloud asked, its all about the cloud |
| 17:24.30 | Samot | Make telephony great again. |
| 17:24.50 | *** join/#asterisk [NC] (~nc@rv1.sabius.net) |
| 17:25.14 | WIMPy | Just make it work again. |
| 17:25.26 | WIMPy | Small steps, you know. |
| 17:26.05 | Samot | And really, I thought it was all about the pentiums. |
| 17:26.38 | Samot | https://www.youtube.com/watch?v=qpMvS1Q1sos |
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| 18:34.17 | rrittgarn | On a 13.11.2 build the system is randomly becoming unresponsive to app_voicemail. The channels become unresponsive to channel request hangup, and the system does not play any audio. After a minimum of 15 minutes we get "Unable to stream password file" and the system resumes processing voicemails. |
| 18:35.00 | rrittgarn | looking at the code, it makes me think there is a lock or something waiting for the file vm-password where Alison so kindly says "Password" however we've formatted that into every possible codec, and have the server running from SSDs - so we don't think it's a disk IO issue |
| 18:37.14 | *** join/#asterisk newtonr (RustyNewto@nat/digium/x-qdjpsjtgmmoiabfu) |
| 18:37.14 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 18:38.34 | *** part/#asterisk jpsharp (~jsharp@c-50-154-22-18.hsd1.fl.comcast.net) |
| 18:38.40 | rrittgarn | So my question is, where should we be looking next? the server is doing ~10-20 concurrent calls into voicemail and an strace on the hung process yeilded nothing (attached after it had hung, so not surprising) - nothing shows funky with lslocks - so i'm running out of ideas short of building for debug and run production from debug so i can do core show locks, etc. |
| 18:41.09 | igcewieling | make sure DNS is working |
| 18:41.40 | igcewieling | Asterisk does all sorts of weird delays when DNS isn't working. one easy thing to do is make sure EVERY IP on the server is listed in /etc/hosts. |
| 18:43.03 | *** join/#asterisk Jesterboxboy (~Thunderbi@80-109-194-26.cable.dynamic.surfer.at) |
| 18:53.55 | masked | Samot: the universe |
| 18:54.01 | masked | Samot: its stopped now |
| 18:54.37 | masked | i tried changing contact= in the endpoint config but then it just tries to route the call back to the opensips server |
| 18:57.03 | Samot | This is an opensips issue. |
| 18:57.25 | Samot | If you remove opensips from the equation, does the endpoint register and work fine? |
| 18:57.53 | masked | opensips /is/ the endpoint |
| 18:58.14 | masked | unless you think i should be using something different than "endpoint" for the opensips connectivity |
| 19:02.52 | Samot | How is the call being generate at OpenSIPs? |
| 19:03.01 | Samot | How is the call being generated at OpenSIPs? |
| 19:03.09 | Samot | Because I see you using Zoiper.. |
| 19:03.11 | Samot | Which is the endpoint. |
| 19:03.22 | Samot | If you register your Zoiper to Asterisk is it fine? |
| 19:03.35 | Samot | Or are you registering to OpenSIPS? |
| 19:04.38 | masked | im registering it to opensips |
| 19:04.52 | Samot | OK so fix your OpenSIPS configuration. |
| 19:05.26 | Samot | Because when you get the INVITE from Zoiper, it terminates to OpenSIPS and you're the one telling OpenSIPS to route the call and how to route it. |
| 19:06.35 | drmessano | Must Asterisk on my iPhone pls |
| 19:06.37 | drmessano | Help now |
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| 19:18.22 | Samot | Get a 3.5mm input.. |
| 19:18.24 | Samot | Oh wait.... |
| 19:20.43 | igcewieling | an iPhone with a chan_dongle supported bluetooth dongle 8-| |
| 19:39.53 | masked | Samot: if i put asterisk on the realworld ip on a different port, then route the calls to that ip:port, the contact is correct and i get audio |
| 19:40.21 | masked | surely there is a way to rewrite that contact in asterisk |
| 19:40.47 | masked | if not there should be |
| 19:41.47 | Samot | masked: Zoiper -> OpenSIPS |
| 19:41.58 | Samot | That's where Zoiper sends the call. |
| 19:42.09 | Samot | What you do with it on OpenSIPS is up to you. |
| 19:42.18 | Samot | The way you are routing the call to Asterisk is not right. |
| 19:44.20 | masked | okay |
| 19:44.36 | masked | well i'm doing the opensips ebootcamp atm, we dont do this sorta stuff till next week |
| 19:44.42 | masked | so i guess i'll just wait |
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| 20:08.35 | igcewieling | Poycom phone assembly number -> phone name info, in case anyone wants it. http://pastebin.com/PWHbptE4 |
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| 21:43.42 | *** join/#asterisk [TK]D-Fender (~joe@64.235.216.2) |
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| 22:27.22 | Get_The_Fish | I have a 13.7.2 server that "all of the sudden" (according to the client) started taking 20 seconds to connect calls. |
| 22:31.44 | Get_The_Fish | it appears that Asterisk is replying to the initial INVITE with a 401, but then sits on the subsequent INVITE and does no processing for 20 seconds or so before sending an 200 |
| 22:32.03 | Get_The_Fish | Any ideas on what could be causing this/where to start looking? |
| 22:35.14 | [TK]D-Fender | DNS failures, other lockups, etc |
| 22:35.24 | [TK]D-Fender | You are 7 releases behind. upgrade and retest |
| 22:36.54 | Get_The_Fish | Unfortunately this is the freepbx distro, so my options are 13.7.2 (which I'm using), 13.9.0 and 13.9.1 which both have a nasty ODBC bug that causes crashes. Customer uses ODBC. |
| 22:37.23 | Get_The_Fish | And the call I'm making is from a UA registered to the box and dialing *97. |
| 22:37.29 | [TK]D-Fender | then either get pact packages or pick your poison |
| 22:37.33 | [TK]D-Fender | past* |
| 22:37.35 | Get_The_Fish | sorry, voicemail rather |
| 22:38.00 | [TK]D-Fender | Sucks where you are. Sucks where they offer. |
| 22:38.06 | [TK]D-Fender | <PROTECTED> |
| 22:38.21 | Get_The_Fish | Yep. |
| 22:38.56 | Get_The_Fish | Thanks [TK]D-Fender, thats what I get to tell the client. Hopefully he'll listen and we'll be install freeswitch. |
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| 22:47.09 | [TK]D-Fender | Sure, have fun with that instead... |
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| 23:31.47 | drmessano | FreePBX distro doesn't have packages past 13.9.1? |
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