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| 00:31.37 | Cubber | it seems that the yum update in asterisknow udpated php5.1 to php5.3 which has seriously broken my amportal and freepbx so now asterisk will not load. |
| 00:31.48 | Cubber | any suggestions on how I can fix this? |
| 00:32.14 | Cubber | downgrade will not work becuase dependencies are missing for php5.1 and freepbx will just be uninstalled |
| 00:32.14 | TazzNZ | Cubber: check the log files for permission errors |
| 00:32.30 | TazzNZ | the other option is check the sessions directory's permissions |
| 00:32.36 | TazzNZ | I recall something broke that |
| 00:33.25 | Cubber | this is what an amportal restart produces: http://pastebin.com/bDABgtRD |
| 00:33.52 | Cubber | along with the following on the freepbx web portal: http://pastebin.com/kZJCV455 |
| 00:33.58 | Cubber | all permissions look fine |
| 00:34.10 | TazzNZ | uhm - there seems something big wrong there |
| 00:34.37 | TazzNZ | I can't spend time on it now, maybe someone else can help, I have to run |
| 00:34.58 | Cubber | yah I have been working on this for over 3 hrs now and it is a production system |
| 00:35.07 | Cubber | needs to be online by 7 am |
| 00:35.17 | Cubber | damn updates |
| 00:35.53 | TazzNZ | I suspect there is something more than updates wrong here |
| 00:36.01 | Cubber | that is all that changed |
| 00:36.17 | Cubber | system has been running solid for months. I just did a yum update and rebooted |
| 00:36.18 | TazzNZ | I run AsteriskNOW too and am fine on php5.3 |
| 00:36.50 | Cubber | asterisk will start if I clear out /etc/asterisk |
| 00:36.54 | Cubber | otherwise it segfaults |
| 00:36.56 | TazzNZ | NO ! |
| 00:37.01 | TazzNZ | don't do that |
| 00:37.05 | Cubber | I did it to a backup directory |
| 00:37.08 | Cubber | then put it back to test |
| 00:37.25 | TazzNZ | anyways - have to run |
| 00:37.27 | Cubber | I usually use CLI only asterisk and never have an issue but this customer wanted teh GUI |
| 00:37.27 | TazzNZ | later |
| 00:37.28 | WIMPy | couldn;t start rasterisk earlier today because some lib had bben upgraded. I seriousely hate that kind of shit. |
| 00:39.43 | Cubber | I can I migrate all of my current settings to a new install if necessary? |
| 00:39.50 | Cubber | just backup DB? |
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| 02:16.54 | cdrakka | I'm running asterisk 1.8 on ubuntu 12.04, mix of SIP and Dahdi. Everything has been working pretty smoothly. The other day tried to dial into a conference. Called the number, no problem. Enter the code for the conference, and it registers every number I dialed, except for '0' ... ex: I type 220333 and it tells me I dialed 22333. Any idea why this might be happening? is 0 special? |
| 02:57.31 | [TK]D-Fender | no, you just have poor DTMF detection |
| 02:57.37 | [TK]D-Fender | And the 0's are iffy |
| 02:57.42 | [TK]D-Fender | hold it longer to test |
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| 03:10.05 | ChannelZ | Was that over a DAHDI channel? |
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| 07:23.58 | Guest65603 | morning all |
| 07:25.14 | jmls | looking for a good source of docs for the agentpool stuff in 12 |
| 07:25.36 | jmls | I found an example dialplan earlier, but now (of course) I can't refind it .. ;) |
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| 07:44.20 | WIMPy | Does pjsip allow me to use a to-domain that is different from the hostname? |
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| 08:57.04 | Ast001 | Hello I have Asterisk 1.8.27 with load of 0.50 or above when noone calls. As soon as I turn off Asterisk load come back to 0.00 which lead me to conclusion that Asterisk is doing something. Can you help me to find out what is going on ? |
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| 09:04.25 | Ast001 | here are options I use to start it http://paste.ubuntu.com/7542579/. |
| 09:06.17 | Tim_Toady | Ast001: monitor your system, use top, iotop, iptraf etc to see what causes the load |
| 09:06.54 | Ast001 | yes I am monitoring them for days turn off everything else except asterisk and has load. As soon as I turn off Asterisk load drops to 0 |
| 09:07.02 | Tim_Toady | if it is asterisk see whats going on, check the logs, login to the console with verbocity set to 3 and see if there is anything going on |
| 09:07.40 | Ast001 | I did that to and did not see anything on CLI. I turned on verbosity to 3 debug to 3 also but nothing. Core show calls showed me 0 calls. |
| 09:07.50 | Tim_Toady | you have port 5060 accessible from the internet? have you taken any measure to prevent sip attacks? |
| 09:07.58 | Tim_Toady | they are pretty common these days |
| 09:08.17 | Ast001 | yes 5060 is running sip, I use fail2ban to block any attacker on 5060 |
| 09:09.44 | Ast001 | I don't see anything in /var/log/asterisk/messages, in case of attacks I would see some ip try to register. |
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| 09:18.25 | Ast001 | I can try to disable sip and restart and check if it is because of sip. |
| 09:19.55 | Ast001 | now load falls to 0.08 after I unloaded sip module. It seems it is connected with sip. |
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| 09:20.27 | Ast001 | but I can't see any message regarding false registration attempt on messages. |
| 09:21.25 | Tim_Toady | configure the logger properly |
| 09:21.54 | Tim_Toady | also get a network trace with tcpdump or ngrep to check whats going on with SIP traffic to 5060 |
| 09:22.42 | Ast001 | /var/log/asterisk/messages => notice,warning,error that writes me NOTICE and false registration in NOTICE I should see those if any. |
| 09:27.13 | Tim_Toady | if the attacker was trying to register, he might just trying to guess extensions by sending INVITE or OPTIONS messages |
| 09:27.25 | Tim_Toady | thats why i suggested a network trace |
| 09:28.41 | Tim_Toady | also dont yet rule out other causes for this increased load, a sip attack is just one possible cause |
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| 09:51.06 | Ast001 | ok i will check trace |
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| 10:57.30 | skrusty | does asterisk support sip uri headers? e.g. sip:me@you.com?header=value |
| 10:59.08 | wdoekes | skrusty: what are those headers supposed to do? |
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| 11:00.12 | skrusty | well, reading voip-info im trying to work it out... im trying to figure out how to pass some information to an AGI that gets dialed via a SIP uri |
| 11:01.49 | skrusty | it mentions that ? is considered as sip headers, but i am unsure what converts them to sip headers and if that's true at all. passing something via ?header=value does not appear in the SIP_HEADER func |
| 11:02.51 | skrusty | sip:user:password@host:port;uri-parameters?headers |
| 11:02.58 | Tim_Toady | skrusty: pass some custom SIP headers, have asterisk read them and pass the values over to AGI |
| 11:03.27 | skrusty | i can only use a uri, i.e. i have no control over the device sending the call |
| 11:03.53 | skrusty | or initiating the call should i say, i just get to give it a SIP Uri |
| 11:04.23 | skrusty | i was hoping that the schema for it would support adding the headers via that stated above... or maybe im just being crap! :) |
| 11:05.03 | skrusty | i can see the uri is being passed to asteisk with the ?header=value in the "To" field |
| 11:05.46 | skrusty | oddly, when i do Get Varialbe SIP_HEADER(to), it returns nothing |
| 11:10.26 | wdoekes | pbx_builtin_setvar_helper(tmp, "SIPURI", i->uri); |
| 11:10.35 | wdoekes | @ skrusty |
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| 11:26.13 | skrusty | wdoekes: sorry, was afk |
| 11:27.09 | skrusty | it's not in the sip uri, that seems to return the contact header |
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| 11:28.58 | Joel_re | hey, does anyone know whats the expected latency for voice data over a gsm network? |
| 11:29.10 | Joel_re | Im seeing around 700ms to 1 sec latency |
| 11:29.13 | Joel_re | is that normal? |
| 11:30.15 | Tim_Toady | normal for a congested network i guess |
| 11:30.39 | Chainsaw | Joel_re: There is pretty hefty echo cancellation and compression involved. Hundreds of milliseconds would not surprise me. |
| 11:32.32 | Joel_re | damn hmm ok, I built this interface/game which would accept input from asterisk<->gsm |
| 11:32.56 | Joel_re | but the latency is the killer, and I havent been able to fix it |
| 11:33.01 | Joel_re | or probably will never |
| 11:34.28 | Chainsaw | Ah, you're doing in-band DTMF over GSM? |
| 11:34.52 | Joel_re | Chainsaw: yes |
| 11:35.00 | Chainsaw | Joel_re: I am impressed that it works at all. |
| 11:35.28 | Joel_re | Chainsaw: well it works with around ~1 second latency, so its not too much fun |
| 11:35.41 | Joel_re | the only good part is anyone with a mobile phone can call in and be part of it |
| 11:35.51 | Chainsaw | Joel_re: You playing tetris on the side of a building? :) |
| 11:36.39 | Joel_re | Chainsaw: heh, in this case it was just multiple characters scrawling around for a burger kinda thing |
| 11:37.05 | Joel_re | works fine over sip, but getting random people to install a client isnt fun :p |
| 11:37.58 | Chainsaw | Joel_re: WebRTC may be a useful halfway house between SIP and a browser? |
| 11:38.24 | Chainsaw | Joel_re: Now granted, it's probably going to take Asterisk 12 to run that side of things and it isn't the most mature technology in the world... |
| 11:38.44 | Joel_re | Im using the AGI, nodejs, HTML5 for this setup |
| 11:39.01 | Joel_re | but yeah webRTC is one option, I guess |
| 11:39.13 | Chainsaw | Joel_re: Just to reduce your audio latency. |
| 11:39.22 | Joel_re | ah |
| 11:39.45 | Chainsaw | Joel_re: And be able to get at the audio circuitry without tedious things like Flash, if that is part of the game. I presume if you have DTMF input there is audible confirmation of some actions. |
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| 11:41.11 | Chainsaw | Joel_re: That it circumvents call costs is just a side effect and not something I can officially encourage. |
| 11:41.42 | Joel_re | ah, let me look into webRTC, do most mobile phone browsers support webRTC though - something I should read up on |
| 11:41.50 | Joel_re | thanks for the suggestion! |
| 11:41.56 | jaflong | Hi, Is it possible to create a outgoing connection on a channel as a video call. The Dail app with video support maybe |
| 11:45.29 | skrusty | wdoekes: looks like SIP_HEADER doesn't return anything inside <..> |
| 11:45.33 | Chainsaw | Joel_re: I would expect an effort, as it means reducing the reliance on Flash. Both Android & Apple feel this way. |
| 11:46.57 | skrusty | wdoekes: i take that back, looks like it's a bug in AsterNET :/ |
| 11:48.39 | Joel_re | Chainsaw: In this case Im playing with Phasejs and the HTML5 canvase - no flash |
| 11:48.47 | Joel_re | Phaser.js |
| 11:50.52 | Chainsaw | Joel_re: Yes, that sounds like enough of the new & cool stuff that WebRTC won't phase you :) |
| 11:51.54 | Chainsaw | Nearing Kings Cross station, back later. |
| 11:52.21 | wdoekes | skrusty: ok |
| 11:52.52 | skrusty | debug time! :) |
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| 12:06.35 | mistahT | how do i solve this, when i have a dialplan context _.1, and _.n, and in the same context i have h,1. What happens is that next step from h,1 is _.,2 |
| 12:07.47 | mistahT | how do i solve this? |
| 12:08.15 | WIMPy | Do as the warning says and don't use _. . |
| 12:08.39 | mistahT | i can't use anything else but _.,1 for start because i never know the extension |
| 12:08.40 | WIMPy | That's exactely why it warns you. |
| 12:09.06 | WIMPy | It's not numeric? |
| 12:09.07 | mistahT | i just want all to go through here |
| 12:09.22 | mistahT | it is. but prefix can be different always |
| 12:09.29 | mistahT | oh,... |
| 12:09.41 | WIMPy | Use X. |
| 12:09.43 | mistahT | [0-9] right |
| 12:09.46 | WIMPy | What prefix? |
| 12:10.01 | mistahT | first number |
| 12:10.12 | mistahT | could be any from 0-9 |
| 12:10.26 | WIMPy | Or if it can be a single digit, use X! . |
| 12:10.36 | WIMPy | X = [0-9] |
| 12:11.39 | mistahT | _[0-9]. should work |
| 12:11.49 | mistahT | i see |
| 12:12.17 | mistahT | _X! right |
| 12:12.34 | mistahT | thanks, sometimes i need help to wakeup |
| 12:12.47 | mistahT | :P |
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| 13:13.14 | WIMPy | Does pjsip allow me to use a to-domain that is different from the hostname? |
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| 13:46.29 | newtonr | WIMPy, I don't believe so. What exactly do you mean? |
| 13:50.38 | newtonr | WIMPy, you can do something like Dial(PJSIP/mytrunk/sip:blah@x.x.x.x) , if you don't want to use a host that is set in the endpoints AOR/contact already.. |
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| 14:38.49 | *** join/#asterisk kodomo (~gregor@92.63.174.119) |
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| 14:56.06 | kodomo | Hi folks - is there someone who could help me debug an issue between my snom m9 and * 1.8.27.0 (TLS related, I'm afraid)? |
| 14:57.09 | kodomo | I've got the odd problem that my first call after rebooting the snom goes through (deterministically), but all subsequent calls fail... |
| 14:58.45 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 14:59.58 | kodomo | From the snom's perspective, I see the INVITE-Unauthorized-ACK-INVITE(new CSeq) sequence for all calls... but there's no reaction to the second INVITE (except for the first call) |
| 15:01.55 | *** join/#asterisk darkdrgn2k (~darkdrgn2@209.90.253.66) |
| 15:02.06 | darkdrgn2k | hwo are RTP ports assigned in sip |
| 15:02.22 | darkdrgn2k | INVITE packets (m=audio 5394 RTP/AVP 0 8 18 5 99 102) are there any othe rplaces? |
| 15:03.11 | kodomo | odd enough: on the * side (verbose, debug 9, sip set debug...), I see a parse request for the second INVITE upon the first call (but not for the ACK), and a parse request for the ACK on any subsequent call (but not for the respective second INVITE) |
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| 15:11.15 | slykens | Hello all - anyone have experience with dahdi modules running very high cpu servicing IRQs to the point that asterisk and the rest of the system struggle to function? |
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| 15:13.04 | *** mode/#asterisk [+o Qwell] by ChanServ |
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| 15:29.38 | jameswf | slykens: dahdi modules run at 1000 interrupts/sec |
| 15:29.58 | jameswf | slykens: more specifically each device |
| 15:33.38 | slykens | I have a wct4xxp that is generating 160,000 interrupts per second, sometimes more, sometimes less. |
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| 15:41.11 | rahulr92 | Hi. I'm running a test instance of Asterisk 11.8 on an Amazon Ec2 Linux Ami. I'm unable to get my asterisk manager to listen to port 5038. |
| 15:41.57 | rahulr92 | I've added tcp/udp 5038 to security groups in ec2 console. |
| 15:42.16 | rahulr92 | Any help is really appreciated. |
| 15:42.39 | jameswf | slykens: how are you calculating this? |
| 15:43.34 | slykens | cat /proc/interrupts; sleep 1; cat /proc/interrupts - then doing the math. |
| 15:44.23 | jameswf | slykens: pastebin the output of 2 iterations |
| 15:47.13 | darkdrgn2k | <PROTECTED> |
| 15:47.34 | slykens | jameswf: http://pastebin.com/ykgEzH52 |
| 15:48.34 | slykens | jameswf: fwiw, i also have a problem where sometimes on startup the kernel will disable the ira (even with irqpoll set) because it generates 100,000 interrupts immediately before the module services it. |
| 15:50.16 | jameswf | slykens: assuming those are a second apart you probably have a hardware issue (motherboard or card) I would move the card to a different slot then maybe try in a different system. Also unload the module and make sure 25 doesn't continue to increment. |
| 15:52.07 | slykens | jameswf: i've done those things and had similar problems in other systems too, even with new hardware. :/ This system is an Ubuntu 14.04 box but I'm going to build different FreePBX distro system tonight for testing on different hardware. Thanks. |
| 15:53.15 | jameswf | slykens: you may wish to poke tech support on this incase there is a known issue with a certain platform |
| 15:54.07 | slykens | jameswf: I think I might if I can't have joy with testing tonight. This platform had been working fine for about three years prior to an upgrade about a month ago and it's just been a lot of trouble since. |
| 15:54.51 | jameswf | slykens: I assume you are using genuine hardware and not some a clone |
| 15:55.31 | slykens | jameswf: Yes, both cards I have are genuine Digium. I've tried clones in the past with NO success. |
| 15:59.08 | kodomo | Does anyone have a clue as to why asterisk should parse once the 'INVITE', but not the preceding 'ACK', and then the 'ACK', but not the following 'INVITE' in an identical sequence of messages? |
| 15:59.45 | adeeln | is this a valid pattern in asterisk, _45020[0-3,5-7]XXXXXXX ? |
| 16:00.16 | Qwell | adeeln: remove the comma I believe, but yes |
| 16:00.24 | [TK]D-Fender | no "," between the 3 and 5 |
| 16:00.52 | adeeln | ok, so can't do multiple ranges in a single rule |
| 16:01.12 | Qwell | nobody said that |
| 16:02.17 | adeeln | ah, keep forgetting that the comma is a literal comma in that pattern |
| 16:10.19 | leifmadsen | you can do [0-35-7] |
| 16:10.22 | leifmadsen | that is valid |
| 16:10.38 | leifmadsen | every position is 1 char long, so it won't try to match like 35 or something |
| 16:10.53 | leifmadsen | [a-zA-Z0-9*#] is valid for example |
| 16:10.56 | leifmadsen | I use that all the time |
| 16:11.24 | leifmadsen | also, remember that [ ] is a single matching position in the pattern match too |
| 16:11.33 | leifmadsen | e.g. [0-9] would be the equiv of X |
| 16:18.09 | *** join/#asterisk CeBe (~CeBe@brsg-4dbb1a2b.pool.mediaWays.net) |
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| 16:42.59 | jameswf | Just _. #yolo |
| 16:47.01 | Qwell | _! |
| 16:47.02 | Qwell | amateur :p |
| 16:49.38 | *** join/#asterisk davlefou (~davlefou@unaffiliated/davlefou) |
| 16:51.56 | navaismo | always change the context of amateur word (thanks-internet) |
| 16:59.42 | leifmadsen | Qwell: match zero or more chars! |
| 16:59.43 | leifmadsen | nice |
| 17:02.31 | Qwell | leifmadsen: I don't remember exactly what it was, but Tilghman fixed some bug with: |
| 17:02.39 | Qwell | exten => ,1,SomeApp() |
| 17:03.21 | mjordan | for some reason, that makes me sad |
| 17:03.26 | Qwell | everyone was like "wait, what?" |
| 17:03.37 | paulc | .."no extension"? |
| 17:03.58 | leifmadsen | heh |
| 17:03.59 | Qwell | paulc: No, no, that was the beauty of it. There *was* an extension. It was just zero length. |
| 17:04.00 | leifmadsen | run all the things |
| 17:04.07 | leifmadsen | always run all the things |
| 17:04.22 | leifmadsen | that totally sounds like a bug Tilghman would fix :) |
| 17:04.45 | mjordan | I'm trying to think of a channel driver that will send a channel to a context but not provide an extension |
| 17:05.03 | mjordan | even chan_sip will default things to 's' |
| 17:05.17 | Qwell | I don't recall |
| 17:05.19 | *** join/#asterisk Dovid (~Dovid@ool-2f113d03.dyn.optonline.net) |
| 17:05.28 | mjordan | wonders if Local/@default would work |
| 17:05.34 | leifmadsen | maybe! |
| 17:05.37 | Qwell | There was some weird semantic where s was the wrong thing to do. |
| 17:05.45 | leifmadsen | s is always the wrong thing to do |
| 17:05.50 | mjordan | until it isn't |
| 17:05.54 | leifmadsen | then it is |
| 17:06.00 | mjordan | wrong? |
| 17:06.04 | Qwell | yes |
| 17:06.17 | leifmadsen | unknown until it has been viewed |
| 17:06.36 | mjordan | Warnings be damned, man the torpedoes and _. ahead |
| 17:06.47 | Qwell | All of the 'shit' extensions are silly. |
| 17:07.02 | mjordan | yeah. Yet another thing to go back in time and undo. |
| 17:08.00 | paulc | I remember an issue ages ago using AMI to transfer a channel to a different extension, and the h extension firing.. which was weird, and unexpected.. we got round it by having an h-less context that just GoTo'd the right place.. kludgey fix but it worked.. I wonder if that still happens.. I should try sometime.. |
| 17:08.23 | paulc | (and test that CURL in the dialplan still blocks, as per https://issues.asterisk.org/jira/browse/ASTERISK-18708 - personal favourite of mine) |
| 17:08.26 | mjordan | suspects masquerades, in the library, with the wrench |
| 17:10.03 | mjordan | paulc: fairly confident that redirects no longer ever execute the 'h' extension on the masqueraded channel. We fixed a bunch of that stuff about a year or two ago. It _definitely_ won't happen in 12+. |
| 17:10.18 | paulc | @mjordan |
| 17:10.48 | paulc | @mjordan: yeah, I figured it was a bug.. not a big deal but I'd test it again out of curiosity, during a "quiet period" (I'm having a few of those lately) |
| 17:12.52 | *** join/#asterisk davlefouAMD (~david@41.225.229.110) |
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| 17:23.17 | kodomo | ...my problem's very much like this issue... https://issues.asterisk.org/jira/browse/ASTERISK-19003 ... except that it works exactly once before breaking down... |
| 17:23.34 | kodomo | (I do have enabled the compact headers) |
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| 17:48.04 | anonymouz666 | anyone using asterisk with MS LYNC? |
| 17:48.12 | anonymouz666 | LYNC 2013 |
| 17:48.47 | *** join/#asterisk roxlu (~roxlu@5469BA8D.cm-12-2c.dynamic.ziggo.nl) |
| 17:48.52 | roxlu | hi |
| 17:48.54 | matthew-moretalk | Hi Guys really strange problem im having and its been driving me crazy. trying to setup a console monitor which will run asterisk verbose all the time in our office. I ssh into one of our asterisk boxes and run asterisk -r and set verbose to 4. But the console and SSH Connection crashes every single time there is a goto statement in the verbose! |
| 17:49.40 | matthew-moretalk | p.s the ssh client is a raspberry Pi. If I do it from my desktop with putty its fine |
| 17:51.08 | paulc | matthew-moretalk: specifically a goto? as opposed to gosub or any other command/app? that's.. weird.. |
| 17:51.16 | paulc | how about running the console through screen - any difference? |
| 17:51.26 | TazzNZ | anonymouz666: we PoC'ed it |
| 17:52.27 | anonymouz666 | Takapa: did you make the TRANSFER work? |
| 17:53.21 | matthew-moretalk | if I run console though screen (on Pi) it crashes the whole Pi! and specifically a Goto |
| 17:53.59 | TazzNZ | anonymouz666: So a call came into Lync 2013 and transfer it to Asterisk ? |
| 17:54.43 | roxlu | I've been reading up on STUN and understand the process but the RFC doesn't really mention what to do with the data you receive (or I missed it). Therefore I was wondering, lets say I receive the reflexive address/port, what does stun expect me to do with it? (Create a listening socket?) |
| 17:55.33 | *** join/#asterisk BakaKuna (~Thunderbi@82-169-251-128.ip.telfort.nl) |
| 17:56.31 | anonymouz666 | TazzNZ: you call from lync. lync sends to asterisk. asterisk sends to pstn. |
| 17:56.44 | anonymouz666 | then your transfer from lync, to another pstn number. |
| 18:00.58 | *** join/#asterisk newtonr (~newtonr@173-17-135-67.client.mchsi.com) |
| 18:00.58 | *** mode/#asterisk [+o newtonr] by ChanServ |
| 18:02.50 | andi | Hi |
| 18:03.02 | matthew-moretalk | hey! |
| 18:03.28 | andi | Is there an easy way to check if I can connect to a remote pbx? Maybe something like tcptraceroute for udp? I don't know if this is technically possible. |
| 18:04.21 | matthew-moretalk | if you know the Sip Port the pbx is running on you could try telnet? |
| 18:04.21 | Nugget | telnet is eeeeeeevil! |
| 18:04.50 | andi | My provider is telling me that his server cannot reach my asterisk, but I opened up my firewall and I do not have a router in my setup. Therefore the server with asterisk running is at the moment directly connectable from the internet. |
| 18:04.54 | matthew-moretalk | you said that before id even wrote my comment Nugget! |
| 18:05.23 | matthew-moretalk | so I assume you can ping it from the outside world? |
| 18:05.30 | andi | yes |
| 18:05.53 | andi | Do I need to use special udp parameters for telnet? |
| 18:06.22 | matthew-moretalk | have you tried registering a device to the server for example a SIP Handset? |
| 18:06.59 | matthew-moretalk | is it that your provider cant register with your pbx or cant ping it? |
| 18:08.00 | willwh | is it possible to do something like dial in radio with asterisk? |
| 18:08.14 | willwh | user calls in and gets connected to an icecast/shoutcast feed or something similar? |
| 18:08.31 | matthew-moretalk | yes is the simple answer willwh |
| 18:08.42 | matthew-moretalk | I would suggest google is your best friend :) |
| 18:09.56 | andi | matthew-moretalk: Oh, I'm using another sip provider which is working quite well and I have connected some snom 370 phones. :) |
| 18:10.20 | andi | Just this providers has problems to qualify with OPTIONS packets. |
| 18:11.40 | andi | They said they can ping my host, but cannot register. But they have a large setup, therefore I do not know if the host from which they pinged my machine is the same as the one which trys to register with my pbx. |
| 18:12.09 | matthew-moretalk | arr okay iv had that. do some researching into qualifying with notify packets |
| 18:12.43 | matthew-moretalk | or the other option is dont register with them at all and get them to use IP Based Authentication for outbound routing and Sip Endpoints for Inbound |
| 18:12.47 | matthew-moretalk | thats what we do |
| 18:13.03 | andi | They are trying to use ip based authentication, sorry. |
| 18:13.25 | andi | Ok, so they do just qualify and do not register at all. |
| 18:14.08 | *** join/#asterisk hecatae (~philip@host-92-28-0-220.as13285.net) |
| 18:14.15 | matthew-moretalk | for Outbound if your using ip based auth there should be no need to register or qualify at all |
| 18:14.18 | hecatae | hi everyone |
| 18:14.24 | hecatae | any one use Bicom? |
| 18:14.28 | andi | The strange thing is my pbx says everything is ok. I can qualify their host, but they cannot qualify mine. |
| 18:14.49 | matthew-moretalk | strange. tried insecure=invite,port? |
| 18:15.00 | matthew-moretalk | in your trunk config |
| 18:15.14 | andi | What is it then called if they are sending me options packets to set my pbx active and if it's active they will redirect calls to my pbx. |
| 18:15.39 | andi | Sure, I already tcpdumped the traffic. I can see my options packets, but not theirs. |
| 18:16.28 | andi | They are telling me that my internetconnection is kind of broken, therefore the packets cannot pass through. But I do have a Gigabit Port from my housing provider. |
| 18:16.43 | andi | Hmm, I have an idea... |
| 18:16.49 | matthew-moretalk | go on? |
| 18:17.14 | andi | My pbx is a kvm host on one of my servers. I'm trying to sniff the packets at the servers physical network interface. |
| 18:17.24 | *** join/#asterisk dan_j (~IceChat77@unaffiliated/danfromuk) |
| 18:17.45 | matthew-moretalk | arrr try sniffing at your vm then |
| 18:17.59 | matthew-moretalk | see if there getting lost somewhere in between the vm and host |
| 18:18.17 | andi | That's what I already did. The packets do not arrive at the vm. Now I'll have a try at the physical nic. |
| 18:18.27 | matthew-moretalk | ok |
| 18:19.06 | dan_j | Hi. When mixmonitor is recording, does it record directly into the destination location? If yes, how can I tell if the recording has finished (from outside of asterisk). |
| 18:19.37 | kodomo | asterisk appears to shut down the TLS connection after the snom sent its second INVITE (i.e., my asterisk machine sends a TCP FIN)... *sigh* |
| 18:19.40 | *** join/#asterisk grimzz (~bidon@modemcable230.78-37-24.static.videotron.ca) |
| 18:20.08 | grimzz | heya! |
| 18:20.44 | grimzz | have a problem with business Avaya IP Phone |
| 18:20.59 | grimzz | i know its Asterisk here, but i got nothing to loose |
| 18:21.22 | paulc | dan_j: I look for files with a modified time > a few minutes ago |
| 18:21.25 | *** join/#asterisk Alex_Bkash (cbdf5c49@gateway/web/freenode/ip.203.223.92.73) |
| 18:21.39 | grimzz | something in conversation, voice get muted for like 10 seconds, then comeback |
| 18:21.44 | dan_j | paulc: I was worried you were going to say something like that. |
| 18:22.50 | paulc | dan_j: I actually have a PHP script running that talks to AMI and monitors stuff, looking for link and unlink events, so you could tell that way.. but mostly, for the volume that I'm looking at, I just periodically check dates/times on the files |
| 18:25.31 | dan_j | I'm using asterisk 11.5.1 and it seems to be broken. When i start recording, I set a variable __RECORDING=1. But in the h extension, it doesnt appear to have the variable set. Other __variables seem to be listed fine. |
| 18:25.49 | dan_j | The only difference i can see is that the __recording variable is set within a macro |
| 18:26.09 | dan_j | Has anyone seen an issue setting __variables from within a macro? |
| 18:26.36 | paulc | dan_j hmm.. if you change the variable to __DANRECORDING, do you still lose it in the h extension? |
| 18:27.03 | paulc | gets dragged away by coworkers - back in a bit |
| 18:34.01 | *** join/#asterisk Johnny5 (~Johnny5@nat/digium/x-gszlmckgicixckrn) |
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| 18:38.50 | kodomo | help!... no one here who's familiar with asterisk/TLS ? This issue's driving me nuts... :-| |
| 18:43.31 | mjordan | kodomo: People are familiar with it, they either (a) aren't familiar with whatever you're seeing, or (b) you're running into the issue that occurs when packet sizes get large. |
| 18:44.04 | mjordan | While there are patches that work around that issue, no one has solved the root cause of that problem, which is that the TLS implementation in Asterisk incorrectly handles large packets when reading from its FILE pointer |
| 18:44.20 | mjordan | You can try the patches that work around the issue |
| 18:44.36 | mjordan | You can try getting the phone/Asterisk to send smaller pickets (hint: reduce the size of the SDP) |
| 18:44.41 | mjordan | s/pickets/packets |
| 18:44.52 | mjordan | Or you can fix the TLS implementation in Asterisk (or pay someone to do it) |
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| 18:55.01 | kodomo | mjordan: unfortunately, packet size seems _not_ to be the issue |
| 18:56.21 | kodomo | I actually am looking at the source code... but it would be an immense help to chat with someone who's familiar with it... |
| 18:58.38 | kodomo | (I've been the one reporting abovementioned issue... and the workarounds worked for me with the old version of asterisk... but they don't with 1.8.27.0) |
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| 19:22.22 | newtonr | kodomo, if you are looking at the source code and have questions about it, you can try asking in #asterisk-dev . Thats where development discussions are held |
| 19:27.23 | dan_j | When starting recording by using a dtmf code, is there any way to choose the channel that the recording should start on? The reason I ask is that at the moment, if recording is started then the call is transferred, the recording stops on transfer. |
| 19:27.51 | dan_j | If I can choose the caller's channel, then the call should be recorded till the final call ends. |
| 19:28.43 | kodomo | newtonr: ah - thx - thus 'wrong channel' - thx... |
| 19:31.26 | dan_j | I've set AUDIOHOOK_INHERIT(MixMonitor)=yes but that doesnt seem to help. the recording still ends on transfer |
| 19:35.54 | dan_j | Any ideas how I can diagnose this or what outputs I need to post in order for you to assist? |
| 19:50.52 | leifmadsen | dan_j: yes, you choose the channel through the configuration in features.conf to operate on self or peer |
| 19:51.06 | leifmadsen | you need to start the recording on the channel that is going to continue existing after the transfer |
| 19:51.16 | leifmadsen | if the channel that was performing the transfer goes away, and recording is on that channel, it will stop |
| 19:51.26 | leifmadsen | call recording in Asterisk is a bitch to deal with, especially around transfers |
| 19:52.06 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 20:05.13 | *** topic/#asterisk by mjordan -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: Asterisk 11.10.0 (2014/05/29), 1.8.28.0 (2014/05/29); Standard: Asterisk 12.3.0 (2014/05/29); DAHDI: DAHDI-linux 2.9.0 (2014/01/30), DAHDI-tools 2.9.0.1 (2014/01/31); libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- Read the Code of Conduct bit.ly/1hH6P22 |
| 20:05.42 | mjordan | leifmadsen: except in 12. |
| 20:05.45 | mjordan | just sayin'. |
| 20:06.46 | dan_j | leifmadsen: thanks. I'll check it out. |
| 20:06.57 | navaismo | feature request: CRTL+R in asterisk console |
| 20:07.00 | Chainsaw | mjordan: Is that a security upgrade? Just so I know whether I handle this tonight or tomorrow morning. |
| 20:07.41 | leifmadsen | mjordan: +1 |
| 20:08.05 | leifmadsen | mjordan: thinking about just externalizing it with rtpproxy and keeping asterisk entirely on private IPs and limiting the external entities |
| 20:08.57 | WIMPy | newtonr: Sorry. I was away. The issue I have is that chan_sip uses whatever you specify in the host=line as domain part of the to header. |
| 20:10.18 | mjordan | Chainsaw: just a normal one |
| 20:10.33 | Chainsaw | mjordan: Ah, good. Thanks. |
| 20:10.33 | mjordan | Chainsaw: if was a security release, you'd get a single release announcement stating as such |
| 20:11.17 | Chainsaw | mjordan: They're so neatly on the same day on all branches, I got a bit worried. |
| 20:11.27 | mjordan | leifmadsen: for your kind of set ups... yes. Asterisk should be a big old dumb media/bridging engine that does nothing with the outside world |
| 20:11.38 | mjordan | Chainsaw: we usually do that :-) |
| 20:11.49 | Chainsaw | mjordan: Duly noted :) |
| 20:12.43 | dan_j | leifmadsen: i changed it to peer, instead of self, however now the peer hears the announcement that says "recording", rather than the side that triggered the recording. |
| 20:13.31 | dan_j | Is there anywhere to fix that? IE, can I use peer, but play an announcement to 'self'? |
| 20:13.37 | leifmadsen | welcome to my nightmare :) |
| 20:15.24 | dan_j | Oh dear. A part of me is wondering whether i'm going to start having to look for an alternative solution. |
| 20:15.32 | dan_j | Has it been improved in later versions? |
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| 20:25.10 | [TK]D-Fender | Later than? |
| 20:25.19 | dan_j | 11.5 |
| 20:25.44 | [TK]D-Fender | [15:51]leifmadsencall recording in Asterisk is a bitch to deal with, especially around transfers [16:05]mjordanleifmadsen: except in 12. |
| 20:26.01 | dan_j | didnt know that was aimed at me |
| 20:27.19 | dan_j | I think i'm going to have to wait for 13 rather than going to 12 which is EOL quite soon. |
| 20:28.30 | dan_j | I know features can start on either self or peer. I was wondering if its possible to start two separate features using the same dial code. |
| 20:28.48 | dan_j | Therefore I'd be able to start recording on the peer, but playback a "recording started" message to the 'self' |
| 20:29.27 | [TK]D-Fender | checkout time, BBL |
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| 21:00.01 | kodomo | if anyone cares: the issue was a race condition in sip_tls_read in line 3031 of channels/chan_sip.c of asterisk 1.8.27.0 :-| |
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| 21:19.01 | navaismo | jira page? |
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