| 00:00.40 | WIMPy | qmr, anthony[1]: Loos like the peer is unreachable. |
| 00:01.08 | anthony[1] | ok, what do you mean? Does Peer mean SIP client? |
| 00:01.21 | WIMPy | yes |
| 00:01.29 | qmr | WIMPy: as in the person I'm trying to call? |
| 00:02.00 | anthony[1] | Thank U for response, I am testing Asterick on my PC, Asterick as well as two soft phones are all installed on one PC |
| 00:02.12 | WIMPy | Only if that person has an IP connection and talks sip. |
| 00:03.31 | anthony[1] | Both sip clients can call Asterisk and hear playback(hello work) so I wonder why they cannot call eachother |
| 00:03.50 | WIMPy | Unregistered? |
| 00:03.56 | [TK]D-Fender | anthony[1]: Because * has no IP to contact the peer you are tryig to dial |
| 00:04.04 | [TK]D-Fender | They have not registered |
| 00:04.30 | anthony[1] | They are registered in Sip.conf and X-lite user interface confirms registration, so I wonder what could be problem |
| 00:05.32 | anthony[1] | what is best way to confirm regitration of SIP clients on Asterisk? |
| 00:05.46 | [TK]D-Fender | anthony[1]: "sip show peer X" |
| 00:06.09 | anthony[1] | Where X is IP of SIP client? |
| 00:06.10 | [TK]D-Fender | anthony[1]: PASTEBIN it along with your failed call with SIP DEBUG enabled. "sip set debug on" |
| 00:06.23 | [TK]D-Fender | X = PEER NAME from sip.conf |
| 00:06.37 | [TK]D-Fender | [whatyouputhere] |
| 00:06.53 | qmr | is there a bit to flip to allow registration on the server? |
| 00:07.02 | anthony[1] | sip show peer 1000 and sip show peer 1001 |
| 00:07.46 | enki | Is there a way I can get Asterisk to register with something other than "s@IP", and have something different depending on the trunk? |
| 00:07.59 | qmr | where do I get this magical sip command.. doesn't seem to be installed |
| 00:08.16 | [TK]D-Fender | qmr: Show us that it isn't. |
| 00:08.18 | [TK]D-Fender | ~pb |
| 00:08.19 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 00:08.20 | [TK]D-Fender | ^^^^ |
| 00:08.24 | WIMPy | enki: Either user registrt ..../exten or callbackexten=exten |
| 00:08.33 | WIMPy | -r |
| 00:08.42 | WIMPy | I should go to sleep |
| 00:08.44 | enki | is callbackexten a sip.conf variable? |
| 00:08.54 | WIMPy | enki: Either use register ..../exten or callbackexten=exten |
| 00:08.56 | qmr | [TK]D-Fender: not in my path maybe? |
| 00:09.04 | WIMPy | yes |
| 00:09.06 | [TK]D-Fender | enki: register => user@host/youshouldputsomethingheresoasteriskdoesntjustput"s" |
| 00:09.17 | [TK]D-Fender | qmr: No such thing as ""path |
| 00:09.21 | enki | ah, awesome thanks to both of you |
| 00:09.22 | [TK]D-Fender | qmr: SHOW US |
| 00:09.52 | enki | ive been running into issues because I have two trunks setup to the same server (different accounts), but inbound calls would always go to the first trunk and outbound calls on the second one would fail |
| 00:09.54 | qmr | ooh, asterisk -r puts me on conosle |
| 00:10.12 | enki | was getting digest authentication issues on outbound |
| 00:10.38 | carrar | 407 |
| 00:10.42 | carrar | proxy authenticate! |
| 00:10.43 | enki | I got that fixed (added fromuser and authname - not sure which fixed it but it's working now) |
| 00:11.05 | enki | but i needed some way of diferentiating inbound calls. |
| 00:11.49 | qmr | how do I allow registration on the server? |
| 00:12.04 | WIMPy | qmr: host=dynamic |
| 00:12.08 | [TK]D-Fender | enki: Register statments have nothing to do with matching the inbound. Now that MAY tell them an EXTEN to dial back with that you can identify but has nothing to do with the PEEER that * may attribute the call to. |
| 00:12.37 | enki | hrmmm. so the calls for both trunks (inbound) will still mostly default to the first trunk's default context? |
| 00:13.08 | qmr | WIMPy: user.conf , sip.conf , ... ? this is annoying because just using extension without registering things werked before ... not sure what changed, and if the error is in my client or server |
| 00:13.28 | carrar | BUICKs have large trunks |
| 00:13.43 | [TK]D-Fender | enki: typically its the same host IP and often they won't send you a username or even really auth the call at all. Since the IP will be the same you need to judge based on the exten they send or some other bit in the packet. |
| 00:14.49 | [TK]D-Fender | qmr: what you consider as "working" before doesnt tell us anything. For all we know * was rejecting auth and treating a call FROM your device as an unauthed misc call... |
| 00:14.51 | enki | [TK]D-Fender, correct, they don't. Originally I found a differating sip header that I was using, but I just found out that, depending on the way the call comes in, it's not always present. But if using the /xxx after the register statement will send the calls into xxx@me, then I can use that to route them internally. |
| 00:15.27 | [TK]D-Fender | qmr: And you are asking rather vague things like "Is there a magic bit to flip to allow registration". Look at your calls and actual communications with your peer. And then SHOW US. |
| 00:15.45 | carrar | How about a Magic Cookie! |
| 00:16.07 | carrar | noms on some z9hG4bK |
| 00:20.39 | enki | will the callbackexten=xxx part force it into a certain extension in the context specified for the trunk in sip.conf? |
| 00:21.35 | [TK]D-Fender | no |
| 00:21.36 | qmr | [TK]D-Fender: I'd be happy to show you whatever if you specify what to show. sorry if you already did and I missed it |
| 00:21.55 | carrar | TK, should add "showme" to the bot |
| 00:22.02 | carrar | ~showme |
| 00:22.41 | [TK]D-Fender | carrar: thats also my "catch-prase" at the ooffice. Basically because people's use of "story time" with useless crap gets everyone nowhere. |
| 00:22.57 | carrar | haha |
| 00:23.05 | [TK]D-Fender | qmrwhere do I get this magical sip command.. doesn't seem to be installed |
| 00:23.23 | [TK]D-Fender | [TK]D-Fenderqmr: Show us that it isn't. |
| 00:23.34 | carrar | module load magical_do_everything.so |
| 00:23.53 | [TK]D-Fender | qmr: Go to * CLI. Enable SIP DEBUG ( I already showed HOW). LOOK at the actual calls and comms. |
| 00:24.19 | WIMPy | qmr: Eitehr you specify the host in your config or you register. |
| 00:24.36 | [TK]D-Fender | enki: What you shove on your register statement will have NO impact on what PEER * matches the call to. |
| 00:26.02 | enki | Gotcha. What I meant was, if I use that line in the sip.conf entry for the trunk the calls are coming in under, will that always send it to that exten in whatever context is defined regardless of what's in the invite? |
| 00:26.41 | carrar | Someone needs to write a book! |
| 00:26.46 | [TK]D-Fender | enki: No. The opposite. It tells them what you want as you register and they do whatever the hell THEY feel like. |
| 00:27.11 | [TK]D-Fender | enki: But if youre lucky and it would be relevant to do so, maybe they'll use it |
| 00:27.16 | [TK]D-Fender | ~osmosis |
| 00:27.17 | infobot | [~osmosis] Osmosis is the act of beating yourself on the head repeatedly with THE BOOK, until some measure of absorption has occured ... or at least until your unconsciousness restores peace to the channel ... |
| 00:28.04 | enki | I was able to fix inbound calls using the register=>...../123 setting. I have both trunks sending calls to the same default context, which uses _X. as the extension in there. I then do some callerid fixes, and sent the call to {EXTEN}@different-context, and process them (depending on which trunk they came in on) that way. |
| 00:28.42 | [TK]D-Fender | enki: never even use an actual context named [default] |
| 00:29.08 | enki | im not |
| 00:29.09 | [TK]D-Fender | enki: for the rest, glad its working |
| 00:29.13 | enki | i meant MY default context :D |
| 00:29.39 | enki | which is $providername_remote-trunk-in |
| 00:30.00 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 00:30.55 | enki | on the outbound side of things, do you know why those calls were originally failing? I think I have that fixed with those added lines in sip.conf, but I'm curious why it was failing digest authentication since * was trying to use the correct authentication. Calls from the first trunk worked, the second one failed. |
| 00:30.55 | [TK]D-Fender | enki: Sounds OK then |
| 00:31.09 | [TK]D-Fender | enki: I'm not going to play psychic on that one.... |
| 00:31.45 | enki | <PROTECTED> |
| 00:32.00 | enki | it basically was getting the correct user, but thought it should be the other one. |
| 00:32.36 | [TK]D-Fender | that sounds like incoming, not outgoing.... |
| 00:32.52 | [TK]D-Fender | Again.... I debug and comment on things I can see.... |
| 00:32.55 | carrar | Oh come on TK, you know you magically know |
| 00:32.59 | enki | Adding fromuser= and authname= seems to fix it. I can make calls on either trunk (even at the same time) with no issue. except if I try to call one from the other directly, then it fails. But I have no need to ever do that outside of when I just tested it. |
| 00:33.30 | enki | [TK]D-Fender, well, it's technically outbound from my PBX's perspective, but inbound from my provider's perspective (which I work for :D) |
| 00:33.32 | carrar | TK knows all |
| 00:34.03 | carrar | He's hooked up with some serious crystal |
| 00:35.14 | [TK]D-Fender | Ironic given my medicine cabinet would attract attention in the US |
| 00:36.05 | enki | i assume it was authname that actually fixed the digest authentication. I guess without that, it was sort of an inverse error, where the provider side was seeing a call coming in from what it thought was a different peer. |
| 00:43.13 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 00:44.29 | enki | on an unrelated note, I won one of those Digium D40 phones in their facebook give away. While I am not running the "certified branch" and thus can't take advantage of some of the integration, I have to say that as a standard SIP phone, I am woefully unimpressed. |
| 00:45.59 | [TK]D-Fender | enki: I have a number of specific points of contention as to their implementation as well... |
| 00:47.26 | enki | I might fire up a test box to play around with those features, but I tested it on a standard 1.8 box. The phone itself is severely limited in functionality, the configuration options (especially the webpage) are minimal, and the user interface on the phone is not very intuitive. |
| 00:48.09 | enki | With the hard phone market so competitive nowadays, I'm shocked that Digium would suddenly jump into it without some impressive features to make it a 'must have'. |
| 00:50.22 | [TK]D-Fender | enki: well they do have specific API's that help it sell with their commercial product Swirchvox |
| 00:50.38 | [TK]D-Fender | enki: Remeber that it is sold as 2 flavours : community & proprietary |
| 00:50.58 | enki | True. |
| 00:52.09 | *** join/#asterisk nanoha-sama (~nanoha-sa@nanoha-sama.freenode.bouncers.smb.curriegrad2004.ca) |
| 00:52.28 | enki | I picked up two other play phones recently too. The Yealink VP530 and the CloudTC Glass 1000. The Yealink is pretty sweet, and I've been playing around with the video functionality. I really like Yealink phones. The CloudTC on the other hand......horrible. |
| 00:52.43 | enki | They took a pretty neat idea (IMHO) and executed poorly on it. |
| 00:54.05 | [TK]D-Fender | enki: yealink ripped off Cisco & Polycom design ideas and Feels like a cheaper version of either. |
| 00:55.33 | enki | well, I personally dislike the Cisco (SPA) phones. Outside of the 525G2, they feel cheap to me and have a lot of bugs. Polycom, outside of the Kirk line, is great. I think the Yealink is the next best option. It might not be as solidly built as Polycom, but they have some nice features and the price point is great. |
| 00:56.09 | [TK]D-Fender | enki: enki I meant the REAL Cisco, not the Linksys inherited ones |
| 00:57.03 | [TK]D-Fender | enki: I have a thing for audio and build quality. There is a massive difference between my D40 & any Polycom for example. |
| 00:57.18 | [TK]D-Fender | enki: Plenty of opinions on Aastra here as well |
| 00:59.36 | enki | I've actually never used Aastra. We used to sell Snom, stopped. Grandstream is utter crap. Right now we sell Cisco (SPA), Yealink and Polycom primarilly. Customers, hands down, prefer Yealink over Cisco(SPA). We support the 79xx Ciscos, but don't sell them. They are well built phones, but their SIP functionality is seriously lacking, and Cisco has no intention of fixing it. |
| 00:59.59 | *** join/#asterisk mjordan (~mjordan@user-24-214-136-35.knology.net) |
| 00:59.59 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 01:02.47 | [TK]D-Fender | enki: Every product has their flaws. Polycom's is writing on screen to small on higher models, lack of built in BLF's |
| 01:03.18 | [TK]D-Fender | SPA = shittier screen, weight too low. Slightly tinny speakerphone & handset. |
| 01:04.00 | [TK]D-Fender | 79XX = expensive, poor SIP implementation, and licensing |
| 01:04.01 | enki | Yeah, it boggles my mind that Polycom doesn't release a phone with built in BLF keys. We've been asking for that for years and they keep telling us "it's coming soon!" |
| 01:04.29 | [TK]D-Fender | Aasta has some serious plusses, but equally serious backdraws |
| 01:05.31 | carrar | Digium phones still have a DHCP issue? |
| 01:05.36 | [TK]D-Fender | Aastra = tinny audio all around, handset has NO weight, phone gets dragged around. SHIT rubber buttos, and horrific use of even a large LCD on 5i series. Crashes more than many others as well. |
| 01:05.51 | enki | carrar, mine didn't. That was the only thing it did well ;). |
| 01:06.03 | carrar | I test all 3 |
| 01:06.06 | carrar | they all were broked |
| 01:06.09 | enki | Grandstream is by far the shittiest phone I've ever touched. |
| 01:06.10 | carrar | tested |
| 01:06.25 | carrar | giving the tftp server via dhcp |
| 01:07.48 | enki | I think my biggest pet peeve (so far) with the D40 is the back button varies between the fourth soft key and the Line1 button. |
| 01:09.14 | *** join/#asterisk bmg505 (~leon@196-210-141-90.dynamic.isadsl.co.za) |
| 01:09.18 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 01:10.19 | [TK]D-Fender | enki: that took me foreever to figure out. that it psychotically unituitive |
| 01:10.30 | [TK]D-Fender | is* |
| 01:10.47 | enki | carrar, upon further investigation, I don't think option66 works on mine. I provisioned it by hand so didn't try that. But I don't see it even attempting to pull a config. |
| 01:11.05 | [TK]D-Fender | enki: Writing is TINY, and the freakiest thing : no physical hookswitch. a mag sensonr ... on the MOUTHPIECE side |
| 01:11.07 | carrar | yup |
| 01:11.18 | enki | [TK]D-Fender, OMG, I never noticed that |
| 01:11.19 | carrar | makes it a SUCK for a mass deployment option |
| 01:11.32 | carrar | in a remote from the PBX setup |
| 01:11.52 | [TK]D-Fender | enki: I like the physical switch.... hold handset, slap phone... |
| 01:11.57 | enki | carrar, yeah, I was saying before how it was really built around that specialized Asterisk implementation. But if you don't use it, it's a horrible phone. I can't speak to how it works with that active since I haven't tested it yet. |
| 01:12.12 | enki | [TK]D-Fender, same here. |
| 01:12.36 | enki | it does seem (physically) to use the same EHS as the polycom though. |
| 01:13.23 | enki | also, the volume adjustment only shows a numeric, not a sliding scale. I dislike that. |
| 01:13.53 | [TK]D-Fender | enki: firware re-write would do a lot for them... |
| 01:14.28 | [TK]D-Fender | enki: Actually, the D40 looks like a cross between Polycom & SPA |
| 01:15.27 | enki | [TK]D-Fender, yeah, I thought the same thing. More like an SPA, but has some Polycom traits. |
| 01:15.46 | enki | But the firmware needs a complete overhaul to be a viable phone. |
| 01:16.01 | enki | Speaking of wackiness, I noticed RCA has a VoIP line now too |
| 01:16.11 | enki | i didnt even realize they were still in business. |
| 01:17.48 | *** join/#asterisk kikohnl (~keith@udp278187uds.hawaiiantel.net) |
| 01:27.38 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 01:29.10 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 01:37.08 | [TK]D-Fender | checkout time, later all |
| 01:38.49 | *** join/#asterisk bmg505 (~leon@196-210-141-90.dynamic.isadsl.co.za) |
| 01:56.38 | *** join/#asterisk deo (~dnepangue@203.177.214.75) |
| 01:57.54 | anthony[1] | How to fix Asterisk boot error: Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist? |
| 02:01.57 | anthony[1] | afer editing SIP.conf and extensions.conf, group and owner become root. Is it best to edit conf files as user asterisk? |
| 02:02.02 | anthony[1] | if so, how? |
| 02:08.25 | carrar | su - asterisk; vi /etc/asterisk/sip.conf |
| 02:11.21 | anthony[1] | thank you |
| 02:13.00 | enki | you probably don't want the '-' as there is no need to run login scripts |
| 02:13.09 | enki | su asterisk is sufficient to change the user |
| 02:13.15 | anthony[1] | Thank you so much |
| 02:14.13 | anthony[1] | What is externalip in file gtalk.com? I know what NAT stands for I just do not know what it actually means. Any clues appreciated. |
| 02:14.32 | anthony[1] | gtalk.conf not gtalk.com |
| 02:14.43 | enki | is your asterisk box NAT'd or does it have a public IP assigned to one of the interfaces? |
| 02:15.36 | anthony[1] | When using #su asterisk, the console returns error mssg: "This account is currently not available." |
| 02:16.42 | enki | it probably has /bin/false, or something similar, for a shell and is being denied as a result |
| 02:16.54 | enki | you can 'chown asterisk.asterisk <filename>' |
| 02:18.53 | enki | you could also run: sudo -u asterisk (command) |
| 02:21.26 | qmr | What would prevent video call from working? |
| 02:21.38 | anthony[1] | Asterisk 1.8 is running on CentOS 5 in VirtualBox |
| 02:21.43 | carrar | cat urine on the router |
| 02:22.49 | qmr | I had a video call working when I called locally to a linphone.org account.. but I can't get local users video conferencing |
| 02:24.57 | enki | qmr, possibly a codec mismatch. does audio work? |
| 02:27.15 | qmr | enki: yes, audio seems to be working well, and working nice enough over VPN too. my guess was codec mismatch, I know it has to be same on both ends .. but this is from same hw config and sw config to same .. |
| 02:27.26 | qmr | here is verbose output from my laptop trying to call my phone - http://privatepaste.com/a1f2c2435a |
| 02:28.20 | qmr | I called, tapped 'video' button frequently on my phone, then hung up |
| 02:31.27 | *** part/#asterisk mjordan (~mjordan@user-24-214-136-35.knology.net) |
| 02:32.05 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 02:33.57 | qmr | just tried disabling all codecs in both clients save VP8, still no vide |
| 03:22.10 | *** join/#asterisk deo (~dnepangue@203.177.214.75) |
| 03:33.12 | *** join/#asterisk Hyper_Eye (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 04:13.48 | enki | qmr, did you disallow all but that in asterisk for both (all) peers too? |
| 04:14.13 | enki | i've had issues with h263 and h264 when I tried allowing both, even though both endpoints should have negotiated to the same one. |
| 04:56.36 | *** join/#asterisk Korolev (~Korolev@c-50-140-71-78.hsd1.fl.comcast.net) |
| 05:19.34 | *** join/#asterisk bordr415 (~bordr415@23-24-156-49-static.hfc.comcastbusiness.net) |
| 05:20.25 | bordr415 | Having a problem with mixmonitor recording: |
| 05:20.28 | bordr415 | [2012-09-28 22:46:51] WARNING[14156]: file.c:1235 ast_writefile: No such format 'ð۵' |
| 05:20.28 | bordr415 | [2012-09-28 22:46:51] ERROR[14156]: app_mixmonitor.c:511 mixmonitor_save_prep: Cannot open /var/spool/asterisk/monitor/2012/09/28/exten-210-210-20120928-224651-1348894011.1397.ð۵ |
| 05:20.40 | bordr415 | any suggestions? |
| 05:21.52 | Korolev | si si |
| 05:22.19 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 05:27.05 | *** join/#asterisk deo (~dnepangue@58.71.19.178) |
| 05:29.08 | *** part/#asterisk deo (~dnepangue@58.71.19.178) |
| 06:00.30 | *** join/#asterisk deo_ (~dnepangue@203.177.214.75) |
| 06:00.50 | *** join/#asterisk dimer (~de@2001:470:1f0b:ba6:a8:25e1:aa60:cb75) |
| 06:00.51 | *** part/#asterisk deo_ (~dnepangue@203.177.214.75) |
| 06:17.56 | *** join/#asterisk deo_ (~dnepangue@58.71.19.178) |
| 06:28.22 | *** join/#asterisk deo___ (~dnepangue@222.127.13.226) |
| 06:30.47 | *** join/#asterisk evilman_home (kvirc@2.92.147.223) |
| 06:39.23 | *** join/#asterisk deo___ (~dnepangue@58.71.19.180) |
| 07:15.22 | *** join/#asterisk deo_ (~dnepangue@58.71.19.180) |
| 07:25.48 | *** join/#asterisk deo_ (~dnepangue@222.127.13.226) |
| 07:25.57 | *** part/#asterisk deo_ (~dnepangue@222.127.13.226) |
| 07:33.46 | *** join/#asterisk deo_ (~dnepangue@222.127.13.226) |
| 07:33.58 | *** join/#asterisk gg608f (~Adium@c-67-180-129-182.hsd1.ca.comcast.net) |
| 08:01.22 | ChannelZ | bordr415: what crazy characters are you typing? |
| 08:01.32 | bordr415 | i'm not |
| 08:01.37 | bordr415 | its set as wav |
| 08:02.16 | ChannelZ | show your dialplan for MixMonitor - is it a variable you're building elsewhere that's getting trash in it? |
| 08:03.38 | bordr415 | let me pull it up, hang on |
| 08:04.04 | bordr415 | also, the extension characters change on almost every call... (just to make it more interesting) |
| 08:04.38 | ChannelZ | It's probably confused by your period in the filename |
| 08:04.59 | ChannelZ | the one before those last 4 digits |
| 08:05.22 | bordr415 | still looking, i didn't build the system, was placed by the guy before me |
| 08:05.56 | bordr415 | shouldn't it be in /etc/asterisk ? |
| 08:07.51 | ChannelZ | well /etc/asterisk/extensions.conf |
| 08:08.07 | bordr415 | ya, there are 20+ extensions files |
| 08:08.16 | ChannelZ | unless you're using realtime in which case it's off in a database somewhere |
| 08:08.30 | ChannelZ | ah.. well they're #included in each other then. extensions.conf is the starting point |
| 08:08.52 | bordr415 | looking throught them, never thought about the . as a problem but you're prob right |
| 08:10.06 | ChannelZ | It's probably a bit of a bug in the sense it should be coughing up junk like that.. I imagine it's like a buffer overrun, that junk is some other piece of data belonging to something else or uninitialized memory or something |
| 08:10.15 | ChannelZ | s/should/shouldn't/ |
| 08:11.28 | wdoekes | who generates that filename? |
| 08:12.09 | bordr415 | found it extensions_add_19.conf |
| 08:12.16 | ChannelZ | Whoever wrote the dialplain :) |
| 08:12.19 | bordr415 | it is rather long... |
| 08:12.22 | ChannelZ | plan |
| 08:12.39 | bordr415 | 4401 lines |
| 08:12.42 | ChannelZ | It's obviously a bunch of date/timestamps and some other numbers |
| 08:12.45 | wdoekes | dialpain? |
| 08:12.55 | wdoekes | *bad*um*ts* |
| 08:13.09 | ChannelZ | sounds like |
| 08:16.34 | bordr415 | how do you think i should go about debugging it? lots of lines in here, hate to have to go through them 1 by 1 |
| 08:17.46 | ChannelZ | well the console should tell you the context, extension and priority |
| 08:18.06 | ChannelZ | ...or search for MixMonitor I should imagine... |
| 08:18.33 | bordr415 | lol, i grep'd the directory for it and ran out of shell scroll back |
| 08:18.49 | ChannelZ | There's that many instances of MixMonitor?? |
| 08:19.17 | bordr415 | yes... they used a variable to turn it on and off. i am assuming it didn't work prior and that is why it was off... |
| 08:19.46 | ChannelZ | I can only imagine what this dialplan looks like |
| 08:19.59 | bordr415 | to be frank, they had a full time "phone guy" and a full time IT guy, i'm the IT guy now doing phones as well... |
| 08:20.34 | bordr415 | have had some experience in the past. but not a master by any means |
| 08:20.45 | *** join/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 08:21.44 | ChannelZ | well like I said you should be able to find a particular instance of it from the console output |
| 08:21.51 | ChannelZ | On the console, type "core set verbose 3" |
| 08:22.07 | bordr415 | currently set to 999 |
| 08:22.08 | ChannelZ | Then call and do whatever you do to trip that mixmonitor happening |
| 08:22.33 | bordr415 | <PROTECTED> |
| 08:22.34 | bordr415 | [2012-09-29 02:15:02] WARNING[17967]: file.c:1235 ast_writefile: No such format 'ðß |
| 08:22.34 | bordr415 | [2012-09-29 02:15:02] ERROR[17967]: app_mixmonitor.c:511 mixmonitor_save_prep: Cannot open /var/spool/asterisk/monitor/2012/09/29/exten-210-210-20120929-021502-1348906502.6.ðß |
| 08:22.52 | ChannelZ | Then look for a line that says "Executing [XXX@context:XX] MixMonitor( .... ) |
| 08:23.07 | ChannelZ | prior to what you just pasted |
| 08:23.13 | bordr415 | Executing [s@macro-dial:3] AGI("SIP/210-00000002", "dialparties.agi") in new stack |
| 08:23.44 | ChannelZ | Oh, good. So it's in an AGI script. |
| 08:23.52 | bordr415 | the wierd part is just a few lines up: |
| 08:23.55 | bordr415 | -- Executing [record@sub-record-check:2] MixMonitor("SIP/210-00000002", "2012/09/29/exten-210-210-20120929-021502-1348906502.6.wav,,") in new stack |
| 08:24.01 | ChannelZ | yeah that's the one |
| 08:24.24 | ChannelZ | so there should be an extension called "record" in a context called "sub-record-check" |
| 08:25.43 | ChannelZ | (so search for [sub-record-check] and it should be there) |
| 08:26.22 | bordr415 | looking |
| 08:27.12 | bordr415 | found it |
| 08:34.26 | bordr415 | i'm not seeing anything that looks odd |
| 08:38.43 | *** join/#asterisk YoMomma (~YoMomma@cpe-142-129-178-68.socal.res.rr.com) |
| 08:40.17 | *** join/#asterisk deo_ (~dnepangue@203.177.214.75) |
| 08:42.53 | WIMPy | reminds about the thing with the 2nd . in the name. |
| 08:43.03 | *** part/#asterisk deo_ (~dnepangue@203.177.214.75) |
| 08:45.24 | bordr415 | i removed the . and made it a dash (confirmed on the output of the CLI, however, still no file created and still has the goofy extension. Manually removed the variable for the extension and placed in wav and -wav and .wav 1x each and still no go, did have the goofy ending on it though. |
| 08:45.46 | bordr415 | must be from the filename variable. will have to hunt that down. |
| 08:45.51 | WIMPy | hmm |
| 08:47.00 | bordr415 | not to argue, but i did change that on another box that is setup the same as this one and was able to at .wav.wav.wav.wav to the extension without issue, it appears as though asterisk doesn't care how many extensions you put on. |
| 08:47.20 | WIMPy | ok |
| 08:47.56 | bordr415 | i'm thinking it's almost 3 in the morning and the system works (without recordings, will tackle this in the morning) :) |
| 08:48.23 | bordr415 | thanks for the help gentlemen, it appears i'm on the right track now |
| 08:49.30 | WIMPy | I think I have seen such a thing happening because of some input it didn't like, bit if it's not the . then I can't remember what it was :-( |
| 08:54.12 | bordr415 | actually i found this a while ago, before i signed on... |
| 08:54.38 | bordr415 | http://ibot.rikers.org/%23asterisk/20110530.html.gz |
| 08:55.14 | bordr415 | deltaray was having a similar problem with the pipe |
| 08:58.13 | *** join/#asterisk fisted_ (~fisted@unaffiliated/fisted) |
| 09:22.37 | *** join/#asterisk deo_ (~dnepangue@122.53.72.218) |
| 09:41.40 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 10:32.32 | *** join/#asterisk felimwhiteley (~quassel@089-101-203026.ntlworld.ie) |
| 11:11.59 | *** join/#asterisk mathi (~Matthew@109.133.227.234) |
| 11:12.01 | mathi | hi |
| 11:12.19 | mathi | how can I check if a variable isn't empty in my dialplan? |
| 11:14.54 | wdoekes | mathi: ${LEN(${var})} or $["${var}"!=""] |
| 11:15.31 | mathi | ah thanks! |
| 11:22.52 | *** join/#asterisk BorjaGVO (532b11fd@gateway/web/freenode/ip.83.43.17.253) |
| 11:26.55 | BorjaGVO | Hi everyone, can anyone help me with something? It's a pretty basic thing but still wonder how to do it and how it works. In the dialplan, if a have a context ([a-context]) and inside it an include with the context [a-context-custom]...how can I change the [a-context] , just touching the [a-context-custom]? |
| 11:29.24 | WIMPy | Asare you asking how to change somethign, |
| 11:29.31 | WIMPy | oops |
| 11:29.36 | WIMPy | Are you asking how to change something, without changing it? |
| 11:30.07 | *** join/#asterisk _zoom_ (~M.Eissa@196.1.219.122) |
| 11:30.11 | _zoom_ | hi, |
| 11:30.46 | _zoom_ | for those who is doing gsm termination, how to hide my lines numbers? |
| 11:33.09 | BorjaGVO | WIMPy: Well, I guess yeah, you are not touching anything in the context EXCEPT the context you've included...I say this because that's how FreePBX sets your dialplan, with every singe context with a "same-context-custom" and it's driving me nuts. I might be thinking about it the wrong way...I don't know |
| 11:37.02 | WIMPy | BorjaGVO: Well, you cann only add to a context via an include, but you cannot override anything. |
| 11:37.31 | WIMPy | _zoom_: What kind of termination and what line numbers? |
| 11:38.45 | BorjaGVO | I see...in that case I'll give it a new try and rethink about how to change things..Thanks =) |
| 11:40.31 | _zoom_ | <PROTECTED> |
| 11:41.05 | WIMPy | Not? |
| 12:35.36 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 13:09.55 | *** join/#asterisk Bryanstein (~Bryanstei@shellium/admin/bryanstein) |
| 13:20.29 | *** join/#asterisk aross42 (~aross@216.16.232.86) |
| 13:26.02 | *** join/#asterisk l2trace99 (~l2trace@70-119-141-5.res.bhn.net) |
| 13:28.46 | *** join/#asterisk wonderworld (~ww@dsdf-4db54c4e.pool.mediaWays.net) |
| 13:29.24 | *** join/#asterisk aross42 (~aross@216.16.232.86) |
| 13:31.23 | *** join/#asterisk bipul (~vipul@unaffiliated/bipul/x-4918593) |
| 13:32.13 | *** join/#asterisk aross42 (~aross@216.16.232.86) |
| 13:44.12 | *** join/#asterisk SeRi (~wtf@c-98-200-53-71.hsd1.tx.comcast.net) |
| 13:51.34 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 13:53.47 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 13:58.03 | wonderworld | hi, i want only specific sip user agents to be able to connect to a server. can this be configured wihout hacking the source code? |
| 13:58.54 | [TK]D-Fender | no |
| 13:59.37 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 14:01.00 | wonderworld | [TK]D-Fender: thanks |
| 14:19.10 | *** join/#asterisk vinhdizzo (~vinh@vqn-portege.ics.uci.edu) |
| 14:23.22 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 14:24.27 | *** join/#asterisk felipealmeida (~user@mvx-187-16-79-187.mundivox.com) |
| 15:06.15 | *** join/#asterisk vinhdizzo (~vinh@vqn-portege.ics.uci.edu) |
| 15:13.37 | *** join/#asterisk YoMomma (~YoMomma@cpe-142-129-178-68.socal.res.rr.com) |
| 15:38.31 | *** join/#asterisk autofsckk (~que@unaffiliated/autofsckk) |
| 15:45.11 | autofsckk | p3nguin: hello, are you there? do you still make PKGBUILD for arch? |
| 15:46.19 | *** join/#asterisk vassilux (~vassilux@31.35.29.247) |
| 15:48.40 | *** join/#asterisk j4m3s_ (~j4m3s_@adsl-98-77-104-115.mia.bellsouth.net) |
| 15:56.26 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 15:58.07 | *** join/#asterisk cusco (~tralala@a79-168-182-209.cpe.netcabo.pt) |
| 15:58.10 | cusco | hello |
| 15:58.54 | cusco | if I am to generate a event on AMI, is there a dummy command/app I can call? |
| 16:01.10 | *** join/#asterisk hackeron (~hackeron@gentoo/user/hackeron) |
| 16:06.44 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 16:06.44 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 16:09.49 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 16:18.45 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 16:47.16 | *** join/#asterisk sideffect (sideffect@gateway/shell/bshellz.net/x-zgecfssyrglzgohx) |
| 17:11.17 | *** join/#asterisk odenkos (~odenkos@unaffiliated/odenkos) |
| 17:14.29 | *** join/#asterisk nisam (nisam@115.241.192.103) |
| 17:17.14 | mathi | hi |
| 17:17.22 | mathi | my dialplan stops here: |
| 17:17.27 | mathi | same => n,Record(/records/files/${UNIQUEID}-name:wav) |
| 17:17.38 | mathi | I can't record anything, it just hangs up |
| 17:18.14 | *** join/#asterisk mjordan (~mjordan@user-24-214-136-35.knology.net) |
| 17:18.15 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 17:18.31 | mathi | any ideas? |
| 17:21.08 | nisam | hi |
| 17:21.37 | nisam | which is the best opensource sip phone supporting g729..? |
| 17:21.47 | *** join/#asterisk Matthias (~Matthias@2001:15c0:670f:ffff::2) |
| 17:27.43 | *** join/#asterisk evilman_home (kvirc@128-73-24-44.broadband.corbina.ru) |
| 17:28.18 | *** join/#asterisk YoMomma (~YoMomma@cpe-142-129-178-68.socal.res.rr.com) |
| 17:32.34 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 17:32.34 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 17:33.34 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 17:33.34 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 17:35.04 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 17:35.04 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 17:38.31 | *** join/#asterisk l2trace99 (~l2trace@70-119-141-5.res.bhn.net) |
| 17:47.04 | *** join/#asterisk rbowles (~bb@75-147-63-201-NewEngland.hfc.comcastbusiness.net) |
| 17:56.43 | *** join/#asterisk fergus (~fergus@178.124.149.113) |
| 18:20.22 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 18:22.47 | *** join/#asterisk aross42 (~aross@216.16.232.86) |
| 18:27.10 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 18:51.41 | *** join/#asterisk TimeRider (~steve@027a41dd.bb.sky.com) |
| 19:16.52 | *** join/#asterisk bradafoster (~brad@pool-173-74-203-181.dllstx.fios.verizon.net) |
| 19:17.37 | *** join/#asterisk h0m3r_ (h0m3r_@187.64.25.199) |
| 19:19.38 | *** join/#asterisk mjordan (~mjordan@user-24-214-136-35.knology.net) |
| 19:19.39 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 19:32.06 | *** join/#asterisk Galen (~Galen@rrcs-24-43-17-235.west.biz.rr.com) |
| 19:32.10 | *** part/#asterisk bradafoster (~brad@pool-173-74-203-181.dllstx.fios.verizon.net) |
| 19:33.50 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 20:02.25 | *** join/#asterisk Luke-Jr (~luke-jr@2001:470:5:265:222:4dff:fe50:4c49) |
| 20:03.37 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 20:21.37 | *** join/#asterisk wudles (~wudles@gateway.secureinstrument.com) |
| 20:24.43 | *** join/#asterisk YoMomma (~YoMomma@cpe-142-129-178-68.socal.res.rr.com) |
| 20:31.01 | *** join/#asterisk chasingSol (~ao@41.232.127.52) |
| 20:48.23 | *** join/#asterisk ChannelZ (bobm@burner.com) |
| 20:51.09 | *** join/#asterisk YoMomma (~YoMomma@cpe-142-129-178-68.socal.res.rr.com) |
| 21:03.43 | *** join/#asterisk j4m3s_ (~j4m3s_@adsl-98-77-104-115.mia.bellsouth.net) |
| 21:09.05 | *** join/#asterisk brookshire (mbrooks@hijacked.us) |
| 21:12.15 | *** join/#asterisk tomaw (tom@freenode/staff/tomaw) |
| 21:15.06 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 21:19.18 | *** join/#asterisk Greenlight (ircN@client-86-31-251-6.oxfd.adsl.virginmedia.com) |
| 21:26.00 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:32.16 | *** join/#asterisk j4m3s__ (~j4m3s_@adsl-98-77-39-24.mia.bellsouth.net) |
| 21:33.13 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 21:33.13 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 21:39.29 | *** join/#asterisk n3hxs (~n3hxs-5@TOROON63-3096776761.sdsl.bell.ca) |
| 21:43.43 | *** join/#asterisk vinhdizzo (~vinh@vqn-portege.ics.uci.edu) |
| 21:45.37 | *** join/#asterisk evilman_home (kvirc@2.92.147.50) |
| 21:54.21 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 21:57.17 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 22:25.24 | *** join/#asterisk volga629 (~volga629@76-10-130-18.dsl.teksavvy.com) |
| 22:26.25 | volga629 | Hello Everyone, [2012-09-29 16:19:54] WARNING[10494] app_voicemail.c: Unable to read password [2012-09-29 16:19:54] ERROR[10494] app_voicemail.c: No vmstate found for user:, mailbox 101. Asterisk use IMAP storage form vm. |
| 22:26.42 | volga629 | for |
| 22:27.33 | volga629 | it only one user affected |
| 22:35.09 | *** join/#asterisk fergus_ (~fergus@178.124.149.113) |
| 22:36.38 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 22:37.07 | *** join/#asterisk Praise- (~Fat@unaffiliated/praise) |
| 22:56.55 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 22:58.50 | *** join/#asterisk YoMomma (~YoMomma@cpe-142-129-178-68.socal.res.rr.com) |
| 23:25.14 | *** join/#asterisk YoMomma (~YoMomma@cpe-142-129-178-68.socal.res.rr.com) |
| 23:27.52 | *** join/#asterisk Maliuta (nikolai@donetsk.lusan.id.au) |
| 23:56.45 | volga629 | it look like some DTMF issue -- Incorrect password '' for user and I checked all settings which wasn't change for 2 weeks at least |