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| 01:37.32 | gnudna | hey guys how can i see who is making similar attempts |
| 01:37.55 | gnudna | Call from '' to extension '3011972597396094' rejected because extension not found in context 'main' |
| 01:38.25 | gnudna | im in what seems a never ending battle to secure asterisk |
| 01:39.10 | gnudna | how can someone force asterisk to call out from the outside? if they are unable to register a devce? |
| 01:39.38 | gnudna | also i see from time to time --> app.c: Huh....? no dial for indications? |
| 01:43.20 | WIMPy | There's no need to register in order to place a call. It's only needed to receive calls. |
| 01:45.22 | gnudna | ok |
| 01:45.48 | gnudna | how does one secure an asterisk server from outside people making calls then? |
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| 01:45.57 | gnudna | i have added te deny allow options |
| 01:46.25 | WIMPy | By not letting them do anything you don;t want in the context they will end up in. |
| 01:47.14 | gnudna | ok |
| 01:47.22 | gnudna | and my other question? |
| 01:47.31 | gnudna | <PROTECTED> |
| 01:47.48 | WIMPy | has never senn that. |
| 01:47.55 | gnudna | ok |
| 01:48.29 | gnudna | seems some calls get lost when put on hold |
| 01:48.43 | gnudna | or they get put back into the queue |
| 01:49.09 | gnudna | kinda random from what im told and consistent so hard to debug at the moment |
| 01:50.07 | gnudna | sorry meant inconsistent |
| 01:51.34 | gnudna | would love someone to talk to...and look over the configs |
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| 02:04.46 | gnudna | thank you for the pointers wimpy |
| 02:04.59 | gnudna | goodnight |
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| 02:59.41 | andross | does anyone here attend the mhvlug |
| 02:59.50 | andross | serious question |
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| 03:18.37 | iprouteth0 | urgh.... Still fighting with SIP-TLS and SRTP |
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| 03:45.55 | igcewieling | iprouteth0: like everyone else 8-| |
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| 03:53.39 | iprouteth0 | igcewieling: I feel like I'm close, but can't get any further progress |
| 03:54.06 | iprouteth0 | Seems like I can get SRTP to work on a peer that isn't using sip-tls and I can get sip-tls to work but not with SRTP |
| 03:54.23 | igcewieling | have you searched the asterisk mailing lists? I don't recall seeing much disucssed there, but it can't hurt |
| 03:54.42 | iprouteth0 | been through a good handful of threads but I'm sure I've missed some |
| 03:54.51 | iprouteth0 | I haven |
| 03:55.18 | iprouteth0 | I haven't been using the dial plan directives however. I tried once but asterisk did not seem to like those directives |
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| 03:55.50 | iprouteth0 | though it's possible I was not using them correctly. The documentation for sip-tls and srtp is pretty sparse |
| 03:56.17 | iprouteth0 | not really certain what the best way to go about troubleshooting is either |
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| 04:07.01 | igcewieling | the best way is wait a year until this new feature settles down 8-) |
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| 04:14.25 | iprouteth0 | I'm beginning to think you're right igcewieling |
| 04:14.39 | iprouteth0 | I'm running it on an openwrt router and usually run it over openvpn |
| 04:15.02 | iprouteth0 | but i'm on trunk and the package repository updated with a new kernel so I need to relash the kernel on the device to install openwrt |
| 04:15.07 | iprouteth0 | that can be a pain sometimes |
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| 04:19.26 | igcewieling | or you can assume your calls are not that interesting and it is unlikely someone wants to listen to them. did you also add armored conduit to your landlines and add armed guards to your wiring closet? |
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| 04:28.20 | iprouteth0 | lol. My router is in a central office so it does have adequate physical security :) |
| 04:28.47 | igcewieling | secure VoIP would be nice and should happen eventually, but there is no urgency to it. |
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| 05:38.48 | iprouteth0 | drivin me nuts.... Wish I could get it working.... Feels like I |
| 05:38.52 | iprouteth0 | I'm so close! |
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| 07:21.29 | elmargol | How much memory do you think does debian stable + asterisk need for receiving fax and 1-2 calls? |
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| 07:31.51 | izbushka | hi |
| 07:31.54 | phix | hai |
| 07:32.07 | andross | elmargol: you mean in terms of hardware? |
| 07:32.17 | phix | elmargol: just put 16Gb in your server :) |
| 07:32.23 | elmargol | I have it running in a VM |
| 07:32.24 | phix | ram is cheap |
| 07:32.26 | andross | elmargol: you need a raspberry pi |
| 07:33.00 | andross | also you dont wanna do faxing unless you are prepared for it not to work most of the time |
| 07:33.17 | elmargol | andross, i have fax running and it seems to work |
| 07:33.20 | andross | buy a vps using your spare bitcoin from bitvps |
| 07:33.27 | izbushka | is res_fax.so enough for receiving faxes or I should get res_fax_digium or something? |
| 07:33.44 | elmargol | I just need to receive fax. no need to send |
| 07:35.21 | James87 | you dont need much memory for that 512 mb will do |
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| 07:50.30 | bulkorok | izbushka: you should use res_fax_spandsp |
| 07:51.21 | izbushka | thank you, i suspected that |
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| 08:14.38 | fling | May you please help me setting up ata? |
| 08:14.58 | fling | 303/303 10.0.2.23 D 5060 OK (11 ms) |
| 08:15.37 | fling | It is AddPac AP100, it is up but I can't call, got busy, nothing in console |
| 08:17.05 | kaldemar | enable verbosity and sip debug and pastebin the result of a whole call. |
| 08:17.53 | fling | kaldemar: also I see 'Local Domain name (SIP userpart of authentication)' field in 'SIP (Session Initiation Protocol)' ata's settings page |
| 08:17.59 | fling | and the field is empty |
| 08:27.53 | fling | kaldemar: changed analog phone and it started working automagically |
| 08:27.55 | fling | kaldemar: thanks ;D |
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| 08:54.16 | Rico29 | hi all |
| 08:54.51 | Rico29 | I still have a problem where agents in Queue stop ringing when queue is full, and I have to restart asterisk to restore normal activity |
| 08:54.56 | Rico29 | asterisk 1.8.20.1 |
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| 08:55.41 | *** join/#asterisk jkroon (~jkroon@dsl-244-28-165.telkomadsl.co.za) |
| 08:56.06 | jkroon | hi guys, how do I go about trouble-shooting an ever increasing number of open fd's and failure to clear hung up channels from asterisk? |
| 08:56.44 | iprouteth0 | fling: that empty field is also known as "authuser" |
| 08:57.00 | iprouteth0 | most devices will use what you put in the username field if it is left blank |
| 09:03.47 | fling | iprouteth0: ok, thanks |
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| 09:25.59 | wdoekes | jkroon: enable ref count debugging |
| 09:26.02 | jkroon | ok guys, i've got asterisk 11.3.0 leaking file descriptors like a sift. |
| 09:26.57 | jkroon | wdoekes, DEBUG_FD_LEAKS ? |
| 09:27.18 | jkroon | i'm reasonably sure it's not the FDs themselves but some other lock issue, so will DEBUG_THREADS get me core show locks too? |
| 09:27.34 | wdoekes | debug_threads will get you 'core show locks' |
| 09:28.07 | jkroon | i've now done everything I can to debug my own custom code, and it just stops getting called, my suspicion is that on channel hangup something happens that prevents the CDR system from sending the call to the cdr handlers. |
| 09:28.28 | wdoekes | debug_fd_leaks sounds useful |
| 09:28.30 | jkroon | and that there is some lock somewhere there that then holds open all the various channels. not sure if batch mode will do anything to that. |
| 09:28.33 | jkroon | enabled. |
| 09:30.22 | wdoekes | I was talking about REF_DEBUG |
| 09:30.37 | wdoekes | if you enable that, it will write to /tmp/refs |
| 09:30.46 | wdoekes | and you can read that using the refcounter utility |
| 09:31.00 | jkroon | ok, will enable that too ... |
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| 09:31.56 | wdoekes | (See "To find out why objects are not destroyed" in astobj2.h) |
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| 09:36.25 | jkroon | wdoekes, 'REF_DEBUG' not found |
| 09:36.37 | wdoekes | jkroon: 11:31 < wdoekes> (See "To find out why objects are not destroyed" in astobj2.h) |
| 09:38.29 | jkroon | grr, wdoekes I see ... ok, that's no use as I don't even know *what* objects it are that's not getting destroyed. |
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| 09:43.31 | wdoekes | jkroon: you can enable it for all objects |
| 09:43.47 | wdoekes | just put REF_DEBUG on top of the .h file |
| 09:44.52 | wdoekes | and you don't *need* the custom descriptive tags.. it will still record the function/lineno/file |
| 09:46.17 | jkroon | perhaps not a bad idea to make that a config option in menuselect ... ? |
| 09:51.36 | jkroon | ok, given that core show locks shows an insane number of locks, how do I go about figuring out who *has* the lock (where was it locked from)? |
| 09:53.59 | jkroon | ok, the output from asterisk -rx "core show locks" is 5671 lines long and can be downloaded from http://www.uls.co.za/asterisk-locks.txt |
| 09:54.04 | jkroon | any help appreciated. |
| 09:55.03 | jkroon | cdr_mysql?! |
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| 10:00.31 | jkroon | ok, cdr_mysql it probably is, now I need to get debug out, since it's killing the swtich., |
| 10:01.14 | James87 | can anyone tell me what dundi is used for? |
| 10:01.34 | jkroon | for finding routes to destinations. |
| 10:01.38 | jkroon | very, very handy :) |
| 10:02.01 | jkroon | so if you have multiple servers to which an agent can register DUNDi can be used to locate them. |
| 10:02.08 | jkroon | as but one example |
| 10:03.00 | James87 | hmmm sounds interesting but is it only used when you have multiple servers? |
| 10:05.45 | *** join/#asterisk Greenlight (~email@cpc1-dund9-0-0-cust142.16-4.cable.virginmedia.com) |
| 10:06.08 | jkroon | or if you're peering with someone else who's willing to route certain routes for you :) |
| 10:06.24 | jkroon | but yea, mostly just in multi-server setups in my example. |
| 10:06.30 | James87 | ok sounds clear, thanks |
| 10:07.25 | jkroon | the original idea that was posed was that you can peer with a bunch of people who's willing to advertise what "free" routes they offer and will allow anyone to route, mostly aimed at an american market from that perspective, so a whole network of people willing to route calls for other people (free of charge) in return for them routinng calls for you again |
| 10:07.33 | jkroon | that's the theory, will never work in ZA. |
| 10:07.33 | *** join/#asterisk DennisG_NL (~DennisG_N@82-170-131-186.ip.telfort.nl) |
| 10:09.33 | v0lZy | hey |
| 10:11.04 | v0lZy | jkroon: it routes as in it passes through them? |
| 10:11.06 | phix | hai |
| 10:12.05 | v0lZy | jkroon: In the sense that voice packets go through the whole chain to the destination |
| 10:12.20 | jkroon | as in for example, let's say I sit at free.uls.co.za using protocol SIP, then I can via dundi publish that I'm willing to route calls for 27873513298 free of charge, and your contact for the call would be SIP/arbuser:arbpass@free.uls.co.za/0873513298 |
| 10:12.51 | jkroon | so now on DUNDi a query goes out for 0873513298 and I respond with that response, and you initiate a Dial() on the returned result. |
| 10:13.52 | v0lZy | so... I'd call you and you'd call the number? |
| 10:14.16 | v0lZy | and all the packets would travel from me to you to the destination |
| 10:14.19 | v0lZy | and back again? |
| 10:14.50 | v0lZy | What if that number isnt registered to you, can it still be advertised? |
| 10:16.19 | jkroon | yes, except in my hypothetical case that number belongs to me, so giving a free inbound call is not a major issue as I only pay for bandwidth. |
| 10:16.45 | jkroon | yes, and that's the *risk*, I can "steal" calls. |
| 10:16.51 | jkroon | so it's a trust-based mechanism. |
| 10:17.54 | jkroon | not only that ... i'd be able to listen in on your calls using Monitor() etc ... so you REALLY have to trust those that you peer with. |
| 10:18.18 | jkroon | it's also convenient as a fail-over mechanism, again, mostly in my scenario where i've got multiple servers. |
| 10:19.19 | v0lZy | so its concievable that its chained |
| 10:19.40 | phix | jkroon: using rtps or what ever it's called would solve the listening in business? |
| 10:19.46 | v0lZy | I want A, you advertise A, but route it through someone else who routes it through someone else etc? |
| 10:19.59 | v0lZy | phix: I dont think it would. |
| 10:20.05 | Greenlight | I would presume the final endpoint would still be able to listen in |
| 10:20.08 | *** join/#asterisk DennisG (~DennisG@82-170-131-186.ip.telfort.nl) |
| 10:20.22 | v0lZy | phix: he could accept the call locally on the fly and forward it while recording it. |
| 10:20.36 | phix | oh, no man in the middle protection? |
| 10:20.37 | jkroon | phix, no, because I'd need to make you go to sips://@server.of.my.choice/??? where I can still decrypt the rtp (even though it's rtps) |
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| 10:22.01 | v0lZy | you could just answer the call as a local extension, and then send it through a separate dundi lookup or something as well, couldnt u? |
| 10:22.14 | DennisG | it's normal that you can decrypt the data on the endpoint of the transfer ;) |
| 10:22.33 | v0lZy | i think the issue here is that whoever is doing the lookup doesnt know what the endpoint is |
| 10:23.15 | v0lZy | so you can declare yourself as the endpoint, initiate a separate call to the true destination, then bridge those two calls to a conference in which you could listen in... |
| 10:23.53 | v0lZy | nasty. |
| 10:24.20 | phix | sounds dicey |
| 10:25.26 | v0lZy | very. |
| 10:25.31 | jkroon | v0lZy, yes, i chain a lot, for example, client A registers (and passes calls) to server Z, server Z has an IAX/2 link with server Y, and client B registers to Y. No A places a SIP call to A, Z uses DUNDi to figure out that B is registered to Y, then (over IAX/2) passes the call to Y which then finally passes the call (over SIP) to B. |
| 10:25.36 | v0lZy | think about all the telephone based banking thats done these days. |
| 10:25.53 | jkroon | v0lZy, or just execute Dial() on the raw unanswered extension and do the right thing. |
| 10:26.34 | jkroon | v0lZy, depends on the setup - I know for a *fact* that everybody that *can* answer my DUNDi queries is *authorized* and *authoritative* and more importantly - UNDER MY CONTROL. |
| 10:26.58 | v0lZy | No wonder DUNDi didnt catch on. |
| 10:27.36 | v0lZy | not in the independant network sense anyway. |
| 10:28.22 | v0lZy | Think how much potential for abuse there is in this telephone banking stuff |
| 10:28.57 | *** join/#asterisk Bradada (~Bradada@220-135-49-159.HINET-IP.hinet.net) |
| 10:28.58 | v0lZy | i.e., suppose you are a bastard administrator from hell and that one of your users is using your services to contact a bank |
| 10:29.22 | v0lZy | you can either listen in on their exchange, or, pretend that you're the bank |
| 10:29.43 | v0lZy | I suppose there's some two way authentication through a third party done in such scenarios though |
| 10:29.48 | jkroon | you can do that if you admin the user anyway. |
| 10:30.05 | v0lZy | I sincerely hope the banks don't depend on the secure phone lines... |
| 10:30.11 | Greenlight | You also at present give such trust to your existing ITSP |
| 10:30.39 | v0lZy | Greenlight: yeah, i know, but think about how many sip providers are out there |
| 10:30.48 | v0lZy | a bajillion |
| 10:31.15 | v0lZy | makes you wonder. |
| 10:31.24 | jkroon | ok, thank you very much cdr_mysql ... you might just have lost me my biggest client :( |
| 10:31.26 | jkroon | *sigh* |
| 10:31.41 | v0lZy | All you'd need to do is build up a registry of the numbers of the telephone banking services... |
| 10:31.58 | Greenlight | What did cdr_mysql do ? |
| 10:34.39 | jkroon | well, it takes a lock on it's lock_mysql, and some path doesn't release, that causes a bunch of calls from the cdr core to lock up in cdr_mysql, and since this eventually puts back-pressure on the rest of asterisk consumes all available resources, and causes major issues. |
| 10:34.54 | v0lZy | gonna go pick up my monitor brb |
| 10:35.02 | Greenlight | Ouch ;/ |
| 10:35.40 | jkroon | I don't think batch mode will resolve the issue on the core because I *suspect* that even then the generated cdr record maintains a ast_channel* reference to the channels, which means that the associated channels will hang around until the cdr's successfully process one day. |
| 10:35.43 | Greenlight | Would cdr_adaptive_odbc in batch mode allevaiate that? |
| 10:36.05 | Greenlight | Ahh |
| 10:36.47 | Greenlight | Yea; we run in batch mode, and it can cope with back end db issues, but I guess if they never get processed things are gonna go haywire eventually espeically if it's holding a ref |
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| 10:44.31 | James87 | good explanation jkroon, you should blog it somewhere :) |
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| 11:21.35 | jkroon | Greenlight, yea, haven't looked at main/cdr.c (IIRC) yet, or the ast_cdr struct for that matter yet. But this was nasty. And I suspect I'm down one client. A big one. |
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| 11:23.01 | Greenlight | Ouch - client pretty pissed then (or going to be)? |
| 11:23.46 | Greenlight | Is it the loss of CDR's that's the problem for them, or the system crash ? |
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| 11:41.25 | jkroon | pissed doesn't quite describe it. they couldn't care less about the CDRs. me losing CDRs means they don't pay for calls. The lockups (and therefor lack of service) is an issue. |
| 11:42.23 | James87 | you dont't have backups or redundant servers? |
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| 12:05.09 | s-hell | Hello everyone! I'm trying to get this running: http://zwizwa.be/-/pool/20120418-152928 |
| 12:05.39 | s-hell | But for some reason the script isn't executed. It seems that my asterisk ignores the hangup |
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| 12:13.08 | jkroon | James87, yes i do. somehow the issue bit on both of them at the same time ... go figure. |
| 12:13.23 | jkroon | and even restarting asterisk did not help, unload cdr_mysql ... issue gone. |
| 12:26.10 | jkroon | ok, here is the strange thing - I cannot find *anything* in cdr_mysql that looks like it should cause the problem. |
| 12:26.53 | jkroon | except that I can correlate the lockups approximately with reload |
| 12:30.56 | *** join/#asterisk keycruncher (~Adium@c-174-55-112-94.hsd1.pa.comcast.net) |
| 12:39.37 | Greenlight | I've seen issues on a few boxes with massive dialplans (they were running FreePBX) that would lockup about 30% of the time when busy and doing a reload, apparently it's a known issue. When it happened I would have to kill the asterisk process, although I didn't attempt to unload cdr_mysql |
| 12:49.52 | jkroon | Greenlight, my dialplan is getting rather large ... approaching 25k lines of dialplan. |
| 12:50.09 | jkroon | after filtering comments etc it's only about 14k lines though. |
| 12:50.37 | Greenlight | I'm not sure of the exact cause, but if reloading is causing lockups then I'd guess you're perhaps being effected |
| 12:50.51 | Greenlight | Apparently, turing off console verbosity helps |
| 12:54.45 | Greenlight | For us, chanting "please don't crash, please don't crash" when issuing a reload also seemed to help. YMMV :) |
| 12:55.31 | *** join/#asterisk serafie1 (~erin@nat/digium/x-wwkfzefwsakgohcw) |
| 12:55.32 | jkroon | Greenlight, ... hmm, i'll add code to reduce verbosity to 0 before issuing reload |
| 12:55.38 | jkroon | rofl |
| 12:56.01 | jkroon | every time i do it by hand (except when chan_sip has DNS issues) it works. |
| 12:58.09 | Greenlight | If it is the same issue, they yes it's a very odd one. A "core show channels" during the lockup would either hang completely or spit out at a rate of about a channel a second. |
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| 13:15.40 | jkroon | Greenlight, no, i still get core show channels. |
| 13:15.53 | jkroon | but unloading cdr_mysql merely alleviated the issue. we still have some of it. |
| 13:17.15 | jkroon | i wonder whether it's simply "sheer load" |
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| 13:36.09 | jkroon | Greenlight, at what call concurrency did your issues start cropping up? |
| 13:40.06 | Greenlight | Hmm.. it's difficult to say. I've seen it happen on boxes with 30 simultaneous calls, but perhaps 1000 extensions. |
| 13:40.18 | Greenlight | Certainly seems to happen more frequenlty with more calls in progress |
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| 13:56.09 | Katty | cries. |
| 13:57.36 | Katty | monday morning meetings suck :< |
| 13:57.53 | igcewieling | I'm too lazy to do the research, but I'm pretty sure I saw something about a bug related to "sip show channels" and a phone registration happening at the same time causing some kind of deadlock |
| 13:58.20 | igcewieling | could be core show channels, not sure. |
| 13:58.36 | *** join/#asterisk RetentiveBoy (~paul@74-95-28-33-Atlanta.hfc.comcastbusiness.net) |
| 13:59.10 | igcewieling | jkroon: "dialplan show" will tell you the actual number of dialplan lines at the end. |
| 13:59.31 | igcewieling | you can tell we do a database based dialplan. LOL! 10,000 TNs, "-= 45 extensions (416 priorities) in 16 contexts. =-" |
| 14:03.15 | jkroon | Greenlight, yea, that's the thing - is I consistently operate with ~80-100 concurrent calls at the moment. |
| 14:03.38 | jkroon | igcewieling, 3198 extensions (7518 priorities) in 2924 contexts. |
| 14:03.53 | jkroon | and going DB has it's advantages, but what happens if the link to the DB dies? |
| 14:04.36 | igcewieling | jkroon: Bad Things 8-) Though our AGIs use a locally replicated database for read ops and a central DB for write ops. |
| 14:05.08 | igcewieling | For the most part if the central DB goes down calls will still happen, though stuff like CDR and CEL won't of course. |
| 14:05.40 | jkroon | yea, CDR and CEL I can deal without, got ways to "buffer" that. |
| 14:05.48 | igcewieling | one of my Great Annoyances with Asterisk's Realtime is it does not appear to support such a design, so we don't use Realtime, we use AGIs |
| 14:05.55 | jkroon | can I configure ODBC with a primary/backup DB? |
| 14:06.38 | igcewieling | jkroon: I think func_odbc supports that, but not Realtime |
| 14:06.43 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-nadfodbhxewdpasj) |
| 14:06.43 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:07.05 | *** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com) |
| 14:07.53 | igcewieling | The only realtime we used with sippeers, but again, you can't separate read ops from write ops |
| 14:08.15 | igcewieling | we now regenerate sip.conf from the db once per hour to avoid that. |
| 14:09.28 | file | takes notes |
| 14:09.31 | WIMPy | As long as 'sip reload' works for you... |
| 14:10.43 | WIMPy | Oh, BTW: |
| 14:10.53 | igcewieling | file: for inclusion in Asterisk 16? *tease* |
| 14:11.28 | file | I'm not touching chan_sip, but the new data access layer API for 12 (which is effectively the next generation of realtime) could do such things |
| 14:11.40 | WIMPy | file: chan_sip always sends a to: with the configured hostname. That can be some issue sometimes. |
| 14:11.41 | igcewieling | file: nifty! |
| 14:14.14 | jkroon | because I can probably reduce my dialplan to <500 priorities if I can query things from the database using whatever. issue is that there is certain queries that is better suited to dialplan style lookups than SQL ... and depending on SQL is something I'd prefer to avoid. |
| 14:14.23 | file | it's fun to code for so I pick up ideas for additions/changes where I can |
| 14:14.36 | jkroon | WIMPy, yea, sip reload is a scary operation. |
| 14:15.17 | jkroon | is a seriously disgruntled user at the moment, to be honest. |
| 14:15.31 | WIMPy | jkroon: I always lose my registrations. |
| 14:15.35 | jkroon | i really wish that some of the ideas for ast 12 was already around. |
| 14:15.37 | igcewieling | jkroon: You're using Asterisk, that is the normal state. 8-) |
| 14:15.52 | WIMPy | :-( |
| 14:15.59 | jkroon | WIMPy, fortunately for my core systems i don't register outbound, but people do register to me. |
| 14:16.05 | *** join/#asterisk Leddy (leddy@krypton.evosurge.com) |
| 14:16.22 | jkroon | igcewieling, yea well, if you told me I could spend R100'000 right now and my problems would just *go away* I might seriously consider it. |
| 14:16.26 | WIMPy | needs both |
| 14:19.32 | igcewieling | stuff like this doesn't help either "VZ tech was onsite last week physically ripped out the pairs this circuit was using. XO and our tech got there at the same time. Seen the broken pairs, repunched them down and everything is working." |
| 14:20.22 | jkroon | no, it does not help |
| 14:21.57 | *** join/#asterisk fafaflofly (~bababooey@static-173-87-78-218.sdsl01.roch.ny.frontiernet.net) |
| 14:22.35 | file | igcewieling, I shall add the ability to designate stuff as read-only or write-only |
| 14:22.40 | igcewieling | jkroon: In the Good Old Days Asterisk's bugs were fairly easy to reproduce, it either worked in a specific situation or it didn't. Digium made significant improvements to the release cycle, etc and those sorts of bugs are much less common. These days most Asterisk bugs seem to be the ones which are are hard to find and hard to reproduce. |
| 14:23.18 | igcewieling | file: you understand the reasoning behind the need and the master/slave DB setup? |
| 14:23.40 | file | yes |
| 14:24.00 | igcewieling | file: cool. |
| 14:24.51 | file | it's minor to add |
| 14:33.49 | *** join/#asterisk Invader (~Invader@unaffiliated/invader) |
| 14:33.59 | igcewieling | file: do you know if there is an easy way in Jira to see if Digium still has my disclaimer on file? I sent it in sometime in 2002 I think and have not done anything requiring a disclaimer in years. |
| 14:34.36 | igcewieling | I have a couple of dialplan examples I'd like to submit |
| 14:34.57 | file | doubt it, but it's all digital these days - including signing one |
| 14:36.14 | igcewieling | *nod* I'll resubmit when I clean up the code a little bit. |
| 14:36.23 | Greenlight | igcewieling: Was it you who gave me the srv dns dialling function ? |
| 14:36.25 | igcewieling | Several people here found it useful. |
| 14:36.38 | igcewieling | Greenlight: correct and that is what I was thinking of submitting. |
| 14:36.46 | Greenlight | agrees |
| 14:37.04 | igcewieling | Greenlight: working well for you? |
| 14:37.24 | Greenlight | I did alter a few bits so that the CDR's worked with it, but it's worked perfect for circo 1,000,000 calls per day for a few months |
| 14:37.35 | Greenlight | *circa |
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| 14:38.45 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 14:39.13 | igcewieling | ha! We only use it for PBX endpoints so not high call volume. |
| 14:39.35 | Greenlight | Well I can vouch for it working well under load :) |
| 14:40.31 | Greenlight | And my customers are a lot happier now too, as issues with my wholesale provider no longer effect them, since calls are distributed accross their nodes :) |
| 14:42.10 | igcewieling | That is great to hear. Most of our AGI and Dialplan/AEL code would not be very useful outside our company, but the stuff which would be useful we are willing to release. |
| 14:43.01 | igcewieling | I also have a inbound fax script I hope to release at some point, nothing special |
| 14:43.35 | jkroon | ok, so i have issues even at a concurrency of a mere 27 concurrent calls. |
| 14:43.39 | Greenlight | And very much appriciated you making it available was - save me countless grey hairs, and am sure others could benefit from it too! |
| 14:44.02 | Greenlight | jkroon: This with things hanging after a reloiad ? |
| 14:45.32 | jkroon | Greenlight, no, this was from a clean start, clients are complaining about call breakups ... jitter, I'll see if I can get RTCP monitoring via AMI working tonight so that I can at least start getting a hold on this. |
| 14:51.48 | igcewieling | jkroon: Have you eliminated the possibility of network related issues? |
| 14:52.21 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
| 14:57.11 | igcewieling | jkroon: add a Dumpchan in your "h" extension (and anywhere else needed). The dumpchan info will include rtcp info too. |
| 14:57.58 | jkroon | igcewieling, all of out tests are testing clean, so yes, we're 99% sure there is no network issues. |
| 14:58.40 | jkroon | igcewieling, that's maybe not a bad idea. |
| 14:59.25 | igcewieling | the dumpchan info will also be written to the asterisk logs just like any other CLI output. It can be VERY useful. |
| 15:00.04 | jkroon | indeed. but at around 5GB worth of logs per day I really do NOT look forward to digging through that. |
| 15:00.27 | igcewieling | set your verbose to 1 if you are not trying to diagnose dialplan issues. |
| 15:01.08 | jkroon | no, dialplan has been stable for about 2+ years now. |
| 15:01.11 | *** join/#asterisk sjs205 (~sjs205@host81-151-44-189.range81-151.btcentralplus.com) |
| 15:01.18 | jkroon | except for a few minor issues, but nothing that would explain today. |
| 15:01.48 | jkroon | the *only* change that has happened since *perfectly working* last week to today was that we swapped out the hard drives for faster ones ... |
| 15:08.56 | *** join/#asterisk camerin (hoax@newelite2.bshellz.net) |
| 15:09.20 | *** join/#asterisk gnudna (~sklav@unaffiliated/sklav) |
| 15:10.02 | gnudna | guys quick question im having a nat problem i believe |
| 15:10.11 | gnudna | anybody willing to offer some insight |
| 15:10.40 | Greenlight | ~ask |
| 15:10.41 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 15:10.58 | gnudna | pubip --> fw --> internal asterisk |
| 15:11.13 | gnudna | all traffic to pubip is forwarded internally upd and tcp |
| 15:11.36 | gnudna | i have a remote user who is having issues connecting every so often |
| 15:11.56 | *** join/#asterisk ctaloi (~ctaloi@50-56-202-179.static.cloud-ips.com) |
| 15:12.01 | gnudna | Maximum retries exceeded on transmission ..... |
| 15:12.22 | gnudna | they are connected threw some linksys router and they connect to external ip |
| 15:12.31 | gnudna | in their setting i have nat-yes |
| 15:12.36 | *** join/#asterisk NicoR (~Nico@nat-office2-of1par.ip.rtblw.com) |
| 15:12.42 | gnudna | nat=yes |
| 15:12.43 | jkroon | gnudna, yes, quite likely. |
| 15:13.00 | jkroon | you need to set externip and localnet. |
| 15:13.05 | gnudna | i did |
| 15:13.20 | Greenlight | It only sometimes doesn't work ? |
| 15:13.40 | gnudna | in sip.conf [general] |
| 15:14.05 | gnudna | i have externip=x.x.x.x and localnet=192.168.11.0/255.255.255.0 |
| 15:14.22 | *** join/#asterisk Defraz (~Defraz@mail.pocatellochildren.com) |
| 15:14.46 | gnudna | yeah it was working all of last week |
| 15:15.09 | Greenlight | What's been chaged ? |
| 15:15.12 | gnudna | had some issue on monday of last week |
| 15:15.12 | Greenlight | *changed |
| 15:15.47 | gnudna | i added localnet=172.21.0.0/255.255.0.0 localnet=10.5.0.0/255.255.0.0 since the asterisk has ip in those spaces also |
| 15:15.59 | Greenlight | Is the problem occuring for other users, or just this 1 user ? |
| 15:16.14 | gnudna | i also added ignoresdpversion=yes |
| 15:16.27 | *** join/#asterisk blee (~blee@68.204.217.123) |
| 15:16.32 | jkroon | gnudna, what exactly is the symptom? |
| 15:16.37 | gnudna | because we were having issues with users on hold getting put back into the call queue |
| 15:17.03 | gnudna | only that user is having the issue which is also the only remote user we have connecting to asterisk |
| 15:17.45 | gnudna | this is what im seeing in logs chan_sip.c: Maximum retries exceeded on transmission fcfbfb10-a2850007-0e4a6285-5a294f1d@192.168.2.145 for seqno 101 (Critical Response) -- See doc/sip-retransmit.txt. |
| 15:18.01 | igcewieling | gnudna: did you read doc/sip-retransmit.txt? |
| 15:18.05 | gnudna | yes |
| 15:18.19 | igcewieling | Chances are putting Asterisk "in the DMZ" is the problem. |
| 15:18.51 | sjs205 | Hello all... I've finally managed to integarte asterisk and opensips... But when I dial my voicemail or the talking clock, I can see that asterisk is playing sound. But I can't hear anything. any ideas on how I can debug this? |
| 15:19.17 | gnudna | igcewieling, what do you mean? |
| 15:20.00 | Greenlight | [04:11pm] <gnudna> all traffic to pubip is forwarded internally upd and tcp <-- That sounds like it's in the "DMZ" |
| 15:20.04 | *** join/#asterisk camerin (hoax@newelite2.bshellz.net) |
| 15:20.26 | gnudna | Greenlight, yes but i would assume that that would cause no issues |
| 15:20.30 | igcewieling | gnudna: You only need to forward port 5060/udp. The RTP ports will generate the correct NAT translations when the first outbound audio packet is sent. |
| 15:20.53 | igcewieling | gnudna: NAT Fixup is "fragile", i.e. anything slightly different can mess it up. |
| 15:20.56 | jkroon | igcewieling, my experience dictates otherwise. it depends on who sends the first RTP :) |
| 15:21.20 | jkroon | anyway, i'm off |
| 15:21.35 | gnudna | so i am better just forwarding 5060 udp to asterisk |
| 15:21.43 | Greenlight | Laters jkroon... hope you fix your problem! |
| 15:21.51 | gnudna | and have nat=yes in [general] |
| 15:22.08 | jkroon | Greenlight, so do i ... but i suspect tomorrow is not going to be much better than what today was. |
| 15:22.20 | Greenlight | Alas, I know that feeling |
| 15:22.22 | jkroon | although, i must say, at least this afternoon held up much better. |
| 15:22.32 | Greenlight | Don't jinx it! |
| 15:22.37 | *** join/#asterisk RyanTG (~Thunderbi@65.100.106.194) |
| 15:22.39 | jkroon | the number of complaints reduced significantly. |
| 15:22.41 | igcewieling | "what could possibly go wrong?" |
| 15:22.46 | jkroon | but i'm not overly optimistic. |
| 15:22.53 | gnudna | can i post my sip.conf? for you guys to read? |
| 15:23.01 | Greenlight | ~pb |
| 15:23.01 | 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. |
| 15:23.08 | jkroon | igcewieling, well, i'll just assume that's a rhetorical question |
| 15:23.30 | gnudna | off course |
| 15:23.32 | gnudna | ;) |
| 15:26.54 | fafaflofly | Anyone know how i would setup the asterisk to be able to allow a caller to dial an extension out of the attendant to another PBX that is connected as a SIP extension (sip trunking on PBX side) without creating 300+ extensions in the asterisk that exist on the PBX? |
| 15:26.58 | *** join/#asterisk camerin (hoax@newelite2.bshellz.net) |
| 15:27.18 | ChannelZ | Make them follow a pattern |
| 15:27.53 | ChannelZ | (or rather pick them up as a pattern) |
| 15:27.55 | Greenlight | Sounds like you've got a single extension working, so like ChannelZ said, just adjust that to use a pattern match instead |
| 15:28.16 | Greenlight | Say, extension 200-299 were on the other PBX; |
| 15:28.22 | fafaflofly | ok |
| 15:28.35 | Greenlight | exten => _2XX,1,Dial(SIP/otherpnx/${EXTEN}) |
| 15:28.46 | Greenlight | Something like that |
| 15:29.06 | igcewieling | fafaflofly: Using patterns is the correct way, but do NOT think it is easy to create a simple dialplan to handle the complexities of the real world. dialplans are long, diaplans are ugly, dialplans are not elegany. |
| 15:29.10 | igcewieling | elegant, even |
| 15:29.39 | gnudna | Greenlight, http://pastebin.com/i6kaUyBc |
| 15:30.00 | gnudna | small summary since exntensions repeat with different username/password |
| 15:30.16 | fafaflofly | yeah i'm starting to learn that and yes i have a single extension connected and will try altering that dialplan |
| 15:30.17 | fafaflofly | thanks |
| 15:31.23 | Greenlight | And what's the exact symptom you're getting, gnunda? |
| 15:31.45 | gnudna | problem with ext e402 not being able to connect |
| 15:32.05 | gnudna | that is the remote user |
| 15:32.30 | gnudna | based on general settings do you see anything obvious? |
| 15:32.30 | Greenlight | Can you see him registered ? |
| 15:32.53 | igcewieling | gnudna: love your [general] context |
| 15:33.01 | Greenlight | +1 |
| 15:33.02 | gnudna | ;) |
| 15:33.43 | *** join/#asterisk camerin (hoax@newelite2.bshellz.net) |
| 15:33.58 | gnudna | at the moment the user went to lunch --> some creative use of language just ensued |
| 15:34.23 | *** join/#asterisk dtcrshr (~datacrush@unaffiliated/datacrusher) |
| 15:34.43 | *** part/#asterisk RyanTG (~Thunderbi@65.100.106.194) |
| 15:36.06 | Greenlight | In the meantime you could setup another remote extension to test with? |
| 15:36.28 | Greenlight | (The problem could well be at the user's side) |
| 15:37.04 | gnudna | i though of that but since it is intermittent managment blames asterisk |
| 15:37.14 | gnudna | even thought the 50 other users have no issue |
| 15:37.46 | gnudna | i guess i can use my android phone to test connecting from outside |
| 15:37.55 | gnudna | using that extension even |
| 15:38.06 | NicoR | Hi all, I have some question about billing. |
| 15:38.38 | igcewieling | gnudna: A softphone? Then you would have two problems. |
| 15:39.17 | gnudna | what issue would i have then? |
| 15:39.29 | gnudna | the user has a physical phone btw |
| 15:39.38 | igcewieling | having to diagnose softphone issuses at the same time as the NAT issue |
| 15:39.39 | NicoR | [Billing] Which is the best to use between CEL and CDR? (I was thinking it was CEL but I see that => https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+CDR+Specification ) |
| 15:39.59 | Greenlight | NicoR: I use CDR's for billing purposes. |
| 15:40.24 | igcewieling | NicoR: the BEST way is to use the CDRs from your carrier. If that is not an option, then I recommend CDRs as CEL is a very new feature. |
| 15:41.03 | NicoR | Actual (Asterisk 10) CDR implementation seems to be wrong in "complex" case |
| 15:41.15 | Greenlight | NicoR: How so ? |
| 15:42.38 | Greenlight | I'd always recommend having a separate box that acts as a gateway for generating CDR's if your using them for billing purposes. Perhaps overkill, but much much less to go wrong. |
| 15:43.08 | NicoR | Greenlight: From memory, especially in some transfert case (Sorry I have not specific example) |
| 15:43.09 | file | ahhhhhhhh CDRs! |
| 15:43.49 | Greenlight | NicoR: I recall issues in the past where transferring has made funny results in CDRs. It's one of the reasons why I'd say a separate box is the way to go. |
| 15:44.05 | file | yes, CDRs as they are can be wrong in some cases - they are being redone as part of the Asterisk 12 work and there is a specification at https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+CDR+Specification |
| 15:44.31 | Greenlight | eg. [Normal Asterisk Box] ---> [Asterisk Gateway for CDR's] ---> [Upstream Carrier] |
| 15:44.34 | file | they will be more sane but you will need to do post-processing to interpret and bill accordingly |
| 15:45.19 | NicoR | Greenlight: I understand but in my case I only have access to Asterisk data |
| 15:45.54 | NicoR | file: That exactly the link I post (See up) and that why I doubt of using CEL |
| 15:46.33 | file | if you can't write something to interpret call events and construct your view of what you need for billing then yeah, CEL isn't for you |
| 15:47.30 | gnudna | damn it i was able to connect using a sip phone |
| 15:47.41 | gnudna | going over carrier 3g |
| 15:47.46 | gnudna | no issues |
| 15:48.11 | NicoR | Oh no no. I can write a call events handler but will CEL be maintained in future Asterisk release? |
| 15:48.20 | igcewieling | gnudna: maybe the endpoint router has SIP ALG or SPI enabled? That would explain a lot of your issues. turn them OFF. |
| 15:48.30 | file | it'll still be there, yes |
| 15:48.49 | NicoR | The fact that this page (https://wiki.asterisk.org/wiki/display/AST/Generating+Billing+Information+from+CEL) have not update since 2010 let me doubt |
| 15:48.56 | gnudna | i figured my next step was to connect to their router |
| 15:49.10 | gnudna | on my side linux iptables so no magic options |
| 15:49.23 | Greenlight | My bettings on SIP ALG.... horrid "tech" |
| 15:49.29 | file | CDRs do not provide what CEL provides |
| 15:49.32 | gnudna | brb test logging into queue |
| 15:50.37 | NicoR | @file: I agree they provide a lot more. But I take a look at CEL only to "fix" present CDR issues in some case |
| 15:50.55 | FLeiXiuS | Is there any diagrams that show the order of communications for a SIP call? |
| 15:50.56 | igcewieling | We use CDR Custom to add the billing information we need. |
| 15:51.15 | *** join/#asterisk help_me_pls_ (d9a20f65@gateway/web/freenode/ip.217.162.15.101) |
| 15:51.38 | NicoR | But if in Asterisk 12, CDR are fixed. Does CEL are still a good way? :) |
| 15:51.53 | NicoR | (Only for my case) |
| 15:52.01 | file | CEL will still exist, and you can use it if you wish |
| 15:52.13 | igcewieling | NicoR: CDRs are "fixed" in 12? |
| 15:52.45 | leifmadsen | what does that even mean? :) |
| 15:52.48 | file | igcewieling, CDRs are being redone according to the above specification and should provide what people need |
| 15:52.54 | help_me_pls_ | hello.... i was working on our asterisk and rebooted and now nothing works anymore, cant call out or receive calls in... the error i see when i call with the CLI open is this: [Apr 15 17:43:50] NOTICE[3007]: chan_sip.c:18160 handle_request_invite: Call from '205' to extension '0793294524' rejected because extension not found. |
| 15:53.02 | help_me_pls_ | how do i troubleshoot this |
| 15:53.09 | [TK]D-Fender | help_me_pls_, "sip set debug on" <- |
| 15:53.16 | file | it means instead of the old "you get one record always" it becomes "you get multiple records" and have to post-process and bill accordingly |
| 15:53.48 | NicoR | Ok and does CEL documentation will evolve (Billing one in particular: https://wiki.asterisk.org/wiki/display/AST/Generating+Billing+Information+from+CEL ) ? |
| 15:54.01 | file | I know of noone working on CEL documentation. |
| 15:54.04 | [TK]D-Fender | help_me_pls_, You'll see "found peer XYZ for the call (or NOT. This would be bad), and then "looking for '0793294524' in [CONTEXT]. Go verify WHO it is matching, and WHERE it is looking |
| 15:54.14 | [TK]D-Fender | help_me_pls_, Because it is not finding a match |
| 15:56.44 | NicoR | Is there any project that already use CEL for billing? (An open source one should be a really good point :D) |
| 15:56.45 | help_me_pls_ | http://pastebin.com/Hc3G1D9K this is what im seeing now |
| 15:57.18 | file | NicoR, from my past experience when billing is involved people generally don't like to give out their code/approach |
| 15:57.35 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 15:58.42 | NicoR | @file, I understand. Thanks for your time |
| 16:00.39 | help_me_pls_ | this is what is get when i call out: [Apr 15 17:59:36] NOTICE[3007]: chan_sip.c:18160 handle_request_invite: Call from '205' to extension '0793294524' rejected because extension not found. |
| 16:00.50 | help_me_pls_ | and this is what i get when i call from outside: [Apr 15 18:00:24] NOTICE[3007]: chan_sip.c:18160 handle_request_invite: Call from 'sip-cablecom' to extension '0447227272' rejected because extension not found. |
| 16:02.31 | igcewieling | Remember everyone, when calling from the USA simply matching on 011 will NOT catch all international calls. |
| 16:02.35 | [TK]D-Fender | help_me_pls_, You call is NOT in that pastebin. |
| 16:02.43 | [TK]D-Fender | help_me_pls_, Place another where we can see the whole thing |
| 16:02.48 | igcewieling | This should match all international calls regex ^011|^1684|^1264|^1268|^1246|^1441|^1284|^1345|^1767|^1809|^1829|^1849|^1473|^1671|^1876|^1664|^1670|^1787|^1939|^1869|^1758|^1784|^1721|^1868|^1649|^1340 |
| 16:03.43 | Greenlight | You don't have an international dialling prefix over the pond? |
| 16:03.57 | igcewieling | There are 24 countries which are dialed as 1+ from the USA |
| 16:04.05 | help_me_pls_ | http://pastebin.com/8WBN1SRs |
| 16:04.18 | igcewieling | Greenlight: 1+ is somewhat unique. |
| 16:04.52 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 16:04.59 | Greenlight | Am sure here in UK, *all* international destinations are prefixed with the "00" international prefix |
| 16:05.09 | help_me_pls_ | i opend asterisk -r , sip set debug on, then i called and pasted everthing in that pastebin.... is this correct? |
| 16:05.13 | igcewieling | Greenlight: yeah, that is the LOGICAL thing to do. |
| 16:05.46 | igcewieling | Greenlight: Country code "1" consists of United States and its territories, Canada, Bermuda, and 17 nations of the Caribbean |
| 16:06.13 | Greenlight | Ahh I see |
| 16:06.13 | [TK]D-Fender | help_me_pls_, AGAIN, there is NO CALL in there. |
| 16:06.22 | help_me_pls_ | ??? |
| 16:06.24 | igcewieling | help_me_pls_: try asterisk -rvvv |
| 16:06.27 | [TK]D-Fender | help_me_pls_, Notice the lack of even seeing the ERROR message you showed us earlier |
| 16:07.05 | [TK]D-Fender | help_me_pls_, Make sure it's in there for the next pastebin |
| 16:07.32 | [TK]D-Fender | help_me_pls_, If you don't see any of what you were told should stand out or that you've already told us your were getting then it is not good. |
| 16:07.54 | help_me_pls_ | if i turn on debugging it moves really fast |
| 16:08.03 | help_me_pls_ | my verbosity is way to hig |
| 16:08.07 | help_me_pls_ | how do i lower it? |
| 16:08.17 | [TK]D-Fender | help_me_pls_, Don't. get a bigger buffer <-\ |
| 16:08.34 | Greenlight | Or use the logfile... |
| 16:09.18 | igcewieling | the list, in case anyone cares http://pastebin.ca/2359903 |
| 16:11.22 | help_me_pls_ | is this it? http://pastebin.com/G87N5HXN |
| 16:12.57 | help_me_pls_ | heres my extension.conf if that helps |
| 16:12.57 | help_me_pls_ | http://pastebin.com/6pr3Amt5 |
| 16:16.14 | [TK]D-Fender | heffer, No, that is not it... you are showing us the ERROR onwars.. not the stuff before that CAUSED it |
| 16:16.18 | [TK]D-Fender | help_me_pls_, ^ |
| 16:16.27 | help_me_pls_ | hmm |
| 16:17.07 | [TK]D-Fender | we nee everything that arrived BEFORE that point as well |
| 16:17.44 | help_me_pls_ | http://pastebin.com/1p5g0Dn0 |
| 16:18.16 | *** join/#asterisk giany (~giany@shifu.x83.org) |
| 16:18.24 | igcewieling | help_me_pls_: did you connect to Asterisk as "asterisk -rvvv" or as "asterisk -r" ? |
| 16:18.51 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 16:18.52 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 16:18.59 | giany | hi, when I'm logging like this : asterisk -rcvvvvvvvvvvvvvvvvvv > file.log the file.log contains some chars like "[1;30m+" , any idea how can i get rid of them |
| 16:19.03 | giany | ? |
| 16:19.03 | *** join/#asterisk tzafrir (~tzafrir@212.179.75.202) |
| 16:19.17 | [TK]D-Fender | help_me_pls_, Found peer 'sip-cablecom' for '0793294524' from 62.2.46.12:5060 <- that is the PEER it is matching. |
| 16:19.24 | [TK]D-Fender | help_me_pls_, Looking for 0447227272 in sip-trunk-cablecom (domain 217.162.15.106) |
| 16:19.28 | leifmadsen | giany: disable colours; those are ansi codes |
| 16:19.56 | [TK]D-Fender | help_me_pls_, it is looking for a match for "0447227272' in [sip-trunk-cablecom] |
| 16:20.24 | giany | leifmadsen: thx, how do i disable colors? |
| 16:20.31 | leifmadsen | asterisk -h |
| 16:20.53 | [TK]D-Fender | help_me_pls_, and you have NO match for it |
| 16:20.58 | help_me_pls_ | i used -rvvv (changed the verbosity to 3 instead of 29 like before |
| 16:21.12 | [TK]D-Fender | help_me_pls_, exten => _04472272[0-9][0-9],1,Answer() |
| 16:21.17 | help_me_pls_ | okay... how the hell did that happen after a reboot? |
| 16:21.25 | [TK]D-Fender | help_me_pls_, this will not match the number coming in |
| 16:21.31 | *** part/#asterisk igcewieling (~igcewieli@ip98-183-25-31.pn.at.cox.net) |
| 16:21.35 | [TK]D-Fender | help_me_pls_, this has nothing to do with "rebooting" |
| 16:21.49 | [TK]D-Fender | help_me_pls_, Your dialplan does not match the number they are sending you. |
| 16:21.59 | help_me_pls_ | [sip-trunk-cablecom] exten => _04472272[0-9][0-9],1,Answer() exten => _04472272[0-9][0-9],n,Goto(incoming-calls,${EXTEN:-4},1) |
| 16:22.06 | help_me_pls_ | is this correct? |
| 16:22.15 | help_me_pls_ | that s from my extensions.conf |
| 16:22.16 | [TK]D-Fender | help_me_pls_, no |
| 16:22.42 | [TK]D-Fender | Actually... |
| 16:22.43 | [TK]D-Fender | hrm |
| 16:23.15 | [TK]D-Fender | NO |
| 16:23.23 | giany | leifmadsen: so basically like this : sk channel 1 seems available. Channel can be used for outgoing calls. |
| 16:23.26 | giany | [2013-04-15 16:22:34] NOTICE[11613] provider.c: Asterisk channel 3 seems available. Channel can be used for outgoing calls. |
| 16:23.30 | giany | [2013-04-15 16:22:34] NOTICE[11613] provider.c: Asterisk channel 4 seems available. Channel can be used for outgoing calls. |
| 16:23.33 | giany | [2013-04-15 16:22:34] NOTICE[11613] provider.c: Asterisk channel 6 seems available. Channel can be used for outgoing calls. |
| 16:23.37 | giany | [2013-04-15 16:22:34] NOTICE[11613] provider.c: Asterisk channel 7 seems available. Channel can be used for outgoing calls. |
| 16:23.40 | giany | [2013-04-15 16:22:34] NOTICE[11613] provider.c: Asterisk channel 8 seems available. Channel can be used for outgoing calls. |
| 16:23.44 | giany | [2013-04-15 16:22:34] NOTICE[11613] provider.c: Asterisk channel 9 seems available. Channel can be used for outgoing calls. |
| 16:23.47 | Greenlight | _pb |
| 16:23.47 | giany | [2013-04-15 16:22:34] NOTICE[11613] provider.c: Asterisk channel 10 seems available. Channel can be used for outgoing calls. |
| 16:23.50 | Greenlight | ~pb |
| 16:23.50 | 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. |
| 16:23.51 | giany | [2013-04-15 16:22:34] NOTICE[11613] provider.c: Asterisk channel 11 seems available. Channel can be used for outgoing calls. |
| 16:23.52 | Greenlight | ^^^^ |
| 16:23.54 | giany | [2013-04-15 16:22:34] ERROR[11613] routing.c: Could not compute the cost of the route to <18563588844> on channel <#1> using the JSON Hubring engine. |
| 16:23.58 | giany | [2013-04-15 16:22:34] WARNING[11613] routing.c: Falling back to querying the database for a cost. |
| 16:24.01 | giany | [2013-04-15 16:22:34] ERROR[11613] routing.c: Could not find cost for number <18563588844> on channel <#1>. |
| 16:24.05 | giany | [2013-04-15 16:22:34] ERROR[11613] routing.c: Could not compute the cost of the route to <18563588844> on channel <#3> using the JSON Hubring engine. |
| 16:24.08 | giany | [2013-04-15 16:22:34] WARNING[11613] routing.c: Falling back to querying the database for a cost. |
| 16:24.12 | giany | [2013-04-15 16:22:34] ERROR[11613] routing.c: Could not compute the cost of the route to <18563588844> on channel <#8> using the JSON Hubring engine. |
| 16:24.15 | giany | [2013-04-15 16:22:34] WARNING[11613] routing.c: Falling back to querying the database for a cost. |
| 16:24.19 | giany | [2013-04-15 16:22:34] ERROR[11613] routing.c: Could not compute the cost of the route to <18563588844> on channel <#9> using the JSON Hubring engine. |
| 16:24.22 | giany | [2013-04-15 16:22:34] WARNING[11613] routing.c: Falling back to querying the database for a cost. |
| 16:24.26 | giany | [2013-04-15 16:22:34] ERROR[11613] routing.c: Could not compute the cost of the route to <18563588844> on channel <#10> using the JSON Hubring engine. |
| 16:24.29 | giany | [2013-04-15 16:22:34] WARNING[11613] routing.c: Falling back to querying the database for a cost. |
| 16:24.33 | giany | [2013-04-15 16:22:39] WARNING[11614] app_callrecord.c: record_always not enabled for 0195*0001 |
| 16:24.36 | giany | ^C |
| 16:24.39 | giany | [root@sip50 2238]# asterisk -rcvvvvvvvvvvn > /root/blah |
| 16:24.41 | giany | [root@sip50 2238]# vim /root/blah |
| 16:24.44 | giany | [root@sip50 2238]# vim /root/blah |
| 16:24.46 | giany | god dammit! |
| 16:24.49 | giany | sorry about that |
| 16:24.52 | giany | leifmadsen: asterisk -rcvvvvvvvvvvn > /root/blah would do it? thing is that even like that in /root/blah i still have those chars.. |
| 16:25.30 | help_me_pls_ | soooo... what do i have to do to fix this (im no expert at asterisk by any means) |
| 16:25.31 | [TK]D-Fender | help_me_pls_, pastebin "dialplan show sip-trunk-cablecom" |
| 16:26.51 | help_me_pls_ | <PROTECTED> |
| 16:27.42 | help_me_pls_ | what doe that mean? |
| 16:28.21 | [TK]D-Fender | help_me_pls_, "la -la /etc/asterisk" <- pastebin from linux CLI |
| 16:28.30 | [TK]D-Fender | help_me_pls_, In full with the command itself showing |
| 16:29.31 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 16:30.37 | help_me_pls_ | where exactly do i have to run this command? im using ubuntu server if that matters |
| 16:31.18 | [TK]D-Fender | LINUX CLI |
| 16:31.26 | Greenlight | facepalms |
| 16:34.03 | help_me_pls_ | i think i doing this wrong (im so sry, pls dont hurt me) all i get is "-bash: la: command not found" |
| 16:34.28 | leifmadsen | la is not a command |
| 16:34.29 | chuckf | ls? |
| 16:34.30 | leifmadsen | do you mean ls? |
| 16:34.37 | [TK]D-Fender | yes |
| 16:34.44 | Greenlight | He meant "ls -la" |
| 16:34.48 | [TK]D-Fender | help_me_pls_, ls -la /etc/asterisk |
| 16:34.49 | leifmadsen | heh |
| 16:34.53 | help_me_pls_ | oh |
| 16:34.58 | [TK]D-Fender | My typo |
| 16:35.44 | help_me_pls_ | http://pastebin.com/4ebwgvVv |
| 16:36.01 | help_me_pls_ | and here i was feeling stupid for not knowing which CLI to use it in ^^ |
| 16:36.29 | Greenlight | You, or someone, altered your extensions.conf file today |
| 16:36.34 | [TK]D-Fender | help_me_pls_, this is not looking good that your configs are all owned by root |
| 16:36.44 | Greenlight | It looks like they left a backup there though... |
| 16:36.46 | [TK]D-Fender | help_me_pls_, "diaplan show" <- PB from * CLI |
| 16:37.02 | help_me_pls_ | yes i did, but all i did was add 3 users... once it stoped working after a reboot i removed the changes i made |
| 16:37.37 | *** join/#asterisk mchou (~quassel@unaffiliated/mchou) |
| 16:37.42 | Greenlight | Why did you reboot ? |
| 16:38.22 | help_me_pls_ | cause im stupid thats why |
| 16:38.30 | leifmadsen | am I being punked? |
| 16:38.35 | Greenlight | No, I mean, there must have been a reason |
| 16:38.47 | Greenlight | leifmadsen: ?? |
| 16:38.49 | [TK]D-Fender | help_me_pls_, "dialplan show" <- PB from * CLI |
| 16:40.24 | Greenlight | I suspect you've made a typo when you altered extensions.conf |
| 16:40.38 | help_me_pls_ | http://pastebin.com/LVpcguV2 |
| 16:40.55 | [TK]D-Fender | help_me_pls_, Nothing loaded it seems |
| 16:41.05 | [TK]D-Fender | help_me_pls_, How did you install *? |
| 16:41.05 | help_me_pls_ | well i did try using a differen extension.conf, but that didnt work either.... |
| 16:41.38 | Greenlight | So... why did you reboot? What was changed? It worked before the reboot, but not after? |
| 16:42.02 | help_me_pls_ | and the reason why i rebooted was because i couldnt get the snom 320 phone to work. all iwas gettin when i dialed was 123: NOT FOUND |
| 16:42.15 | help_me_pls_ | and in the CLI i was getting the error |
| 16:42.26 | Greenlight | Right, so whatever happened was prior to the reboot |
| 16:42.27 | fafaflofly | Greenlight, thanks for the dialplan help that did work but how do i pass the digits dialed so they appear to the other PBX as the DID. |
| 16:43.02 | Greenlight | fafaflofly: Digits dialled are the the extension number, no ? |
| 16:43.17 | help_me_pls_ | before the reboot the phones still worked, i could receive calls and make them |
| 16:43.20 | fafaflofly | yes, 3XX |
| 16:43.23 | help_me_pls_ | but just that one üphne didnt |
| 16:44.07 | Greenlight | fafaflofly: Those should be passed accross to the other PBX |
| 16:44.21 | Greenlight | ${EXTEN} gets passed over, and that would be 301 for example |
| 16:45.02 | help_me_pls_ | i was getting an error like this when i tried calling on the snom phone: chan_sip.c:18160 handle_request_invite: Call from '' to extension '' rejected because extension not found |
| 16:45.03 | [TK]D-Fender | help_me_pls_, Lets try this again. |
| 16:45.08 | help_me_pls_ | yes please |
| 16:45.13 | [TK]D-Fender | <[TK]D-Fender> help_me_pls_, How did you install *? |
| 16:45.27 | fafaflofly | interesting, my PBX is saying NO DID received |
| 16:46.32 | [TK]D-Fender | fafaflofly, those are not words Asterisk is programmed to use... |
| 16:47.18 | help_me_pls_ | all i did today was add following to extension.conf: http://pastebin.com/YAmqH95F |
| 16:47.47 | fafaflofly | [TK]D-Fender: no i know, that just what my PBX is reports back when it doesn't receive a DID |
| 16:47.51 | help_me_pls_ | the asterisks has been running for a long time without problem and ive added users like that before.... |
| 16:48.02 | help_me_pls_ | i then tried to configure the phone |
| 16:48.13 | [TK]D-Fender | fafaflofly, You should probably be looking at the call debug in full |
| 16:48.14 | help_me_pls_ | and once that didnt work i rebooted and all hell broke loose |
| 16:48.20 | Greenlight | fafaflofly: You could do a SIP trace, and you'll see them being passed over. Not sure why your other PBX isn't processing them correctly. |
| 16:48.45 | [TK]D-Fender | help_me_pls_, If you aren't going to answer the simple questions I've asked I'm going to simply move on to other matters.... |
| 16:48.51 | Qwell | His config is invalid. Next? |
| 16:48.55 | fafaflofly | Greenlight, i'll do that right now |
| 16:48.58 | fafaflofly | thanks |
| 16:49.33 | Greenlight | fafaflofly: I seem to recall some other PBX's wanting the "from" number if different places... but can't remember any more specifically. |
| 16:49.38 | help_me_pls_ | how did i install? install what? the asterisk? i didnt |
| 16:49.51 | [TK]D-Fender | help_me_pls_, how was it installed? |
| 16:49.53 | help_me_pls_ | someone else did and he is not here anymore.... and that was a long time ago |
| 16:50.06 | help_me_pls_ | i dont know |
| 16:50.16 | Qwell | ...what version of Asterisk are you running? |
| 16:50.36 | Greenlight | My betting is 1.2 |
| 16:50.54 | Qwell | My money is on 1.6.0.x |
| 16:51.05 | [TK]D-Fender | help_me_pls_, since your dialplan is not even loading I am suspecting that your user permissions on the entire folder are now screwed up. |
| 16:51.13 | [TK]D-Fender | help_me_pls_, Go validate who * is set to run as |
| 16:51.18 | Qwell | [TK]D-Fender: invalid syntax causes that |
| 16:51.22 | [TK]D-Fender | Qwell, Which? |
| 16:51.37 | Qwell | all |
| 16:51.38 | [TK]D-Fender | Qwell, I've never seen any that would kill the entire dialplan load |
| 16:51.51 | Qwell | [TK]D-Fender: Add this to your config. [broken |
| 16:51.57 | Qwell | done and done |
| 16:52.04 | [TK]D-Fender | Qwell, News to me.... |
| 16:52.14 | Qwell | Been that was for ~5 years. |
| 16:52.16 | Qwell | way |
| 16:52.29 | Greenlight | I supect just such a typo in extensions.conf |
| 16:52.38 | Greenlight | I also reckon asterisk is running as root anyways |
| 16:53.08 | Greenlight | help_me_pls_: Can you PB your extensions.conf ? |
| 16:53.29 | [TK]D-Fender | Qwell, I don't see a broken example in his pastebin of his dialplan... perhaps I'm a little blind this morning (ish). Can you point it out? |
| 16:53.41 | help_me_pls_ | yes, pasted it above, give me a sec |
| 16:53.54 | [TK]D-Fender | http://pastebin.com/6pr3Amt5 <- his dialplan |
| 16:54.15 | Greenlight | Ahh missed that being pasted |
| 16:54.37 | help_me_pls_ | http://pastebin.com/NzUHXNuH |
| 16:55.17 | help_me_pls_ | how do i validate who * is set to run as? |
| 16:57.35 | Qwell | His permissions are fine... it's all ugo+r |
| 16:57.44 | Qwell | dir might be wrong, but that's rather unlikely |
| 16:58.48 | help_me_pls_ | is there a way to verify my configs? |
| 17:00.12 | Greenlight | Nothing jumps out as obviously wrong with that dialplan, to the point it wouldn't load |
| 17:01.46 | Greenlight | It's possible that the reboot triggered a version change, or something like that, if it's been updated from a repo in the past |
| 17:02.02 | [TK]D-Fender | help_me_pls_, "dialplan reload" <- |
| 17:03.39 | help_me_pls_ | No such command 'dialplan reload' (type 'help dialplan reload' for other possible commands) -> do you mean: /etc/init.d/asterisk reload ??? |
| 17:03.50 | help_me_pls_ | or am i just doing it wrong? |
| 17:03.57 | Greenlight | How odd |
| 17:04.06 | Qwell | help_me_pls_: What version of Asterisk? |
| 17:04.06 | Greenlight | "core show version" |
| 17:04.25 | help_me_pls_ | Asterisk 1.6.1.4 built by root @ asterisk on a i686 running Linux on 2009-08-28 07:24:12 UTC |
| 17:04.57 | Greenlight | There's a module not loaded then |
| 17:04.58 | Qwell | module show like pbx_ |
| 17:06.02 | help_me_pls_ | http://pastebin.com/9DYeaUXf |
| 17:06.22 | Greenlight | heh |
| 17:07.07 | help_me_pls_ | ?? |
| 17:07.21 | Greenlight | I was expecting for something to be missing... alas there's not |
| 17:08.10 | help_me_pls_ | damn |
| 17:08.13 | Greenlight | Try stopping asterisk |
| 17:08.24 | Greenlight | Then starting it with "asterisk -cvvvvv" |
| 17:08.29 | Greenlight | The pastebin that output |
| 17:10.57 | help_me_pls_ | http://pastebin.ca/2359958 |
| 17:11.52 | Greenlight | [Apr 15 19:09:24] WARNING[3267]: config.c:1102 process_text_line: parse error: No category context for line 1 of /et |
| 17:11.54 | Greenlight | ^^^ |
| 17:12.07 | help_me_pls_ | [Apr 15 19:10:38] WARNING[3302]: config.c:1102 process_text_line: parse error: No category context for line 1 of /etc/asterisk/extensions.conf |
| 17:12.19 | help_me_pls_ | what does that mean? |
| 17:12.34 | Greenlight | Remove the first line, and try it |
| 17:12.49 | Qwell | umm |
| 17:12.55 | Qwell | I've got $20 on ^M |
| 17:13.01 | Qwell | any takers? |
| 17:13.02 | Greenlight | You've a blank first line, but why that would cause issues is beyond me |
| 17:13.29 | Greenlight | No way, not taking that bet! :) |
| 17:13.40 | Qwell | help_me_pls_: You edited the file on Windows, didn't you? |
| 17:14.14 | help_me_pls_ | okay so the first line was missing a ; |
| 17:14.16 | help_me_pls_ | [Apr 15 19:13:30] WARNING[3338]: pbx.c:8870 ast_context_verify_includes: Context 'inspectron' tries to include nonexistent context 'outgoing-calls-cablecom-inspectron2' |
| 17:14.20 | help_me_pls_ | no im getting this |
| 17:14.38 | Greenlight | help_me_pls_: That's not a serious error |
| 17:14.42 | fafaflofly | Greenlight, thanks for the help it was sending the DID, the PBX's debug apparently wasn't reading it but the DID tables did get it. Another case of wasted hours looking for a problem that wasn't there. |
| 17:14.48 | sjs205 | hello all, trying to build dahdi kernel module on ubuntu... failes with fatal error: asm/system.h: No such file or directory - kernel headers and sources installed... any idea how to fix? |
| 17:15.02 | Greenlight | fafaflofly: No probs, glad it's working now! |
| 17:15.16 | Greenlight | sjs205: Install the kernel sources |
| 17:15.34 | Greenlight | Umm, on Ubuntu, would it be "apt-get install kernel-devel" ? |
| 17:15.37 | sjs205 | Greenlight, the kernel sources are installed |
| 17:15.43 | sjs205 | Ah, maybe that is the one... |
| 17:15.58 | Greenlight | Not 100% sure as I don't use Ubuntu on daily basis |
| 17:16.19 | Greenlight | But on CentOS the package is called "kernel-devel" |
| 17:16.28 | Greenlight | help_me_pls_: Try making a call now ? |
| 17:16.49 | sjs205 | greanlight *-devel is fedora and the likes... I think it is just linux-source... which is installed |
| 17:17.05 | help_me_pls_ | äwoeihveoabvdfopipdfh¨jhwerf üoiuwreg hoaierjgejrgihgr |
| 17:17.08 | help_me_pls_ | i works |
| 17:17.13 | help_me_pls_ | i love you guys |
| 17:17.14 | Greenlight | Ahh, and did you do ./confdigure again after installing them ? |
| 17:17.24 | help_me_pls_ | i cant fukin believe it |
| 17:17.30 | help_me_pls_ | was it just that missing ; |
| 17:17.32 | help_me_pls_ | ??? |
| 17:17.37 | help_me_pls_ | really? |
| 17:17.39 | Greenlight | Naa, just remove that line tbh |
| 17:17.43 | help_me_pls_ | or am i miisng something here? |
| 17:17.52 | Greenlight | It's perhaps expecting a context on the first line |
| 17:18.00 | Greenlight | ";" makes it ignore it |
| 17:18.08 | Greenlight | Or... you have a funky character there |
| 17:18.12 | help_me_pls_ | <PROTECTED> |
| 17:18.21 | help_me_pls_ | the second like this |
| 17:18.27 | Greenlight | Did you add "===" there ? |
| 17:18.40 | help_me_pls_ | ; =========== textetextetextetexte ==============000000 |
| 17:18.43 | help_me_pls_ | without those zeros |
| 17:18.47 | Greenlight | There were *not* in your pastebin! |
| 17:19.01 | Greenlight | &They |
| 17:19.10 | help_me_pls_ | yes and im very sorry for wasting your time like this |
| 17:19.21 | Greenlight | Well at least it's working now.. |
| 17:19.25 | help_me_pls_ | yeah |
| 17:19.28 | help_me_pls_ | tell me about it |
| 17:19.40 | help_me_pls_ | i couldve gone home 3 hours ago ^^ |
| 17:20.07 | help_me_pls_ | well thanks guys |
| 17:20.17 | Greenlight | Speaking of which, am off, laters |
| 17:20.18 | help_me_pls_ | i think ive had enough asterisk for this week |
| 17:20.31 | sjs205 | Greenlight, I am not configuring manually... I'm using some bizarre 'm-a a-i dahdi' command... which makes NO sense to me... |
| 17:20.40 | sjs205 | Maybe a mnaul install will sort it... |
| 17:21.35 | help_me_pls_ | thank you defender guy, you really helped me, there no way i could ever express the size of my gratitude to you here on irc |
| 17:21.37 | help_me_pls_ | ^^ |
| 17:21.42 | Qwell | wat |
| 17:21.50 | help_me_pls_ | and thank you everybody else that helped me |
| 17:21.58 | [TK]D-Fender | help_me_pls_, You're welcome |
| 17:23.24 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 17:23.31 | help_me_pls_ | im gonna go home now and watch game of thrones.... ill probably be back within a fortnight with mor stupid questions.... till we meet again my saviors :P |
| 17:23.43 | Qwell | help_me_pls_: Tyrion dies |
| 17:23.45 | Qwell | aww |
| 17:23.53 | [TK]D-Fender | :p |
| 17:23.54 | Qwell | (he doesn't actually die) |
| 17:24.09 | [TK]D-Fender | Qwell, You are a bad, bad man... |
| 17:24.20 | Qwell | I was trying to help him get those 3 hours back. :p |
| 17:28.58 | *** join/#asterisk MrMeek (~meekhime@172-4-223-5.lightspeed.toldoh.sbcglobal.net) |
| 17:30.39 | *** join/#asterisk Ashutto (~Ashetic@2001:b05:0:b0:6ef0:49ff:fe7d:3d35) |
| 17:30.42 | Ashutto | Hello |
| 17:31.45 | Ashutto | i'm behind full natted ip, without any chance to have pat to my private ip. is it possible to have a working rtp stream in such conditions ? |
| 17:33.35 | fafaflofly | Greenlight, how do i get the IVR to recognize the new dial patterns so callers can direct dial out of it? |
| 17:43.19 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 17:46.55 | fafaflofly | Greenlight, n/m its been a long day, i created the from-internal and not the ivr-custom |
| 17:50.31 | *** join/#asterisk jagster` (~chatzilla@unaffiliated/jagster/x-9084543) |
| 18:00.48 | *** join/#asterisk igcewieling (~igcewieli@ip98-183-25-31.pn.at.cox.net) |
| 18:02.31 | *** join/#asterisk lvlinux (~n1gg@c-50-142-165-230.hsd1.tn.comcast.net) |
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| 19:05.50 | sjs205 | 2 machines on the same network, internal firewalls off but still no sound when I call my talking clock, although asterisk says it is playing... any ideas? |
| 19:06.36 | [TK]D-Fender | sjs205, Look at the call. |
| 19:06.54 | [TK]D-Fender | sjs205, Optionally (preferably), show US the call. |
| 19:07.30 | sjs205 | [TK]D-Fender, 2 secs, I'll pastebin it... :) cheers |
| 19:09.29 | *** join/#asterisk BrokenArrow (~BrokenArr@unaffiliated/brokenarrow) |
| 19:09.34 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 19:11.32 | sjs205 | [TK]D-Fender, this is my pastebin... http://pastebin.com/SBUXigcn |
| 19:12.42 | *** join/#asterisk serafie (~erin@24.214.158.242) |
| 19:12.59 | sjs205 | it seems like it should be workign, I just hear nothing... |
| 19:20.47 | [TK]D-Fender | sjs205, * CLI verbose 10, sip debug |
| 19:20.50 | [TK]D-Fender | no 3rd party |
| 19:21.19 | sjs205 | 10... I didn't even know there was a 10!!! |
| 19:21.22 | sjs205 | haha |
| 19:23.00 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 19:23.02 | sjs205 | [TK]D-Fender, I've got v @ 10... what is that 'sip debug'? |
| 19:23.24 | [TK]D-Fender | "sip set debug on" |
| 19:23.30 | [TK]D-Fender | No external TCPdump, etc |
| 19:25.32 | sjs205 | here ges [TK]D-Fender |
| 19:25.44 | sjs205 | http://pastebin.com/EyBuF9xN |
| 19:26.18 | *** join/#asterisk bsaggy (~bsaggy@unaffiliated/bsaggy) |
| 19:26.38 | sjs205 | Cheers for looking |
| 19:26.50 | *** join/#asterisk kuku (~kuku@173-167-188-106-Illinois.hfc.comcastbusiness.net) |
| 19:28.22 | igcewieling | sjs205: you have one SERIOUSLY screwed up config Reliably Transmitting (NAT) to 127.0.0.1:5060: |
| 19:28.43 | igcewieling | You're not trying something stupid like running your softphone ON the actual Asterisk server, are you? |
| 19:29.27 | kuku | I'm being told that I'm not sending back a packter after na Invite - what I can use to capture the call and see if the packet is there? |
| 19:30.20 | sjs205 | igcewieling, ehh? where would i have set that? |
| 19:30.34 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 19:30.35 | igcewieling | kuku: chances are you are sending the packet to the wrong IP. Start out with a sip debug, if nothing obvious then use tcpdump or ethereal |
| 19:30.39 | *** join/#asterisk Sorcier_FXK (~nssystem@unaffiliated/sorcierfxk) |
| 19:30.45 | [TK]D-Fender | INVITE sip:AN_time@127.0.0.1:6267 SIP/2.0 |
| 19:30.48 | igcewieling | sjs205: Are you running a softphone on the same box as Asterisk |
| 19:30.52 | sjs205 | I have openSIPS doing all of the sips work and asterisk just doing the media stuff... |
| 19:31.07 | [TK]D-Fender | <--- SIP read from UDP:192.168.5.20:5060 ---> Contact: <sip:alice@192.168.5.10> <- MESSED UP |
| 19:31.11 | [TK]D-Fender | Looking like NAT'd |
| 19:31.18 | igcewieling | Ah. I doubt you'll get a lot of help with that sort of setup, it is too complex |
| 19:31.26 | kuku | igcewieling: I have 40 active calls. Doing a sipd ebug will flood the screen with data |
| 19:31.29 | sjs205 | igcewieling, no, it is running on my desktop |
| 19:31.45 | igcewieling | kuku: then just debug the peer which is having issues |
| 19:32.03 | [TK]D-Fender | sjs205, Why is twinkle SOURCING from one IP and saying its contact is at another? |
| 19:32.09 | sjs205 | Tell me about it... I've lost hair this week, and this is the fisrt time I've ever looked at asterisk and opensips |
| 19:32.30 | igcewieling | sjs205: unless you REALLY understand SIP, using a SIP proxy is just crazy. |
| 19:32.33 | kuku | igcewieling: but then if it goes to a different ip I will not be able to see it ( "sending the packet to the wrong IP" ) |
| 19:32.41 | sjs205 | I'm not using a proxy... that i know of! |
| 19:32.58 | sjs205 | :/ |
| 19:33.03 | igcewieling | kuku: which is why I also said tcpdump |
| 19:33.09 | igcewieling | sjs205: if OpenSIPs is not a sip proxy, what is it? |
| 19:33.21 | sjs205 | A sip server? haha |
| 19:33.23 | kuku | igcewieling: do you have the parameters to capture full pbackets? |
| 19:33.32 | sjs205 | Sorry, yeah, that is how sip experienced I am! |
| 19:33.51 | igcewieling | kuku: tcpdump -i ethX -X -s 4096 -v port 5060 |
| 19:35.13 | sjs205 | So do you guys know what the main issue here is, other than PEBKAC |
| 19:35.16 | sjs205 | ? |
| 19:35.21 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 19:35.22 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 19:36.05 | sjs205 | and [TK]D-Fender Sourcing from one ip and saying its contact is another? |
| 19:36.13 | *** join/#asterisk gusto (~gusto@2001:470:1f0b:a42:224:1dff:fecd:234c) |
| 19:36.32 | [TK]D-Fender | http://pastebin.com/EyBuF9xN <---- 13/23 |
| 19:37.35 | sjs205 | [TK]D-Fender, 192...20 is my sever, port 6267 is the connection between asterisk and opensips... |
| 19:38.00 | igcewieling | sjs205: I'd not touch your config with a 10 foot pole and oven mitts. |
| 19:38.11 | sjs205 | igcewieling, me either! ;) |
| 19:38.33 | igcewieling | I do, however, wish you the best of luck. I assume you have a REASON for using a SIP proxy with Asterisk? |
| 19:39.19 | sjs205 | I thought this was a standard install... most of it is from the tutorial re integration on openSIPS docs |
| 19:39.45 | igcewieling | No, a standard Asterisk install does not use OpenSIPS |
| 19:39.46 | sjs205 | igcewieling, I plan to manage a large number of calls at some point in the future... I thought this was the best option... |
| 19:39.56 | igcewieling | sjs205: define "large number" |
| 19:40.13 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 19:40.13 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 19:40.18 | sjs205 | worst case, 1000... |
| 19:40.24 | sjs205 | haha... don't laugh! |
| 19:40.36 | sjs205 | But even that is extremely unlikely... |
| 19:40.53 | igcewieling | OK, I can see wanting to use a proxy for that, but you should get all your asterisk stuff working as expected before you increase the complexity 100fold by adding a sip proxy |
| 19:41.00 | sjs205 | So would you suggest scrapping opensips and just using asterisk? |
| 19:41.35 | igcewieling | sjs205: it is VERY easy to accidentally allow unauthenticated calls thru Kamailio or OpenSIPS. |
| 19:41.54 | sjs205 | Ahh, okay, sure... so initially I should just try to get asterisk set up and working without the proxy... and then add later? |
| 19:42.11 | igcewieling | sjs205: I suggest you scrap OpenSIPS until you are fully comfortable with Asterisk. I also recommend putting the proxy on a seperate box. |
| 19:42.16 | sjs205 | Actually, the server will not be recieving many calls at all, mostly making them. |
| 19:42.41 | sjs205 | okay... that sounds like it could definitly simplify things somewhat... |
| 19:43.00 | sjs205 | cheers igcewieling... I'll give that a go. |
| 19:44.35 | [TK]D-Fender | sjs205, <--- Transmitting (NAT) to 192.168.5.20:5060 ---> <--- it is NOT NAT'd, it is PROXIED |
| 19:44.43 | [TK]D-Fender | sjs205, set your peers to NAT=NO |
| 19:45.21 | [TK]D-Fender | sjs205, just because your SIP is proxied doesn't mean your MEDIA is. |
| 19:45.45 | sjs205 | okay... sure... [TK]D-Fender... that bit is directly from the tutorial... |
| 19:46.23 | sjs205 | I've spent the better part of two weeks trying to get this setup going... today was the first time I was even able to make a call... |
| 19:46.41 | sjs205 | ha! |
| 19:49.32 | [TK]D-Fender | sjs205, Is this to say you are now getting the expected audio? |
| 19:50.22 | sjs205 | no... haha... I've just made the call... no sound... that was to mearly say haw frustrating these weeks have been, and how excited i was at finally being able to make a call! |
| 19:50.42 | sjs205 | I mean i just made the change, and still no sound... |
| 19:50.46 | sjs205 | pastebin again? |
| 19:51.44 | [TK]D-Fender | yes |
| 19:52.01 | sjs205 | [TK]D-Fender, http://pastebin.com/iBdrEshw |
| 19:52.03 | sjs205 | :D |
| 19:52.05 | sjs205 | Thank you! |
| 19:52.35 | [TK]D-Fender | <--- Transmitting (no NAT) to 192.168.5.20:5060 ---> Contact: <sip:AN_time@127.0.0.1:6267> |
| 19:52.44 | [TK]D-Fender | sjs205, now the contact is a PRIVATE IP |
| 19:52.50 | [TK]D-Fender | sjs205, Your setup is getting worse |
| 19:53.33 | [TK]D-Fender | Actually this is it's own audio |
| 19:53.35 | [TK]D-Fender | WTF |
| 19:53.38 | sjs205 | I think that is due to how the users are shared between opensips and asterisk, that is, the sipsuser table is mearly a view exported from opensips |
| 19:53.54 | [TK]D-Fender | .... |
| 19:54.03 | sjs205 | "It's own audio"??? like music? haha |
| 19:54.18 | sjs205 | it is singing! in the shower! |
| 19:54.25 | [TK]D-Fender | ok, your proxy is passing crap because of all being local. |
| 19:54.38 | [TK]D-Fender | Go learn your proxy or a minimum of NOT installing it on your ASTERISK server |
| 19:55.04 | sjs205 | [TK]D-Fender, okay, sure... so I gues next step is to find the a spare server t |
| 19:56.09 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 19:56.42 | sjs205 | [TK]D-Fender, I "liked" this setup because the users were shared... I'm guessing that I will have to use a script to export the opensips users to asterisk... |
| 19:56.54 | sjs205 | ? |
| 19:57.15 | *** join/#asterisk deviantlinux (~ryan_gpx@unaffiliated/deviantlinux) |
| 19:57.33 | deviantlinux | Anyone have experience with any of the Openvox failover switches? |
| 19:58.31 | sjs205 | actually, that is what I was unsure of, isn't the sipserver/proxy in a better position to manage presence and users etc? |
| 19:58.56 | sjs205 | rather than just using asterisk to manage this? |
| 20:00.31 | [TK]D-Fender | sjs205, You are trying to fly before even leaving the delivery-room.... |
| 20:01.29 | sjs205 | I know... but I work in a tiny company and we are trying to develop a new system... when I say tiny, I mean that I am the only software guy... and this is what I've been tasked to do. |
| 20:02.29 | sjs205 | Anyway... I need a drink... it has been a long day... Tomorrow I'll get a post on the forum and describe exactly what it is that I'm trying to do... I'll link it to you tomorrow if you are at all interested?? Many thanks for your help [TK]D-Fender and igcewieling, it is very much appreciated... |
| 20:04.56 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.56) |
| 20:07.15 | *** join/#asterisk pib1978 (pib1978@2600:3c00::f03c:91ff:fe70:bb80) |
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| 20:24.03 | jagster` | haha |
| 20:24.09 | jagster` | sjs205: i'm thinking of picking up smoking tobacco |
| 20:24.16 | jagster` | after spinning my wheels on asterisk for a week |
| 20:27.14 | sjs205 | jagster, i'M ALREAD ONE THAT ;) |
| 20:27.21 | sjs205 | ON* |
| 20:31.49 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 20:35.14 | *** join/#asterisk Sorcier_FXK (~nssystem@unaffiliated/sorcierfxk) |
| 20:43.28 | *** join/#asterisk pib1951 (pib1951@2600:3c00::f03c:91ff:fe70:bb80) |
| 20:44.54 | igcewieling | other natural smokable plant material may be more effective at relieving stress. |
| 20:47.19 | gnudna | ;) |
| 20:47.57 | ryan_gpx | I'm trying to see if anyone has successfully failed over an OpenVox FD40 device with asterisk after pulling power cables? |
| 20:48.08 | *** join/#asterisk Rahail (~Rahail@67.214.121.163) |
| 20:48.13 | Rahail | hi i got quesiton i install ssh |
| 20:48.17 | Rahail | but i get this error |
| 20:48.27 | Rahail | ./usr/src/asterisk-1.8.21.0/include/asterisk/crypto.h:145 __stub__ast_aes_set_encrypt_key: AES encryption disabled. Install OpenSSL. |
| 20:48.55 | ryan_gpx | it's almost like it's asking you to install openssl |
| 20:49.01 | gnudna | is openssl installed? |
| 20:49.06 | Rahail | yes |
| 20:49.16 | igcewieling | try the -dev packages. and learn your distro |
| 20:51.00 | Rahail | how |
| 20:51.03 | Rahail | sorry not expart |
| 20:51.17 | ryan_gpx | Rahail: this is an asterisk channel not a "teach linux" channel. |
| 20:51.36 | ryan_gpx | Rahail: we don't know if you are on Centos, ubuntu, or what Linux distro you are using |
| 20:51.52 | Rahail | i am on ubuntu |
| 20:51.53 | Rahail | sorry |
| 20:52.08 | Rahail | i did ask linux people but they give me some command i typed it and they told me ssl is there |
| 20:52.22 | gnudna | dpkg -l |grep openssl |
| 20:52.30 | gnudna | what does that display? |
| 20:52.39 | Rahail | ii openssl 0.9.8k-7ubuntu8.14 Secure Socket Layer (SSL) binary and related |
| 20:52.41 | Rahail | ii openssl-blacklist 0.5-2 |
| 20:53.02 | Rahail | ii openssl 0.9.8k-7ubuntu8.14 |
| 20:54.10 | pabelanger | ~pb |
| 20:54.10 | 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. |
| 20:54.15 | pabelanger | Rahail: please ^ |
| 20:55.01 | Rahail | http://pastebin.com/45mB9n9d |
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| 20:58.06 | Morcegolas | Hello, I'm having problems in outgoing calls, incoming is OK, but outgoing always says all circuits are busy now, try again later, can anybody help me please?!? |
| 20:59.47 | WIMPy | Where do you get that message? Are you using some GUI? |
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| 21:00.17 | Rahail | WIMPy me or Morcegolas |
| 21:00.30 | WIMPy | Morcegolas |
| 21:01.16 | Morcegolas | I'm using terminal and GUI. |
| 21:01.32 | igcewieling | Morcegolas: have you tried asking on a channel for your GUI? |
| 21:01.57 | WIMPy | Morcegolas: Then you should ask is the support channel of the GUI you're using. |
| 21:02.32 | Rahail | igcewieling plz help me with openssl |
| 21:02.33 | Morcegolas | Like free pbx right? |
| 21:02.33 | Rahail | enable |
| 21:02.45 | WIMPy | #freepbx |
| 21:02.46 | Rahail | igcewieling http://pastebin.com/45mB9n9d |
| 21:02.58 | igcewieling | Rahail: I have never in my life used Ubuntu |
| 21:03.04 | WIMPy | Rahail: And you should try the support channel of your distro. |
| 21:03.17 | slav3_kitten | cackles like a nutcase |
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| 21:04.01 | Morcegolas | "TRUNK Dial failed due to CHANUNAVAIL HANGUPCAUSE: 0 - failing through to other trunks" but this can be what do you know? |
| 21:04.04 | slav3_kitten | so... my "isp" bought their own IP block, didn't inform me what so ever (how the fuck do you not inform your static IP customers their shit is going to change) |
| 21:04.27 | slav3_kitten | and well the IP addy i'm seeing isn't the one i should be seeing |
| 21:04.31 | WIMPy | Morcegolas: We don;t know what FreePBX has configured for you. Ask in #freepbx |
| 21:04.40 | igcewieling | On CentOS you would run "yum install openssl-devel" |
| 21:04.43 | slav3_kitten | which kinda explains why sip has shit itself, i'm seeing tons of jitter, and up to 50% packet loss |
| 21:04.45 | Morcegolas | Ok thanks |
| 21:09.17 | lvlinux | slav3_kitten: u get ur audio working? |
| 21:09.39 | slav3_kitten | not yet |
| 21:10.08 | lvlinux | hmm thats no good. |
| 21:10.28 | lvlinux | Does anyone here have any experience with Adtran TA6xx units? |
| 21:10.43 | slav3_kitten | they are trying to figure out why the IP i'm seeing externally is different then the IP they say is my static |
| 21:11.13 | lvlinux | they are trying to figure it out? wow so they still say you have the same static ip officially? |
| 21:11.29 | slav3_kitten | so i've had about 8 network "engineers" if you'll call them that playing "lets try this!" |
| 21:11.30 | slav3_kitten | no, the ip officially changed |
| 21:11.38 | slav3_kitten | i'm just not seeing the official new ip |
| 21:11.52 | lvlinux | so you're still seeing the old one? |
| 21:11.58 | slav3_kitten | real d-bag move on their part not to even give me a courtesy call telling me my ip was going to change |
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| 21:12.11 | lvlinux | yeah that's pretty bad... |
| 21:12.57 | lvlinux | did you do any SIP diags like they were talkin in here about the other day? |
| 21:13.17 | slav3_kitten | i've got the feeling half of their shit got mis-configured in the transition |
| 21:13.38 | slav3_kitten | yea i did. shows the external ip i'm seeing via whatismyip.com and everything |
| 21:14.30 | lvlinux | you mean it shows the one you used to have? |
| 21:14.34 | slav3_kitten | i know they had to do static routing for my RTP ports and i have a feeling those are screwed up which is explaining why it just gives no audio either way despite initiating the call |
| 21:14.54 | slav3_kitten | no, it has the one i'm currently seeing but not the one they are telling me is the new one i should have |
| 21:17.07 | lvlinux | by "they had to do static routing" do you mean they had to manually forward RTP to your address? |
| 21:17.24 | slav3_kitten | ya |
| 21:17.44 | lvlinux | that sounds like they are NATting you |
| 21:18.13 | slav3_kitten | well it is kind of natted. |
| 21:18.35 | slav3_kitten | i've got a 10. ip to their main office which has the edge router to the actual internet |
| 21:18.44 | slav3_kitten | because it's a wireless isp |
| 21:18.47 | lvlinux | ahhh |
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| 21:18.54 | jagster` | yeah 10.* is natted usually |
| 21:18.57 | slav3_kitten | got a 2.4ghz ubiquiti nanostation as a "modem" |
| 21:19.21 | *** join/#asterisk haryv (~Netvergen@d216-232-130-90.bchsia.telus.net) |
| 21:19.36 | haryv | cell phone service shut down in boston |
| 21:19.58 | slav3_kitten | supposedly they have fired the network engineer that i worked with to get everything setup correctly |
| 21:20.03 | haryv | good idea dont want remote detonation of bombs. |
| 21:20.17 | haryv | slave whats up |
| 21:22.21 | Rahail | question how do you round robin sip peer |
| 21:25.47 | haryv | you mean, ring one phone then the next and next? |
| 21:26.14 | Rahail | no like letsay this call ring phone 1 |
| 21:26.18 | Rahail | next call come ring phone 2 |
| 21:26.24 | Rahail | next call come ring 3 |
| 21:27.09 | haryv | sounds like a call center setup |
| 21:27.49 | Rahail | we have 5 people answering our incomign calls |
| 21:28.10 | Rahail | we are not doing any que etc call just comes and we want this call do rectired randomly |
| 21:28.14 | Rahail | redirect |
| 21:28.15 | *** join/#asterisk slav3_kitten (~frankthet@unaffiliated/slav3-kitten/x-0866809) |
| 21:28.33 | Rahail | i try using usern sipuser group=1 |
| 21:28.40 | Rahail | then on exten i did sip/r1 |
| 21:28.44 | slav3_kitten | sup haryv |
| 21:28.46 | Rahail | that didnt work |
| 21:28.47 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 21:28.50 | slav3_kitten | just fucking around with my is |
| 21:28.51 | slav3_kitten | p |
| 21:28.52 | haryv | http://forums.asterisk.org/viewtopic.php?f=1&t=8167 |
| 21:29.29 | slav3_kitten | yea apparently my IP should be a .138 instead it's a .25 |
| 21:29.44 | slav3_kitten | they are going to have to do a whole system restart in the morning |
| 21:30.07 | haryv | how many seats in your company |
| 21:30.32 | Rahail | we are not callcenter its nonprofit company |
| 21:30.46 | Rahail | local people come and help us answering phone if other have any quesiton's |
| 21:32.39 | haryv | was asking slave3 |
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| 21:35.10 | Rahail | haryv any idea how can accomplish this |
| 21:35.25 | haryv | look at the site |
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| 21:39.11 | Rahail | any other way around i dont want put them on queue |
| 21:39.41 | [TK]D-Fender | Rahail: what ARE you trying to do? |
| 21:40.06 | Rahail | when ever incoming call comes i want this round robin to different sip peer |
| 21:40.16 | Rahail | letsay call 1 comes goes to sip per 2 |
| 21:40.20 | Rahail | next call sip per 4 |
| 21:40.34 | Rahail | something like that so that way it do not hit sim sip peer |
| 21:40.35 | igcewieling | Rahail: you cannot do what you want to do without complex dialplan code unless you use a qeueu |
| 21:40.41 | [TK]D-Fender | Rahail: that IS what a queue does already. |
| 21:40.57 | [TK]D-Fender | Rahail: So why DON'T you want a queue? |
| 21:40.58 | Rahail | i have lackof knowledge this why i didnt want mass with queue |
| 21:41.22 | Rahail | learning sip and little asterisk was lot challanges for me :) |
| 21:41.38 | [TK]D-Fender | Rahail: Set up a queue. This ain't Raw-Cat Sigh Hence |
| 21:41.57 | [TK]D-Fender | Rahail: Everything else is HARDER |
| 21:42.17 | Rahail | [TK]D-Fender under each sip peer i need to add queue |
| 21:42.25 | [TK]D-Fender | Rahail: No. |
| 21:42.27 | *** part/#asterisk igcewieling (~igcewieli@ip98-183-25-31.pn.at.cox.net) |
| 21:42.28 | Rahail | then redirect the call to queue group |
| 21:42.39 | [TK]D-Fender | Rahail: "core show application queue" <- and go read the sample config |
| 21:42.54 | [TK]D-Fender | Rahail: Things don't "belong" to "sip" |
| 21:43.02 | [TK]D-Fender | Rahail: Your devices are not the center of the universe |
| 21:43.16 | [TK]D-Fender | Rahail: phones are the END of the chain and the LEAST central |
| 21:43.27 | [TK]D-Fender | Rahail: Everything is in the dialplanw hich includes the Queue() application |
| 21:43.38 | Rahail | [TK]D-Fender do you think if you can give me live example and i can use this as refrence |
| 21:43.51 | [TK]D-Fender | Rahail: Read the sample configs |
| 21:48.40 | Rahail | i am reading it |
| 21:48.45 | Rahail | but my head is going what what... |
| 21:49.27 | [TK]D-Fender | Rahail: define your timeouts Define your members. Set if people can enter the queue with no-one there to answer. |
| 21:49.47 | [TK]D-Fender | Rahail: This is ONE tiny config file and just dumping the callers into the queue app. |
| 21:49.52 | [TK]D-Fender | Rahail: This is not complicated |
| 21:51.03 | [TK]D-Fender | Rahail: The alternative is YOU doing al the work of trying to create a distribution group and tracking your agents call-out position yourself which can only do worse job with a whole lot more work and nothing to gain. |
| 21:52.25 | Rahail | exten => _X.,1,NoOp() |
| 21:52.26 | Rahail | exten => _X.,n,Set(STRATEGY=RANDOM) |
| 21:52.26 | Rahail | ^M |
| 21:52.26 | Rahail | exten => _X.,n,Dial(SIP/${IAXVAR(route)}) |
| 21:52.29 | Rahail | some one gave me this |
| 21:52.35 | Rahail | something like this what will happen |
| 21:52.59 | [TK]D-Fender | Rahail: That will not track the order for multiple callers |
| 21:53.40 | [TK]D-Fender | Rahail: And is no such thing as this "strategy" var you are referencing. |
| 21:54.10 | [TK]D-Fender | Rahail: Stop wasting your time with it |
| 21:56.27 | Rahail | let me read more about the queue |
| 21:56.34 | Rahail | and i am sure i will come back to bother you again and again |
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| 22:23.17 | chris_n | is there a way to get at the RPID via the dialplan (assuming it is set)? |
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| 22:25.14 | chris_n | for inbound calls via a sip trunk, I should add |
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| 22:28.07 | [TK]D-Fender | chris_n: "core show function SIP_HEADER" |
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| 22:32.44 | chris_n | tnx [TK]D-Fender |
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| 23:59.57 | Rahail | [TK]D-Fender I couldnt figerout yet |