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| 00:11.59 | *** part/#asterisk Husite (Hus@66.212.152.147) |
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| 00:25.18 | kaushal | Hi |
| 00:25.46 | kaushal | Can we have two instances of asterisk running on the same box ? |
| 00:26.27 | WIMPy | Sure, but there's some limitations, off course. |
| 00:26.45 | kaushal | ok |
| 00:29.28 | kaushal | WIMPy: Any docs or wiki page ? |
| 00:30.02 | WIMPy | Nothing special, just as with any software. |
| 00:30.43 | WIMPy | You can't use the same ports, not the same hardware and you have to keep some dirs seprerate to each copy. |
| 00:33.47 | kaushal | ok |
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| 00:41.25 | kaushal | WIMPy: As per http://www.voip-info.org/wiki/view/Asterisk+administration |
| 00:41.39 | kaushal | Consider to not use mpg123 |
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| 00:41.54 | kaushal | I am using mpg123 player for Outbound Campaign |
| 00:42.07 | kaushal | Do i need to use any other player ? |
| 00:42.38 | WIMPy | It makes more sense to convert it once than for each call. |
| 00:44.38 | kaushal | ok |
| 00:44.59 | kaushal | WIMPy: still not clearly understood |
| 00:45.45 | WIMPy | What? |
| 00:46.07 | kaushal | It makes more sense to convert it once than for each call. |
| 00:47.02 | WIMPy | You have to send alaw or ulaw down your pri. If you play an mp3 it has to be converted for each and every call. |
| 00:47.37 | WIMPy | If you convert it before and play the .alaw or .ulaw file, it only has to be sent. No calculation needed. |
| 00:50.47 | kaushal | ok |
| 00:50.53 | kaushal | WIMPy: Thanks |
| 00:51.26 | kaushal | so using mpg123 i need to firts convert mp3 into alaw or ulaw and then play the campaign ? |
| 00:51.32 | kaushal | first* |
| 00:52.00 | *** join/#asterisk principal93 (46bbb295@gateway/web/freenode/ip.70.187.178.149) |
| 00:52.08 | principal93 | hello |
| 00:52.09 | principal93 | i nee dsome help |
| 00:52.15 | principal93 | i installed asterisknow |
| 00:52.18 | principal93 | but when it boots up its in command line mode |
| 00:52.20 | principal93 | i want a gui |
| 00:52.33 | principal93 | i tried installing this 4 times and it does it over and over again |
| 00:53.01 | kaushal | principal93: please discuss it in #asterisknow |
| 00:53.11 | principal93 | what do we discuss in this chan? |
| 00:53.19 | kaushal | about your issue |
| 00:53.37 | kaushal | WIMPy: am i understanding you correctly ? |
| 00:54.02 | WIMPy | kaushal: That would save you CPU time. |
| 00:54.31 | kaushal | ok |
| 00:55.37 | WIMPy | principal93: Here we usally only discuss Asterisk. Most of us don't know about the GUIs. |
| 00:56.37 | kaushal | WIMPy: so i need to use some third party tools to convert mp3 to .alaw or .ulaw ? |
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| 00:57.18 | kaushal | and make it available to asterisk |
| 00:57.34 | WIMPy | You could use sox, but you can also do it from *CLI. |
| 00:57.42 | kaushal | as per http://www.voip-info.org/wiki/view/Convert+WAV+audio+files+for+use+in+Asterisk |
| 00:57.51 | kaushal | ok |
| 00:58.06 | WIMPy | 'file convert'... |
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| 00:58.58 | kaushal | WIMPy: my dialer use mpg123 currently |
| 00:59.10 | kaushal | so you say use sox instead of mpg123 |
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| 00:59.38 | WIMPy | No I say convert it before playing it. |
| 00:59.44 | kaushal | ok |
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| 02:24.39 | CoffeeIV | Hi guys, I'm having problems with Asterisk crashing. I've captured cores and filed an issue with backtraces: https://issues.asterisk.org/jira/browse/ASTERISK-18349 I'm interested in paying someone to take a look at this, and I'm located in Austin, Texas. Any recommendations ? |
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| 05:01.26 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.0.0-beta1 (2011/07/22), 1.8.5.0 (2011/07/11), dahdi-linux 2.4.1.2 (2011/04/11), dahdi-tools 2.4.1 (2011/03/03), libpri 1.4.12 (2011/07/06) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
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| 07:11.40 | ixyd_ | hi guys i have maxexpiry=180 in sip.conf and plugged off my phone since 5min... so now when i do "sip show peer X" i see "Expire : -1" and "Addr->IP : (null)" because the peer is offline which makes sense to me... but when i do "core show hint X" the sip-devstate of the peer seems to be idle which makes no sense?! do you have any idea for this? or is this a bug in the devstate system?! (see http://pastebin.com/ngtcc8uh for cli output) |
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| 08:08.15 | Moomoc | I have a simple context in a dialplan, where every internal telephone can reach each other. Unfortunately, the invalid rule (i) doesn't work and I'm not sure why. Can anyone have a look at http://nopaste.info/97c35b384d.html ? No matter if I call a 3 digit oder a 4 digit number, always the i-rule isn't executed. |
| 08:11.29 | wdoekes2 | easy. _XXX matches _XXX[anything] |
| 08:11.50 | wdoekes2 | so one- and two-digit numbers will land at i |
| 08:12.31 | wdoekes2 | you could add something like _XXXX,1,Goto(i,1) |
| 08:13.00 | ixyd_ | isnt X only matching for single digits? |
| 08:13.34 | wdoekes2 | yes.. but it doesn't say anything about trailing stuff |
| 08:14.27 | wdoekes2 | alternatively, you could do a REGEX check at prio 1, or a LENGTH check, whichever you prefer |
| 08:14.46 | Moomoc | ok, I'll do a length check |
| 08:15.07 | Moomoc | But, the invalid-rule isn't even executed when I dial e. g. 9999 |
| 08:15.33 | wdoekes2 | after adding _XXXX,1,Goto(i,1) ? |
| 08:15.46 | wdoekes2 | or before, when _XXX does match 9999 and 99999999 and 99999999999999999999 |
| 08:17.34 | Moomoc | ah, ok, thanks :) |
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| 08:23.54 | zkn | Hi, i have set up 2 ITSP's SIP trunks, and for one of them moh is not passed through where as for the other trunk everything works just fine, also all sorts of sound prompts e.g. IVR's etc work for both trunks, but for some reason music on hold is played only for one trunk. how steps should I take to troubleshoot this issue? |
| 08:24.19 | zkn | how=what |
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| 08:29.24 | singler | Moomoc, wdoekes2: _XXX will only match exten of 3 digits |
| 08:29.58 | singler | for more you could use _XXX. |
| 08:30.11 | singler | note the dot at the end, it will match 4 and more |
| 08:31.33 | Tim_Toady | Moomoc: it doesn match i cause the dialplan is wrong, u are using 'same' the wrong way |
| 08:31.50 | Tim_Toady | it must be like same => n,dostuff |
| 08:32.03 | Tim_Toady | not same => i,n,dostuff |
| 08:33.18 | Tim_Toady | and yes _XXX only matches 3 digit numbers, ur pattern is correct |
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| 08:35.02 | Moomoc | Tim_Toady: Ah, yes. I somehow overlooked this :-) |
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| 08:38.19 | Moomoc | singler: Thanks |
| 08:39.43 | merlin8282 | Hi all. I want to configure our asterisk to add ZRTP support to it, so our people can connect to it from the internet in a secure way. The asterisk would work as a trusted mitm. Do you know if this is possible, and if there is no copyright / licensing restrictions ? |
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| 08:43.37 | wdoekes2 | oh, I stand corrected, sorry :) |
| 08:45.37 | Tim_Toady | merlin8282: asterisk 1.8 suppots sip over tls and srtp for secure media, isnt this suitable for u? it s propably a lot easier to setup than zrtp since it requires no patching etc |
| 08:47.37 | merlin8282 | Tim_Toady: this would be sufficient, indeed. Do you know a place where I can find some doc for setting this up ? |
| 08:50.48 | ollii | google is your friend ;) http://www.voip-info.org/wiki/view/Asterisk+SRTP |
| 08:51.50 | ollii | more detailed: (at the bottom of voip-info wiki) |
| 08:51.51 | ollii | https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial# |
| 08:52.01 | ollii | srtp is "useless" without tls |
| 08:52.19 | merlin8282 | ollii: ok, thank you ;) . |
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| 09:03.37 | makmak78 | has anyone knowledge of Oreka opensource recordingserver? |
| 09:04.21 | makmak78 | if so could someone help me build https://bitbucket.org/iulianu/oreka/src |
| 09:04.26 | makmak78 | to run on windows |
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| 09:24.09 | makmak78 | anyone? |
| 09:26.28 | Tim_Toady | makmak78: as the readme says: "I you want to get going fast, get Windows or debian Linux binary packages from the project webpage." |
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| 10:07.06 | makmak78 | Tim_toady: in the precompiled package they seemed to have removed option to record based on X-record:1 Sip header |
| 10:07.22 | makmak78 | thats why i need someone to compile the other package for me |
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| 10:25.17 | kaldemar | makmak78: http://oreka.sourceforge.net/oreka-documentation.html |
| 10:26.07 | kaldemar | this channel is probably not the best place to ask someone to compile a program for you for windows, especially when the software is not asterisk. |
| 10:32.12 | ixyd_ | can someone help me understanding the f() option of the dial application? i dont understand the help-text from 'core show application dial' :( or maybe show an example? |
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| 10:33.01 | awk | hmm, help.. 1.8 asterisk I have a stuck channel, hangup channel doesn't work... Introstat*CLI> hangup request SIP/7796-0000000f |
| 10:33.01 | awk | Requested Hangup on channel 'SIP/7796-0000000f' |
| 10:33.08 | awk | But the channel stays alive.. |
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| 10:39.39 | makmak78 | kaldemar: it is the closest match i could find in the irc world |
| 10:43.41 | awk | channel request doesn't work either.. any suggestions to get rid of this channel, I cannot bounce asterisk |
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| 11:09.34 | awk | anyone? why when ever i ask a question I never get a response yet I help people all the time with their questions... |
| 11:09.38 | awk | seems a little unfair. |
| 11:10.29 | Rienzilla | awk: without having looked at your questions in the past, maybe you ask your questions incorrectly? |
| 11:10.46 | Rienzilla | (google for the smart questions howto, it may help) |
| 11:10.51 | awk | well my question is simple, I have a stuck channel, hangup and channel request doesn't hang it up... |
| 11:10.56 | awk | not sure how else to word that? |
| 11:11.00 | ixyd_ | is chan_sip responsive at all? |
| 11:11.16 | ixyd_ | sometimes i had chan_sip hanging at all....even sip show peers didnt work anymore |
| 11:11.26 | awk | ye, i'm getting new calls all the time.. asterisk 1.8.6 I know 1.8.5 had a bug passing incorrect response back in its agi |
| 11:11.40 | awk | but this is different ... |
| 11:11.43 | ixyd_ | hmm |
| 11:11.59 | awk | is there any other way to kill a channel other than hangup and channel request? |
| 11:12.03 | wdoekes2 | awk: I've had hanging channels.. esp. fax channels |
| 11:12.22 | wdoekes2 | I haven't found a way to kill them except for a (hard) asterisk restart |
| 11:12.34 | awk | dam, guess I must expect that from an early 1.8 release? |
| 11:12.43 | wdoekes2 | this was on 1.6.2 |
| 11:13.32 | awk | I skipped the 1.6 brach |
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| 11:14.48 | awk | I'm also geting a slin to alaw transcoding issue, [2011-08-29 13:13:28] WARNING[9646]: chan_sip.c:6343 sip_write: Asked to transmit frame type slin, while native formats is 0x8 (alaw) read/write = 0x8 (alaw)/0x8 (alaw) |
| 11:14.51 | awk | anyone get this? |
| 11:15.07 | awk | Everytime that happens I get no audio |
| 11:15.24 | awk | It happens maybe 6-10 times a day |
| 11:20.29 | wdoekes2 | I get that with faxes ;) -- I don't recall whether I only got those for failed faxes. (note that I was using local channels.. which may or may not apply to you) |
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| 11:35.08 | awk | blah, found a lovely bug in asterisk, where it pushes load right throug the roof... [2011-08-29 13:30:56] WARNING[20924]: app_dial.c:2165 dial_exec_full: Skipping dialing interface 'SIP/7546' again since it has already been dialed |
| 11:35.12 | awk | [2011-08-29 13:31:03] WARNING[20864]: channel.c:1476 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/7512@default-local-f402;2 |
| 11:35.28 | awk | and then... [2011-08-29 13:31:04] WARNING[20864]: channel.c:1476 __ast_queue_frame: Exceptionally long voice que amd tjem tje load shoots up to 9 then comes down again |
| 11:35.41 | awk | err, and then the load shoots up... |
| 11:36.44 | awk | I don't think 1.8 is nearly ready for production, i'm going to need to try backport what I need but I can't have so many issu |
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| 11:42.12 | awk | Any of you gettin g the same? |
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| 11:49.45 | awk | https://issues.asterisk.org/view.php?id=15609 this looks like the issue but it seems to be still happening |
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| 12:01.42 | merlin8282 | argh ! I must be doing something wrong but I don't see what :/ |
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| 12:01.59 | merlin8282 | I'm configuring TLS for SIP like here https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial# |
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| 12:03.01 | merlin8282 | tlsenable=yes, tlsbindaddr=0.0.0.0, tlscertfile=/etc/asterisk/keys/asterisk.pem, tlscafile=/etc/asterisk/keys/ca.crt, tlscipher=ALL and tlsclientmethod=tlsv1 in sip.conf, but netstat -tanupe shows only ports 5060 and 4569 open, no 5061. |
| 12:03.41 | merlin8282 | I've generated the certificate and keys, of course, and they're at the right place in /etc/asterisk/keys/. |
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| 12:15.17 | fazendeiro | hello guys! |
| 12:15.48 | WIMPy | merlin8282: what does sip reload tell you? |
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| 12:16.44 | fazendeiro | i'm developing a pjsip based softphone. but i'm receving this message: "Unable to set auth for tdta0x81c0c68: can not find credential for asterisk/Digest" |
| 12:17.23 | fazendeiro | has the asterisk a default realm? i'm configuring "asterisk" as realm in the softphone. |
| 12:21.51 | merlin8282 | WIMPy: nothing but "Reloading SIP", "> doing dnsmgr_lookup for '...'" and "> ast_get_srv: SRV lookup for '...' mapped to host ..., port 5060" |
| 12:23.52 | merlin8282 | mmm... TLS SIP Bindaddress: 0.0.0.0:0 |
| 12:24.16 | merlin8282 | WIMPy: thanks, now I see errors :) |
| 12:24.26 | merlin8282 | investigates the problem |
| 12:26.21 | merlin8282 | ok, was permission problem on the cert file -_- |
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| 12:38.34 | ollii | chmod 600 ? |
| 12:38.35 | ollii | ;9 |
| 12:39.07 | merlin8282 | ollii: chown asterisk :p |
| 12:39.15 | merlin8282 | and yes, 600 |
| 12:39.16 | ollii | i spended almost 1,5h hours with hylafax to notice that there only was 660 set for hosts.hfaxd ;) |
| 12:39.23 | ollii | yeah.. |
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| 12:53.58 | OlafsenM | hi guys |
| 12:54.15 | OlafsenM | anybody here... not sleeping? :D |
| 12:54.29 | OlafsenM | @question ? |
| 12:54.29 | OlafsenM | :d |
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| 12:55.37 | WIMPy | ~ask |
| 12:55.37 | 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. |
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| 12:56.30 | OlafsenM | Is there a way I can originate a call through AMI, but get initiate PBX while channel is in PROGRESS? |
| 12:56.59 | OlafsenM | I want to pick up early media, but I can't do that since Originate action starts running PBX on UP state |
| 12:57.02 | OlafsenM | any ideas? |
| 12:57.31 | WIMPy | Exchange the two channels? |
| 12:57.51 | OlafsenM | explain please |
| 12:57.59 | OlafsenM | exchange how |
| 12:58.00 | OlafsenM | ? |
| 12:58.14 | WIMPy | Swap the two arguments. |
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| 12:59.08 | OlafsenM | what two arguments? |
| 12:59.36 | WIMPy | The ones you pass to originate. |
| 13:02.41 | OlafsenM | there aren't two channels |
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| 13:08.55 | WIMPy | Every call has two ends. |
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| 13:23.29 | OlafsenM | WIMPy: ok, can you give me an example? Because I don't see which two parameters to exchange |
| 13:24.13 | OlafsenM | WIMPy: http://pastebin.com/QADWkpq2 |
| 13:25.01 | awk | <syndi> lots of hacking required this shit all works out the box |
| 13:25.16 | awk | sorry, right mouse button |
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| 13:55.12 | Qwell | is having some of this this morning. mmm. http://store.digium.com/productview.php?product_code=810-00038 |
| 13:55.35 | Qwell | We now return you to your regularly scheduled programming. |
| 13:56.55 | WIMPy | OlafsenM: You specify the 1st channel as channel and the 2nd channel as extension or application. |
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| 14:14.30 | anonymouz666 | JonathanRose: the crash in res_jabber in 1.8 is related to iks_send_raw () from /usr/lib64/libiksemel.so.3 ? |
| 14:15.05 | JonathanRose | Are you talking about a crash or are you talking about a failure to compile? |
| 14:15.13 | anonymouz666 | about a crash |
| 14:16.12 | JonathanRose | So this would be about ASTERISK-18078 then? |
| 14:16.21 | JonathanRose | If so, yeah, that crash was related to iks_send_raw |
| 14:16.46 | JonathanRose | Basically if your connection to the jabber server dropped and you had a change of device state, segfault. |
| 14:17.00 | anonymouz666 | oh yeah, exactly that. |
| 14:17.02 | JonathanRose | well, if you were publishing those events anyway. |
| 14:17.20 | anonymouz666 | I am stuck in distributed device state |
| 14:17.34 | anonymouz666 | res_ais is deadlocking and my option with xmpp is also crashing |
| 14:17.44 | anonymouz666 | there's no place to run :) |
| 14:17.56 | JonathanRose | I never could figure out how to actually do anything with the published device states, but since I managed to actually get publishing working, I was able to test the area I needed to test :P |
| 14:18.16 | JonathanRose | As much as I'm still frustrated >_< |
| 14:18.33 | leifmadsen | JonathanRose: ya it's pretty much "automatic" -- what you'd want to be looking at would be something like voicemail notifications being sent to a phone registered to another server opposite of the voicemail server |
| 14:19.34 | JonathanRose | I was under the impression that if I had a custom device state set on two asterisk boxes both connected to the same Jabber server with pubsubnode and I published the events that I'd have those hints change on the other server if I changed it on the first. |
| 14:19.35 | anonymouz666 | big callcenters setup the distributed device state is needed, unless you want to build all queues in just one machine, and then you have to expect weird things because 200 active channels is a lot for asterisk to handle |
| 14:19.57 | JonathanRose | And that didn't happen, so I figure either it's broken or I did something wrong, and of course I'm leaning towards the later. |
| 14:21.21 | anonymouz666 | openais + res_ais is better in my opinion than xmpp stuff |
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| 14:21.55 | leifmadsen | anonymouz666: they are different for different situations |
| 14:22.10 | leifmadsen | I know someone who is using the XMPP stuff heavily |
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| 14:22.42 | anonymouz666 | leifmadsen: it works because the connection to the jabber server does not drop :) |
| 14:23.25 | anonymouz666 | well there's a patch that I can test |
| 14:23.29 | JonathanRose | Well, it shouldn't explode now even if the jabber server drops with fresh checkouts at least. Though someone was saying it wouldn't build for them in dev-mode... and I have no idea why that is. |
| 14:23.50 | JonathanRose | Well, that's not true, I have a loose idea. |
| 14:24.20 | JonathanRose | But the point is, it builds in dev mode on both of my machines and on what seems to be all of our testsuite machines. |
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| 14:26.25 | anonymouz666 | JonathanRose: this patch will be in 1.8.7.0-rc ? |
| 14:26.42 | JonathanRose | I'd assume. |
| 14:26.49 | pabelanger | which? |
| 14:27.42 | pabelanger | JonathanRose: ^ |
| 14:27.55 | pabelanger | we can tag it in JIRA to be sure it is :) |
| 14:27.56 | anonymouz666 | pabelanger: 18078 |
| 14:28.10 | JonathanRose | yeah, it was on this sprint... |
| 14:28.55 | anonymouz666 | JonathanRose: if you like distributed device state so much, there's also the 18318 :-) |
| 14:28.59 | leifmadsen | if it was committed, then it would be in the next rc1 |
| 14:29.32 | anonymouz666 | I am really waiting the sipstorecause option commit |
| 14:30.00 | leifmadsen | it's not already in? |
| 14:30.04 | JonathanRose | anonymouz666: In case you couldn't tell, 'like' isn't quite my relationship so far with distributed device state :P |
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| 14:30.25 | anonymouz666 | :P |
| 14:30.43 | JonathanRose | But I wouldn't be surprised if I end up with it anyway since I've been working with it lately. |
| 14:31.11 | anonymouz666 | it is easy to reproduce and very easy to setup also |
| 14:31.22 | merlin8282 | mmm, does anyone know a SIP softphone that supports TLS *and* that can be used behind a NAT ? Blink does not seem to support NAT traversal... |
| 14:33.00 | anonymouz666 | merlin8282: strange question, is there a phone that can't be used behind a nat? |
| 14:33.08 | anonymouz666 | asterisk will take care of it |
| 14:33.24 | anonymouz666 | sending the signalling through rport |
| 14:33.27 | anonymouz666 | and symmetric rtp |
| 14:33.29 | JonathanRose | Maybe he's trying to dial out of a machine behind a NAT to an Asterisk box behind a NAT or something? |
| 14:33.59 | JonathanRose | I don't know. I haven't worked a lot with NATs :P |
| 14:34.05 | merlin8282 | anonymouz666: blink can connect to the asterisk (which has a public IP) from behind a NAT, but the asterisk can't connect back to blink, because it sends a class C PI address to asterisk... |
| 14:34.23 | leifmadsen | configuration issue |
| 14:34.24 | leifmadsen | nat=yes |
| 14:34.32 | leifmadsen | asterisk will ignore it and send to the external IP if you use that |
| 14:34.34 | merlin8282 | leifmadsen: even if the asterisk is not behind a nat ? |
| 14:34.42 | leifmadsen | it == non-routable IP |
| 14:34.43 | merlin8282 | ah ! |
| 14:34.48 | leifmadsen | yes |
| 14:34.53 | leifmadsen | that's the purpose of nat=yes :) |
| 14:34.57 | leifmadsen | for devices behind NAT |
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| 14:35.08 | merlin8282 | I always thought that this setting was about asterisk being behind a NAT... |
| 14:35.12 | leifmadsen | no |
| 14:35.19 | leifmadsen | that's localnet and externhost |
| 14:35.35 | merlin8282 | okay ! I try it and thank you leifmadsen ;) |
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| 14:37.05 | leifmadsen | anonymouz666: ASTERISK-18318 you referenced looks like you're using res_timing_pthread |
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| 14:37.19 | merlin8282 | leifmadsen: and I suppose that I can set nat=yes for all SIP clients ? I mean, it will work even if the client is not behind a NAT ? |
| 14:37.21 | leifmadsen | anonymouz666: that can definitely cause issues that look like deadlocks, so you shouldn't use that |
| 14:37.43 | leifmadsen | merlin8282: in most cases yes, but i've seen it in some situations cause issues (but I pretty much set it all the time anyways) |
| 14:37.52 | leifmadsen | if you run into an issue, then turn it off ;) |
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| 14:41.30 | merlin8282 | ok ! /window 9 |
| 14:41.34 | merlin8282 | oops |
| 14:43.05 | anonymouz666 | leifmadsen: am I using? it's just loaded like I see with modules show like res_timing_pthread.so, but I didn't configure anything to use it. just by the fact it is loaded it means the modules are using it? |
| 14:44.02 | leifmadsen | well I see it in the backtrace, so I'd make sure it's not loaded at all. There is a priority list for the timing modules if they are loaded and available |
| 14:44.06 | leifmadsen | I'd avoid compiling or loading it |
| 14:44.40 | anonymouz666 | oh yes, thanks for the information will put this module in noload list. |
| 14:45.07 | leifmadsen | let me know if you run into any additional issues after that, but certainly the first thing is to avoid that module |
| 14:47.03 | anonymouz666 | yes, will test this right anow |
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| 14:56.34 | anonymouz666 | it seems everything happens in a big callcenter setup |
| 14:57.02 | anonymouz666 | from people enabling the DND button and people hanging up calls by clicking in the red button (x-lite) |
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| 15:12.33 | psilikon | So the 'Random' application is not in 1.8? So I should just use the RAND function to do what I need to do? |
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| 15:12.51 | anonymouz666 | yes |
| 15:13.15 | anonymouz666 | I use it a lot in a test enviroment |
| 15:13.31 | psilikon | anonymouz666, sweet. Way more fun that way. |
| 15:13.48 | psilikon | I wonder why they got rid of it. |
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| 15:15.32 | JonathanRose | There was a big move to change a lot of applications into functions a while back. |
| 15:16.02 | JonathanRose | Functions are nicer because you can use them as arguments to other applications or as a standalone thing with Set |
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| 15:16.39 | psilikon | JonathanRose, I see. Thank you very much. |
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| 15:18.57 | p3nguin | How would copying callerid from a channel that is calling a phone break isdn or dahdi if the phone forwards the call to another number? https://sourceforge.net/tracker/?func=detail&atid=917048&aid=3398327&group_id=186378 |
| 15:20.10 | WIMPy | doesn't understand the question. |
| 15:20.29 | p3nguin | I thought that might be the case, so I posted the link to the issue filed. |
| 15:21.46 | WIMPy | Shouldn't both callerids be present ideally? |
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| 15:22.16 | p3nguin | Phones only display one caller id number at a time. |
| 15:22.27 | p3nguin | And it should be that of the person calling. |
| 15:22.39 | WIMPy | Mine will display both. |
| 15:22.44 | p3nguin | How? |
| 15:22.59 | SunTsu | display large enough to do so? |
| 15:23.05 | WIMPy | Probably not since they are connected to Asterisk, but on the PSTN they did. |
| 15:23.24 | WIMPy | yes |
| 15:23.46 | p3nguin | Must be some fancy systems you have over there. I have never seen a single phone here that will display more than one caller id number at a time. |
| 15:24.07 | p3nguin | Not phones, and not callid id boxes. |
| 15:25.58 | p3nguin | So, with that in mind, the caller id number on a ringing phone should be that of the caller, not a middle-man phone which has call forwarding enabled. |
| 15:26.04 | WIMPy | Well, it's quite new. It first apperaed in the early 90s. |
| 15:26.05 | p3nguin | SIP doesn't have any problem doing it correctly. |
| 15:26.38 | p3nguin | I don't have any IAX2 phones, but I'd bet IAX2 does it correctly, as well. |
| 15:26.48 | WIMPy | On a forwarded call there can be quite a lot of callerids sent back and forth. |
| 15:27.08 | p3nguin | But on a phone that has only one caller id number slot, it should show that of the caller. |
| 15:27.43 | WIMPy | Which can only work if Asterisk does the forwarding. |
| 15:27.55 | anonymouz666 | leifmadsen: bad news, still happens even with noload, will attach the new debug info. |
| 15:28.39 | p3nguin | If my SIP phone has call forwarding enabled by pressing the cfwd button, would you say asterisk will be doing the forwarding? or is the phone doing the forwarding, since the phone is the device sending the redirect message? |
| 15:28.59 | p3nguin | I'd say it's the phone doing the forwarding. |
| 15:29.04 | anonymouz666 | indeed |
| 15:29.31 | p3nguin | If I use the cfwd button on my SIP phone, the correct caller id number shows up on the destination. |
| 15:29.58 | p3nguin | Not the cid number of my phone, but the cid number of the actual caller. |
| 15:30.19 | p3nguin | My phone isn't making the call, so its cid need not be shown. |
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| 15:31.50 | WIMPy | It is the phone, but Asterisk needs to know, otherwise it would use the configured callerid for the new call leg. |
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| 15:32.27 | WIMPy | just wonders what happens on multiply forwarded calls. Do you get one ID per forwarding? |
| 15:32.32 | p3nguin | There will be a SIP message from the phone. |
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| 15:32.46 | WIMPy | And what if the original call already had two IDs? |
| 15:33.01 | p3nguin | That doesn't happen here, so I wouldn't know. |
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| 15:33.50 | p3nguin | If multiple calls are made to the phone which does a SIP reflection/redirection, every call will be forwarded to the configured destination, and every call will have the original callerid number on it showing on the destination phone. |
| 15:34.29 | p3nguin | But I'm not willing to stop using sccp and go to sip just to fix one broken bit. |
| 15:35.19 | p3nguin | And I have problems believing that no one else ever had a complaint about broken caller id from chan_sccp before. |
| 15:36.03 | p3nguin | s/id /id on call forwarding / |
| 15:37.41 | WIMPy | Why not let Asterisk do the forwarding? That makes more sense anyway. |
| 15:38.05 | p3nguin | It doesn't make sense to me. There's a big button on phones for that. |
| 15:38.30 | p3nguin | You press the forward button on the phone, then enter the number where to forward calls. Done. |
| 15:38.43 | WIMPy | Tell you phone to tell Asterisk when it is pressed. |
| 15:38.51 | p3nguin | When you're done forwarding, press the button again to clear it. |
| 15:39.05 | WIMPy | Those were the days... |
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| 15:40.09 | p3nguin | If I used SIP, I wouldn't have to work AROUND a channel driver problem, because it works correctly in SIP already. |
| 15:41.03 | WIMPy | It's open source. You can fix it :-) |
| 15:41.08 | p3nguin | So... back to my original question: how would passing the correct caller id number on a forwarded call break isdn or dahdi? |
| 15:41.29 | WIMPy | I still don't understand that question. |
| 15:42.10 | WIMPy | Like sending out a callerid on your isdn line you aren't allowed to send? |
| 15:42.25 | p3nguin | That's the reason for not fixing it. The guy said it will cause problems with idsn, dahdi, and lcr if he passes the original caller id on forward rather than the caller id of the phone itself. |
| 15:43.00 | p3nguin | If the phone is not making a call, it should not be providing its own caller id number to a destination. |
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| 15:43.09 | p3nguin | And if the phone is doing a forward, it is not making a call. |
| 15:43.19 | p3nguin | It seems very clear to me. |
| 15:43.22 | WIMPy | If you send an invalid callerid, the switch will replace it with your default ID. |
| 15:43.38 | WIMPy | Usually. |
| 15:43.51 | p3nguin | Yeah, so send the incoming caller id number, which is already valid. |
| 15:44.07 | WIMPy | In American ISDN there seems to be the possibility that it will make the call fail. |
| 15:44.19 | p3nguin | To use a valid callerid? |
| 15:44.19 | WIMPy | It is not valid for your line. |
| 15:44.52 | p3nguin | Sending a callerid of 762 (the cid num of my phone) would do the same thing, would it not? |
| 15:45.00 | WIMPy | The ting is that you aren't forwarding. You are placing a new call. |
| 15:45.17 | WIMPy | That would certainly be invalid, yes. |
| 15:46.08 | p3nguin | I'm going to go back to the example of SIP. It's not a new call in SIP; the phone *redirects* the call somewhere else, and the caller's CID is used rather than the SIP phone's CID. |
| 15:46.16 | p3nguin | Why can't chan_sccp do it the same way? |
| 15:46.38 | WIMPy | NFI |
| 15:47.07 | p3nguin | Asterisk is able to understand a redirect message. The channel driver just has to provide it. |
| 15:47.37 | WIMPy | But I think bristuff was the only ISDN thing that supported forwarding. But you couldn't use that if the forwarding is done on the phone anyway. |
| 15:47.58 | CoffeeIV | Hi guys, I'm having problems with Asterisk crashing. I've captured cores and filed an issue with backtraces: https://issues.asterisk.org/jira/browse/ASTERISK-18349 I'm interested in paying someone to take a look at this, and I'm located in Austin, Texas. Any recommendations ? |
| 15:48.04 | p3nguin | Why would ISDN care of a phone forwards a call? |
| 15:48.16 | p3nguin | s/of/if/ |
| 15:48.28 | WIMPy | Maybe the idea of forwarding forwarding requests from one channel to another doesn't exist? |
| 15:48.34 | *** join/#asterisk ruied_ (~ruied@dial-b1-225-59.telepac.pt) |
| 15:48.52 | WIMPy | If you forward the call, the original callerid would be preserved. |
| 15:49.03 | p3nguin | It should be. That's the point of callerid. |
| 15:49.19 | p3nguin | You should see the caller id of the person who will be on the phone when you answer it. |
| 15:49.24 | WIMPy | If you place another call to forward the 1st, you will send your ID, unless you have CLIP no-screening enabled. |
| 15:49.38 | p3nguin | If Jack calls me, but my phone redirects to you, your phone should say Jack is calling you. |
| 15:50.00 | p3nguin | But with chan_sccp, your phone says I am calling you, when I am NOT calling you -- Jack is. |
| 15:50.13 | beek | ping leifmadsen |
| 15:51.02 | WIMPy | That's the same what would happen if you forward to dahdi. |
| 15:51.29 | p3nguin | Not for me. Mine would work correctly if I have any control over it. |
| 15:51.41 | p3nguin | I can't stand when shit does not work like it is supposed to. |
| 15:52.06 | ruied_ | I have lcr 1.9 (tried also lcr-1.8) and asterisk 1.8.5 I can make calls to ISDN but can't receive when I dial from my mobile to ISDN number. The ISDN provider says the number is unavailable. LCR logs an error cause 16, asterisk doesn't report a thing. (http://pastebin.com/sA3kNf3Y)... |
| 15:53.08 | WIMPy | p3nguin: That's why I wouldn;t call Asterisk a PBX (toolkit). There;s lots of stuff that doesn't work as expected. |
| 15:53.22 | anonymouz666 | leifmadsen: issue updated! |
| 15:54.28 | p3nguin | I also don't see how dahdi wouldn't accept the caller id if a SIP phone forwarded a call. The caller id number would be present, and it would be the same as Dial(Dahdi/g0/somenumber). |
| 15:55.29 | WIMPy | ruied_: Looks like you don;t have a mathing extension. |
| 15:55.51 | WIMPy | p3nguin: dahdi would acept it, but the telco usually won't. |
| 15:56.02 | WIMPy | But I have to go now. |
| 15:56.32 | WIMPy | No. In 9 Minutes :-) |
| 15:56.33 | p3nguin | I'd have to get a new telco that allows me to set my own caller id numbers. |
| 15:57.13 | WIMPy | It obviouesely depends on your telco, yes. |
| 15:57.58 | WIMPy | Here you pay a tenner per month to be able to send anything. |
| 15:58.01 | p3nguin | And even if they won't accept a caller id number of a callER from outside, that also means they will not accept my phone's caller id of 762. |
| 15:58.26 | p3nguin | So we're back to not having valid caller id. |
| 15:58.42 | ruied_ | WIMPy, I have the context '[ISDN]' and 'exten => s,1,Dial(sip/300)' |
| 15:58.50 | WIMPy | If you have DDI and you set the TON to abbrevuated, it would be a valid number. |
| 15:59.05 | p3nguin | Which is just as bad as having valid caller id and having it rejected. |
| 15:59.22 | WIMPy | ruied_: You won;t get a call to s from the network. |
| 16:00.25 | WIMPy | But to sum it up: I don't see how passing the original callerid could be worse. |
| 16:01.16 | ruied_ | hmm, so I should look for an exact match of the number sended by the operator match? |
| 16:01.31 | WIMPy | ruied_: yes |
| 16:01.42 | p3nguin | Anything other than an exact match ... isn't a match. |
| 16:01.53 | WIMPy | Well, it depends on your telcos configuration, Usually you will get the full local number. |
| 16:02.02 | p3nguin | literal or pattern, it must still be a matching extension. |
| 16:02.24 | ruied_ | but shouldn't asterisk report something 'not matching' at asterisk prompt (asterisk -rvvvvvvvvvvvvvvvvvv) |
| 16:02.25 | ruied_ | ? |
| 16:03.05 | WIMPy | ruied_: No. cahn_lcr uses the simple switch. So the call won;t get to Asterisk if there's no match. |
| 16:04.16 | *** join/#asterisk ariel_ (~chatzilla@pdpc/supporter/active/abatista) |
| 16:04.37 | ruied_ | WIMPy, ah, I thought allway that it could be an error in lcr, since it didn't show up a thing in asterisk... |
| 16:04.40 | ruied_ | going to try |
| 16:05.08 | *** join/#asterisk AdvoWork (~AdvoWork@unaffiliated/advowork) |
| 16:05.09 | WIMPy | i.e. it asks if there's a possible match and only sends te call to Aaterisk if there is. |
| 16:05.21 | WIMPy | Ok. CUL |
| 16:05.23 | *** join/#asterisk Bob_Pierce (~WESTMAN\p@216.36.132.162) |
| 16:05.55 | *** join/#asterisk fenrus (fenrus@drickheroin.se) |
| 16:08.19 | _Shadowfax_ | anybody here using asterisk + dahdi for meet me on RHEL 6.1 using EPEL? i'm having a devil of a time figuring out how to load dahdi. |
| 16:08.41 | _Shadowfax_ | (on our dev/mock-prod fedora core box it loads dahdi_dummy in /etc/rc.local with mod probe...) |
| 16:08.57 | _Shadowfax_ | i mean modprobe. !@#$ autocorrect. |
| 16:09.27 | chazzam | did you try running make config in the dahdi source? |
| 16:09.32 | chazzam | then it installs init scripts... |
| 16:09.59 | _Shadowfax_ | this is installing from EPEL packages, which is what we'd prefer, so partly i'm trying to figure out if i need to build from source or not |
| 16:10.25 | *** join/#asterisk navaismo (~IceChat77@201.123.61.67) |
| 16:13.49 | psilikon | I'm using a statement like this: $[${RAND(1,100)} > 60]?machine_sim,s,1) in my dialplan which works great but I would also like the random integer to be printed to the console. Is that possible? |
| 16:14.24 | psilikon | without setting a variable and adding extra lines |
| 16:14.26 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 16:14.28 | navaismo | Use NoOp() |
| 16:14.42 | JonathanRose | NoOp will still produce the output in its final form. |
| 16:14.43 | JonathanRose | So no. |
| 16:15.02 | JonathanRose | I don't think that'll work anyway. |
| 16:15.09 | p3nguin | And to print to the console, the app is Verbose() anyway. |
| 16:15.25 | JonathanRose | Yeah, noop only does its thing with verbosity of 3+ |
| 16:16.25 | navaismo | someone use verbose <3?? |
| 16:16.26 | *** join/#asterisk merlin8282 (~merlin828@AStrasbourg-554-1-203-129.w83-203.abo.wanadoo.fr) |
| 16:16.34 | Qwell | I <3 verbose. |
| 16:16.35 | psilikon | So I will need to use 'Set' and RAND together to set the value of a variable then Verbose(${RAND_VAR_INT}) |
| 16:16.40 | chazzam | navaismo: 0 is <3 |
| 16:16.56 | p3nguin | Use Verbose() and it will print on 0, 1, and 2 as well as 3+. |
| 16:17.23 | p3nguin | But you can control which levels Verbose prints to the console, if you'd like. |
| 16:18.54 | merlin8282 | mmm, could this be a bug ? I've set transport=tls,udp for a sip account. I'm registered to the server (udp, no TLS), all works, etc. If I restart asterisk, it tries to connect to my client with TLS: |
| 16:18.58 | merlin8282 | ERROR[6904]: tcptls.c:382 ast_tcptls_client_start: Unable to connect SIP socket to my.ip.add.ress:5060: Connection refused |
| 16:19.32 | merlin8282 | so I have to disconnect/reconnect my ekiga for this error to stop flooding the asterisk console. |
| 16:20.21 | p3nguin | What is already listening on tcp 5060? |
| 16:21.08 | p3nguin | Anything? |
| 16:21.12 | p3nguin | Anything at all? |
| 16:21.36 | p3nguin | Bueller? |
| 16:30.04 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 16:30.19 | *** join/#asterisk sulex (~sulex@pdpc/supporter/professional/sulex) |
| 16:35.30 | psilikon | Often times when writing a script that interacts with Asterisk in some way I am faced with the decision to put the logic in the dialplan or my script. Is there a best practice on this? Or is it case-by-case? |
| 16:38.43 | p3nguin | Will the logic ever be used by something other than the script? |
| 16:39.02 | *** join/#asterisk paulc (~paulc@unaffiliated/paulc) |
| 16:41.34 | *** join/#asterisk BenC[UK] (~bcummins@cpc2-lock3-2-0-cust77.6-1.cable.virginmedia.com) |
| 16:41.42 | BenC[UK] | Guys, I am gettin gthis: channel.c: Channel allocation failed: Can't create alert pipe! Try increasing max file descriptors with ulimit -n |
| 16:41.50 | BenC[UK] | I have tried ulimit -n 4096 |
| 16:41.53 | BenC[UK] | but still getting it |
| 16:42.01 | BenC[UK] | do I need to restart asterisk after the ulimit? |
| 16:42.17 | p3nguin | Why do you have any ulimit at all? |
| 16:42.18 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 16:42.23 | BenC[UK] | I dont know :S |
| 16:42.28 | p3nguin | unset it. |
| 16:42.40 | BenC[UK] | can you tell me how :$ |
| 16:44.06 | BenC[UK] | was efault 1024 |
| 16:45.34 | psilikon | p3nguin, no |
| 16:46.18 | p3nguin | benc[uk]: Did you try ulimit -u unlimited? |
| 16:46.36 | p3nguin | err, wait |
| 16:46.46 | p3nguin | You said file descriptors, not processes. |
| 16:46.54 | p3nguin | benc[uk]: Did you try ulimit -f unlimited? |
| 16:46.59 | p3nguin | my mistake |
| 16:47.18 | p3nguin | Crap. -f is for file size. |
| 16:47.31 | BenC[UK] | -u is processes |
| 16:47.36 | BenC[UK] | its -n |
| 16:47.38 | p3nguin | benc[uk]: Did you try ulimit -n unlimited? |
| 16:47.41 | BenC[UK] | yes |
| 16:47.43 | BenC[UK] | just did |
| 16:47.45 | p3nguin | I had to look that up. |
| 16:47.52 | BenC[UK] | [root@DIALER ~]# ulimit -n unlimited |
| 16:47.52 | BenC[UK] | -bash: ulimit: open files: cannot modify limit: Operation not permitted |
| 16:48.31 | p3nguin | Is that a virtual machine guest? |
| 16:51.29 | BenC[UK] | yeah |
| 16:51.47 | BenC[UK] | res_timing_timerfd.c:171 timerfd_timer_ack: Read error: Bad file descriptor |
| 16:51.49 | BenC[UK] | getting this now |
| 16:51.51 | p3nguin | The host has you limited, I'd guess. |
| 16:51.51 | malcolmd | yes, you need to restart asterisk after you change the ulimit. the ulimit applies to the console you're currently logged in from. so if you set ulimit on console B and then restart Asterisk on console A, then it won't take affect. |
| 16:52.05 | BenC[UK] | ok |
| 16:54.43 | p3nguin | Is there a way to do ExecIf with Playback and a file? If the file exists, Playback(file), otherwise don't try. If ExecIf() can do it, I don't know how to write the condition. |
| 16:55.22 | p3nguin | Can I do it like I would in bash? |
| 16:56.37 | *** join/#asterisk nightrid3r (~kvirc@ip-83-134-66-91.dsl.scarlet.be) |
| 16:59.05 | Tim_Toady | p3nguin: propably by using system(), check the return value and proceed |
| 16:59.39 | Tim_Toady | im not sure if dialpan provides any file test applications |
| 17:00.28 | *** join/#asterisk irroot (~irroot@41.122.199.173) |
| 17:04.26 | *** join/#asterisk hoornet (~hoornet@93.103.24.156) |
| 17:07.59 | fenrus | set the ulimit in the initscript you use to start apache.. |
| 17:08.08 | fenrus | (im a bit late..) |
| 17:09.06 | anonymouz666 | don't use System() |
| 17:09.16 | anonymouz666 | use STAT() to check if the file exists |
| 17:09.32 | ChannelZ | I use illegal immigrants |
| 17:09.43 | p3nguin | STAT() looks like what I needed. |
| 17:09.44 | ChannelZ | Oops! Did I say that out loud? |
| 17:09.55 | navaismo | :-| |
| 17:10.17 | anonymouz666 | I went to UFC RIO |
| 17:14.48 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 17:15.02 | *** join/#asterisk The_REV (~The_REV@c-76-21-1-80.hsd1.ca.comcast.net) |
| 17:15.53 | ChannelZ | Death on twooooo leeeegggggssss! |
| 17:16.15 | irroot | cool i only have one leg mean im safe ?? |
| 17:17.24 | ChannelZ | no in fact that just makes you easier to knock over. |
| 17:17.50 | *** join/#asterisk acidfoo (~nib@modemcable094.94-70-69.static.videotron.ca) |
| 17:21.26 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 17:22.19 | *** join/#asterisk sbszulu (~dundubala@41.19.214.95) |
| 17:23.32 | leifmadsen | beek: pong? |
| 17:23.48 | p3nguin | anonymouz666: Will STAT() accept a relative file path, such as local/recording.wav, or does it need to be the full path? |
| 17:24.13 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 17:24.38 | leifmadsen | p3nguin: pretty sure it's full path |
| 17:25.06 | leifmadsen | I'm not sure what the path would be relative to |
| 17:25.41 | p3nguin | Yep, full path made it work. |
| 17:26.05 | p3nguin | I guess apps have a startpath where functions have no reason to have one. |
| 17:27.48 | p3nguin | ExecIf($[${STAT(f,/var/lib/asterisk/sounds/local/recording.wav)}],Playback,local/recording) |
| 17:27.53 | beek | leifmadsen: I had a question about bug reports... I have one that's killing me: ASTERISK-18207 |
| 17:28.22 | beek | It was reported a month ago and I was just wondering how often user reports like this are examined. |
| 17:28.42 | leifmadsen | supposed to be often, but I've been very busy with my wedding, so I have some catching up to do |
| 17:29.03 | beek | Ah... are you "hitched" now or still planning? |
| 17:29.25 | leifmadsen | I am no longer single :) |
| 17:29.31 | beek | Congrats! |
| 17:29.34 | leifmadsen | thanks :) |
| 17:29.34 | beek | (I think) |
| 17:29.37 | leifmadsen | lol |
| 17:30.47 | jaytee | it starts with "I do" and then after 10 years it's like someone installed a pull string on the back of your neck and if they pull it your head nods and you say, "Yes, dear!" |
| 17:31.58 | beek | I was told by a friend of mine that: As soon as I said "I do" I found that "I don't" |
| 17:32.03 | _Corey_ | Ours is next year and it feels like she brings up something or other new every day... wedding=exhausting |
| 17:32.34 | KavanS | meh |
| 17:32.47 | KavanS | if you're whipped before, and you're whipped afterwards IMO |
| 17:32.56 | KavanS | I haven't been married, so what do I know though? :) |
| 17:33.04 | beek | leifmadsen: Thanks! I hope that the description was clear enough. |
| 17:33.11 | navaismo | niiaaa I already 10 years with my wife |
| 17:33.18 | KavanS | I would like to believe it's a compromise for both...to each sides benefit |
| 17:33.56 | beek | Of the weddings of which I have been a part (best man, usher, etc) I'm batting about .225, which sucks even in baseball. |
| 17:34.24 | jaytee | so basically you play for the Yankees? :-) |
| 17:34.29 | beek | :D |
| 17:35.13 | jaytee | is a die hard Red Sox fan. Nothing fills me with schadenfreude so much as the Yankees losing :-) |
| 17:37.55 | *** join/#asterisk thansen (~thansen@c-67-177-32-87.hsd1.ut.comcast.net) |
| 17:38.54 | *** join/#asterisk bitbandit (~mojojo200@c-98-202-116-168.hsd1.ut.comcast.net) |
| 17:39.05 | thansen | what's the proper way to get some debug output for asterisk crashes? I'm not getting anything :( |
| 17:40.47 | p3nguin | ~backtrace |
| 17:40.47 | infobot | backtrace is a debugging tool that is useful when trying to track down asterisk crashes (use -g with asterisk to generate a core, then read doc/backtrace.txt). See also https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace |
| 17:41.45 | p3nguin | thansen: Maybe that page can help you get something useful to submit. |
| 17:42.40 | thansen | p3nguin: ok, thanks...I've had chronic 'random' crashes for years now and I guess it's finally time to get to the bottom of it |
| 17:45.20 | leifmadsen | _Corey_: it doesn't get better :) make sure you get your suits done early, and plan any bachelor parties and such early as well -- about the last 60 days she is going to get crazy and overwhelmed and if you haven't even done that there will be hell to pay :D |
| 17:45.59 | leifmadsen | _Corey_: beyond that, most of what you need to do is just drive her to places and be somewhat involved. It will help a lot later on when you need the brownie points :) |
| 17:46.51 | beek | is taking notes. |
| 17:47.41 | beek | leifmadsen: I just tidied up my description a bit to make it clearer. |
| 17:47.53 | leifmadsen | beek: awesome |
| 17:47.54 | _Corey_ | leifmadsen: Yeah, I'm inclined to try to tune it out as much as possible. I won't be in the same room with a TV playing Bridezillas or any of the other innumerable awful bridal shows... |
| 17:48.20 | *** join/#asterisk Eitan (~Eitan@adsl-99-22-192-148.dsl.lsan03.sbcglobal.net) |
| 17:48.24 | leifmadsen | _Corey_: I hear that... I usually let her watch them and put the TV on split screen and play video games :) |
| 17:48.53 | _Corey_ | I did visit one 'bridal expo'--which was a horrible experience--and have visited a couple venues with her. Lately I've just been closing my eyes and humming loudly, which seems to wokr. |
| 17:48.53 | *** join/#asterisk userty (~user@206.47.201.86) |
| 17:48.56 | _Corey_ | work even |
| 17:49.04 | beek | Hmmm... now that you've cloned the bug will those changes be propogated as well? |
| 17:49.06 | leifmadsen | that'll only work so long :) |
| 17:49.23 | leifmadsen | beek: the clone is just for internal time tracking purposes |
| 17:49.26 | beek | hands _Corey_ a bottle of Jack's |
| 17:49.30 | _Corey_ | yeah, but I'm going to enjoy it while it does... |
| 17:49.32 | leifmadsen | all work is done on the issue you filed |
| 17:49.39 | beek | Cool. |
| 17:49.50 | leifmadsen | so you'll see everything (if anything :)) being done |
| 17:50.27 | _Corey_ | beek: I've been buying Johnnie Walker by the quart... I have many strategies |
| 17:51.07 | beek | _Corey_: Please remove all sharp objects from your home and office... you don't want to be tempted. |
| 17:51.10 | _Corey_ | leifmadsen: Anyway, congrats. I definitely envy anyone who is done with it. |
| 17:51.59 | _Corey_ | beek: I just found out yesterday that the tickets I bought for a whisky tasting event conflict with another wedding next month... weddings will destroy the universe |
| 17:52.03 | leifmadsen | _Corey_: I'll tell you what all my married friend told me -- the day is going to fly by when it happens, so when it gets there, sit back and enjoy it, take it all in, because it'll be over before you know it |
| 17:52.18 | beek | And then reality will set in. |
| 17:52.22 | leifmadsen | :) |
| 17:52.29 | p3nguin | My first wedding was a traditional wedding. My second was courthouse-and-judge style. |
| 17:52.32 | _Corey_ | Yeah, hopefully someone will take some photos. I may be drunk |
| 17:52.35 | leifmadsen | I've lived with my wife for over 2 years and bought a place with her, so nothing is really different ;) |
| 17:52.52 | _Corey_ | leifmadsen: Yeah, same here |
| 17:53.31 | beek | _Corey_: so negotiate with her and tell her that all of the money that will be spent on a wedding would be put to better use on a nice long cruise together. |
| 17:53.39 | leifmadsen | _Corey_: it gets better especially after all the big things are done :) |
| 17:53.55 | leifmadsen | beek: he's too far in if the venue has been chosen :) |
| 17:54.15 | Eitan | hey guys, what is a good place to find reliable asterisk/freepbx contracters for hire? need some stuff done |
| 17:54.16 | _Corey_ | I tried the whole "we're in vegas so let's skip all that nonsense..." thing over new years but it didn't work |
| 17:54.43 | beek | Nothing says "I love you" better than saying "I do" in the Elvis Chappel |
| 17:54.57 | _Corey_ | That's what I said |
| 17:55.03 | _Corey_ | eh |
| 17:55.44 | leifmadsen | I wish we did the remote wedding thing as it would be significantly cheaper and easier |
| 17:55.49 | leifmadsen | but the day was still quite fun |
| 17:55.57 | leifmadsen | no I have to work hard for a year to pay it off :D |
| 17:56.09 | p3nguin | Now it's all over and you don't have to worry about it again. |
| 17:56.12 | _Corey_ | I'm stuck paying too, so that doesn't help :) |
| 17:56.33 | beek | So you tell her: If you want a traditional wedding doesn't that mean that your father pays? |
| 17:56.38 | *** join/#asterisk timahvo1 (~rogue@41.212.123.197) |
| 17:56.42 | _Corey_ | I'm sure it will be fine, it's just the unending lead-up that's the killer |
| 17:57.02 | _Corey_ | beek: They paid her student loans off, so I can't really call them on it |
| 17:57.29 | leifmadsen | MIMO |
| 17:57.37 | leifmadsen | ~mimo |
| 17:57.40 | _Corey_ | seriously |
| 17:57.51 | leifmadsen | infobot: mimo is <reply> Money In Money Out |
| 17:57.51 | infobot | leifmadsen: okay |
| 17:57.51 | _Corey_ | goes back to humming with his eyes closed |
| 17:58.08 | leifmadsen | I think I have that syntax wrong again |
| 17:58.10 | beek | LOVE IT! |
| 17:58.21 | leifmadsen | nope got it right |
| 17:58.22 | leifmadsen | nice |
| 18:01.16 | anonymouz666 | leifmadsen: anything else I can do about the res_ais issue to speed up things? or just sit back and wait :) |
| 18:01.30 | leifmadsen | sit back and wait unless you can provide a patch :) |
| 18:01.44 | leifmadsen | otherwise, you can try and make it someone elses itch to scratch, or pay someone to scratch the itch |
| 18:01.57 | leifmadsen | (if the latter, I can recommend someone) |
| 18:02.03 | fenrus | i can report from world scout jamboree 2011 that we successfully used asterisk in a prepaid "call-home" telephony system :) |
| 18:02.12 | leifmadsen | fenrus: nice! |
| 18:02.31 | fenrus | leifmadsen, it was awesome :) |
| 18:02.35 | leifmadsen | I approve |
| 18:13.31 | *** join/#asterisk sbszulu (~dundubala@41.16.138.27) |
| 18:23.03 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 18:26.15 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
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| 18:35.01 | *** join/#asterisk felimwhiteley (~quassel@46.7.101.58) |
| 18:43.47 | beek | leifmadsen: I was just looking at the topic for this channel. Shouldn't dahdi-linux and dahdi-tools both be 2.5.0? |
| 18:53.21 | *** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 18:56.54 | *** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 18:58.07 | *** join/#asterisk delphus (~delphus@unaffiliated/delphus) |
| 18:59.05 | delphus | using asterisk 1.8.5.0 with realtime, there is no mysql column to configure encryption, right ? I'm using sip and trying to setup srtp in realtime mode... |
| 18:59.32 | leifmadsen | beek: probably -- since I don't do those releases, I often miss when they do get releases and forget to update the topic |
| 19:01.11 | *** topic/#asterisk by leifmadsen -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.0.0-beta1 (2011/07/22), 1.8.5.0 (2011/07/11), dahdi-linux 2.5.0 (2011/08/08), dahdi-tools 2.5.0 (2011/08/08), libpri 1.4.12 (2011/07/06) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
| 19:01.41 | bmoraca_work | Asterisk 10.0.0-beta? wtf? |
| 19:02.04 | p3nguin | ~asterisk10 |
| 19:02.04 | infobot | Asterisk 10 -- http://blogs.digium.com/2011/07/21/the-evolution-of-asterisk-or-how-we-arrived-at-asterisk-10/ |
| 19:02.21 | bmoraca_work | pulling a Cisco, apparently |
| 19:02.25 | leifmadsen | nope |
| 19:02.33 | leifmadsen | the blog explains why quite well |
| 19:02.43 | leifmadsen | Asterisk 10 == Asterisk 1.^H^H10 |
| 19:05.19 | *** join/#asterisk BuenGenio (~Gene@cm61-10-82-188.hkcable.com.hk) |
| 19:05.42 | bmoraca_work | ah |
| 19:07.07 | *** join/#asterisk billmania (~bill@38.98.130.98) |
| 19:10.04 | bmoraca_work | hmm |
| 19:10.13 | bmoraca_work | Asterisk 10 looks like it's got some good features |
| 19:10.43 | irroot | bmoraca_work indeed starting to roll it out |
| 19:11.43 | bmoraca_work | i'm in the middle of building a multitenant system, and might build it around 10 instead of 1.8. might also have to upgrade my softswitch for the t.38 gateway functionality. that'll be nice and useful... |
| 19:11.44 | leifmadsen | ya it does -- the video conferencing and HD conference audio is sweet |
| 19:11.50 | leifmadsen | a lot of work in ConfBridge() was done |
| 19:15.35 | bmoraca_work | is the t.38 gateway functionality going to be licensed? |
| 19:15.39 | voipguynumber1 | hola |
| 19:18.22 | anonymouz666 | leifmadsen: the 1.8 confbridge does video mixing? |
| 19:18.32 | leifmadsen | anonymouz666: no |
| 19:18.52 | leifmadsen | anonymouz666: it was changed and is no in testing with Astersk 10 |
| 19:18.57 | leifmadsen | s/no/now/ |
| 19:19.08 | leifmadsen | bah |
| 19:20.41 | *** join/#asterisk [sr] (~kvirc@pal-213-228-140-150.netvisao.pt) |
| 19:21.23 | delphus | seriously folks, I've been trying to setup a sip peer using mysql realtime and there is no column for encryption meaning that it would only works if I put in the general sip lines, forcing every peer to use srtp. I know that its possible to set for a specific peer when not using realtimne. does anyone know anything aobut it ? thank you in advance. |
| 19:23.01 | anonymouz666 | leifmadsen: why always the good stuff is not in the version we are using? :P |
| 19:24.02 | leifmadsen | anonymouz666: because it requires major changes :) |
| 19:24.46 | leifmadsen | well, I guess I mean more invasive than would be reasonable for a dot-release within a release branch |
| 19:25.43 | Gugge | delphus: cant you just create the column, like all other settings |
| 19:26.51 | JonathanRose | Delphus: Is the problem that you don't have the clumn for encryption in your sip peer table or is it that having it just doesn't do anything? |
| 19:27.29 | JonathanRose | If it's the first, just add the column via SQL or your favorite DB management tool. |
| 19:27.50 | JonathanRose | *column |
| 19:30.28 | leifmadsen | alter table sippeers add column encryption after qualify; (or something like that :)) |
| 19:30.45 | leifmadsen | oh, and of course specify the type for your column heh |
| 19:30.51 | Gugge | that wouldnt work in my setup, as i use a view :) |
| 19:31.22 | leifmadsen | add or replace view foo as select .... :) |
| 19:31.26 | Gugge | :) |
| 19:31.27 | voipguynumber1 | wonders who greenwolf is... |
| 19:33.13 | irroot | leifmadsen Gugge dont forget that asterisk only works with taglen 80 |
| 19:33.24 | leifmadsen | taglen? |
| 19:33.49 | Qwell | as opposed to 32 |
| 19:34.46 | irroot | yeah not the keylen its the taglen in the sip packet |
| 19:34.58 | irroot | can be 32/80 |
| 19:35.16 | irroot | if the device uses 32 asterisk barfs |
| 19:35.57 | acidfoo | is there wiki pages on how to use asterisk unit test framework ? thank you |
| 19:36.08 | Gugge | i dont have any idea what you are talking about :) |
| 19:36.17 | irroot | is off to bed soon |
| 19:36.48 | irroot | Gugge dont stress about it if you see a option "tag length" only set it to 80 |
| 19:37.10 | Gugge | does that have anything to do with realtime columns? :) |
| 19:40.41 | *** part/#asterisk irroot (~irroot@41.122.199.173) |
| 19:41.47 | acidfoo | leifmadsen, do you know if there is wiki pages on how to use asterisk unit test framework ? |
| 19:42.22 | leifmadsen | redirects to pabelanger |
| 19:42.43 | delphus | JonathanRose: in fact, what seems to be the problem is that there is no encryption column in sql table. so I would need to create a specific peer in sip.conf or set it global |
| 19:43.29 | pabelanger | acidfoo: What are you looking to do? Just run the testsuite or actually write a test? |
| 19:44.14 | voipguynumber1 | can a op run /massunban ? |
| 19:44.22 | JonathanRose | delphus: Why don't you just add the column to the table? |
| 19:46.05 | delphus | JonathanRose: will try now, I didn't know I could just add columns out of the blue. |
| 19:46.13 | JonathanRose | Yeah, you can. |
| 19:46.18 | delphus | JonathanRose: thank you. |
| 19:46.21 | JonathanRose | No problem. |
| 19:46.29 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 19:47.15 | pabelanger | voipguynumber1: why? |
| 19:47.47 | *** join/#asterisk devil_evoxxx (~d3v1l@87.13.67.31) |
| 19:48.50 | p3nguin | a op? maybe an op. |
| 19:49.32 | devil_evoxxx | hi all |
| 19:54.21 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 19:57.53 | delphus | JonathanRose: it worked. so simple and easy. thanks for the tip. |
| 20:00.47 | Gugge | bahh, im the first to mention it, and he gets the thankyou :P |
| 20:01.43 | *** join/#asterisk sbszulu (~dundubala@41.17.232.183) |
| 20:02.12 | delphus | Gugge: my bad, I didn't see it in the first time, thank you as well :) |
| 20:02.37 | Gugge | Heh, no problem. Im just glad you got it fixed :) |
| 20:02.50 | delphus | yeah, not even tricky, shame on me. |
| 20:03.43 | anonymouz666 | app_queue crashing with 220 active calls :/ |
| 20:05.33 | JonathanRose | Delphus: If you made the DB table based on some of the documentation, it'd be no wonder you'd have missed that fact. It isn't exactly overtly explained anywhere I looked anyway :P |
| 20:05.57 | JonathanRose | One thing you need to be wary of though is that not everything in the realtime peer table is 1 to 1 with the normal peer definitions. |
| 20:08.42 | psilikon | I going out on a limb here: Is there anyway to test amd strictly from the dialplan? |
| 20:09.43 | devil_evoxxx | i'm using asterisk 1.8.5.0 , i'm testin this new LTS version ( my latest version of working asterisk is 1.4.37). I've observed a different behavior during Dial application. In asterisk 1.8 if i dial an extension in asterisk 1.4 for example Dial(SIP/10|60), if extension 10 is not reachable the call fall in the next extension ( i have sometimes like Goto(exception,s-${DIALSTATUS},1) ). In asterisk 1.8.5.0 if the extension is not reachable the call not fall |
| 20:10.48 | JonathanRose | psilikon: perhaps originate? Like you could originate out to a line you know has an answering machine and use an extension with AMD on it. |
| 20:11.26 | JonathanRose | Not sure if it'll work or not as I haven't touched AMD yet. |
| 20:11.50 | p3nguin | devil_evoxxx: First of all, SIP/10 is not extension 10, it's a SIP device named 10. Second of all, I think your paragraph was truncated at "the call not fall." |
| 20:12.40 | *** join/#asterisk johnnyasterisk (~johnnyast@89.18.71.44) |
| 20:13.19 | *** part/#asterisk johnnyasterisk (~johnnyast@89.18.71.44) |
| 20:13.56 | devil_evoxxx | Thankyou for reply p3nguin, yes i want to say Sip device named 10. I paste here the next paragraph after truncating |
| 20:14.08 | devil_evoxxx | the call fall in the next extension ( i have sometimes like Goto(exception,s-${DIALSTATUS},1) ). In asterisk 1.8.5.0 if the extension is not reachable the call not fall to the next exception, why? |
| 20:14.36 | _Corey_ | Anyone have the link to that blog posting a while ago about MOH and cellular codecs? I thought it was written by someone at Digium and am having trouble finding it. |
| 20:14.41 | *** join/#asterisk linuxgecko (~gecko@99-182-113-98.lightspeed.clmboh.sbcglobal.net) |
| 20:15.03 | p3nguin | devil_evoxxx: You mean if the DEVICE is not reachable. |
| 20:15.16 | devil_evoxxx | yes |
| 20:15.23 | *** part/#asterisk merlin8282 (~merlin828@AStrasbourg-554-1-203-129.w83-203.abo.wanadoo.fr) |
| 20:15.30 | devil_evoxxx | here can you see a piece of my dialplan: http://pastebin.com/Mj2zss72 |
| 20:15.45 | p3nguin | The result of DIALSTATUS should work the same way in both branches 1.4 and 1.8. |
| 20:16.36 | p3nguin | I use a similar method in my dialplan. |
| 20:16.44 | devil_evoxxx | but in 1.8 i've got this if the sip device is not reachable |
| 20:16.46 | devil_evoxxx | http://pastebin.com/N3cTvNP7 |
| 20:16.53 | *** join/#asterisk CoffeeIV (~rgr@dsl093-217-226.aus1.dsl.speakeasy.net) |
| 20:18.20 | eja | hi guys, is there a way to test via the cli console which extension asterisk will match given a dialed number? |
| 20:18.26 | p3nguin | devil_evoxxx: You shouldn't have used host=0809958742 in your sip.conf for that device. |
| 20:18.39 | p3nguin | eja: yes. dialplan show extension@context |
| 20:19.34 | psilikon | JonathanRose, Thanks for the input. That is actually how I was doing it but I wanted to accomplish the same thing without having have a physical endpoint with an answering machine. |
| 20:19.36 | WIMPy | ignore 1 -* |
| 20:19.58 | eja | thanks p3nguin! |
| 20:20.09 | JonathanRose | psilikon: I have no idea if this would work or not, but perhaps Asterisk would detect its internal voicemail application the same way it would an answering machine? |
| 20:20.19 | devil_evoxxx | in sip.conf the device is configured as : http://pastebin.com/SSHN8NYn |
| 20:20.57 | devil_evoxxx | and host is set to dynamic not to 0809958742 |
| 20:21.04 | psilikon | JonathanRose, That is kinda what I am after. Or better yet if I could force the signal via a channel variable or something similiar. |
| 20:21.18 | p3nguin | devil_evoxxx: Somewhere there is an error, because asterisk was trying to look up host 0809958742. |
| 20:23.01 | devil_evoxxx | and where do you look ? |
| 20:23.21 | p3nguin | The first thing I would do is reload the sip module and see if it happens again. |
| 20:24.30 | devil_evoxxx | reloaded and..yes it'happens again :( |
| 20:25.41 | prgmrchris | reformat |
| 20:26.15 | devil_evoxxx | it's a fresh machine..just make for testing today |
| 20:28.34 | devil_evoxxx | mmm, ok i've observed that if i call a phone not set in sip.conf asterisk give me the error post below |
| 20:28.55 | devil_evoxxx | otherwise, it work |
| 20:29.29 | p3nguin | Reformat? Right, because the file system is corrupted. Pfft. |
| 20:35.11 | eja | is there a way to filter debug info on the console? |
| 20:35.12 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 20:35.23 | p3nguin | Filter it for what? |
| 20:35.35 | eja | tough to isolate anything with all the other noise |
| 20:35.37 | eja | called number |
| 20:36.51 | *** join/#asterisk DelphiWorld (~VoIpGuy@openvpn/user/DelphiWorld) |
| 20:36.55 | p3nguin | You lost me. I thought you were doing something like sip debug. |
| 20:36.59 | DelphiWorld | everyone wish moy a happy birth day |
| 20:37.09 | *** join/#asterisk godmachine-x6 (~godmachin@unaffiliated/godmachine-x6) |
| 20:37.11 | p3nguin | Happy birthday, moy! |
| 20:37.27 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202) |
| 20:37.34 | p3nguin | I don't even know moy. |
| 20:38.05 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 20:39.52 | *** join/#asterisk bchia (~Adium@nat/digium/x-zjgphjzlthkgkxut) |
| 20:40.02 | *** join/#asterisk CoffeeIV (~rgr@dsl093-217-226.aus1.dsl.speakeasy.net) |
| 20:40.35 | *** join/#asterisk mhaddog (~mhaddog@z65-50-116-231.ips.direcpath.com) |
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| 20:43.57 | *** join/#asterisk Kotake (~Kotake@unaffiliated/kotake) |
| 20:43.59 | Kotake | hi guys |
| 20:44.09 | Kotake | <t_dot_zilla> fuck #asterisk #asterisk #asterisk #asterisk #asterisk #asterisk #asterisk #asterisk #asterisk #asterisk |
| 20:44.11 | Kotake | why is he mad? |
| 20:44.24 | p3nguin | Probably because he got banned. |
| 20:44.31 | *** part/#asterisk delphus (~delphus@unaffiliated/delphus) |
| 20:44.41 | Kotake | ah |
| 20:44.43 | Kotake | why'd he get banned? |
| 20:44.49 | p3nguin | Where was he spewing that nonsense? |
| 20:44.49 | Kotake | he posted that in a non-trolling channel |
| 20:44.57 | Kotake | what a *censored* |
| 20:45.06 | Kotake | it's a secret place. |
| 20:45.22 | *** join/#asterisk nny (~SM@cpe-174-107-223-014.sc.res.rr.com) |
| 20:45.22 | p3nguin | From what I could determine, he got banned for impersonating Linus Torvalds. |
| 20:45.28 | Kotake | hahahahah |
| 20:45.35 | nny | anyone have an opinion to what the most stable verison of asterisk currently is? |
| 20:45.37 | voipguynumber1 | that's enough to get banned? |
| 20:45.47 | linuxgecko | i wish goggle would make up theri mind with chan_talk coding needs :-( |
| 20:45.47 | Kobaz | anyone want a Grandstream 8 Port FXO (GXW-4108) |
| 20:45.51 | *** join/#asterisk xol (~xol@nat-216-228-242-218.byuh.edu) |
| 20:45.54 | voipguynumber1 | did people really think he was linus? |
| 20:45.58 | *** join/#asterisk enroxorz (~enroxorz@65-117-22-130.dia.static.qwest.net) |
| 20:45.58 | Kotake | no way |
| 20:46.01 | Kotake | can you post logs in pm? |
| 20:46.06 | p3nguin | As I said, that's my determination. It could have been something related to that, while not actually being that reason. |
| 20:46.09 | enroxorz | wait, someone pretended to be linus here? |
| 20:46.18 | voipguynumber1 | i'm Linus |
| 20:46.22 | p3nguin | Give me a few minutes, and I'll dig it up and paste it. |
| 20:46.22 | Kotake | same. |
| 20:46.38 | enroxorz | thanks p3nguin |
| 20:46.40 | JonathanRose | I'm pretty sure he was banned for being a troll in general. |
| 20:46.42 | enroxorz | i gotta read this |
| 20:46.46 | leifmadsen | that guy was banned for being a colossal troll |
| 20:47.14 | leifmadsen | (colossal is my new word of the day) |
| 20:47.17 | CoffeeIV | i have a * 1.8.5.0 compiled from source on an Ubuntu server, and it keeps crashing. I've gotten cores and backtraces and filed an issue: https://issues.asterisk.org/jira/browse/ASTERISK-18349 I'd like to pay someone to fix this and look over my server and make it reliable. I'm in Austin, Texas. Any recommendations ? |
| 20:47.17 | Nugget | Don't mess with Texas. |
| 20:47.18 | *** join/#asterisk SaborSole (~SaborSole@70.44.118.243.res-cmts.sefg.ptd.net) |
| 20:47.21 | *** join/#asterisk ruied_ (~ruied@pa4-84-91-140-68.netvisao.pt) |
| 20:47.21 | voipguynumber1 | aren't trolls small by nature |
| 20:47.28 | SaborSole | What is this I dont even? |
| 20:47.39 | voipguynumber1 | it's like saying he was a giant midget |
| 20:48.25 | navaismo | CoffeIV have you tried with CentOS |
| 20:48.43 | linuxgecko | Kobaz: you use it to turn 8 PSTN lines into sip ? |
| 20:48.55 | CoffeeIV | navaismo: no, it would be a bit of pain to do that - do you have a reason to think that would make a difference ? |
| 20:48.58 | ruied_ | WIMPy, This afternoon, the exact match of the DDI in asterisk coming from ISDN worked, thanks.. |
| 20:49.19 | Kobaz | linuxgecko: yeap |
| 20:49.55 | Kobaz | customer wound up switching to audiocodes because it has more features, but this box works fine |
| 20:50.10 | navaismo | CoffeIV only personal one, All of my Asterisk run in centos and no crashes, other issues i have but no crahses |
| 20:50.18 | CoffeeIV | navaismo: if it was a Ubuntu vs. CentOS thing, I would like track down what library or whatever was the difference, and file the appropriate bug reports in Ubuntu or Debian, or pay someone to do that. I'd like to fix my problem in a long-term sort of way |
| 20:50.19 | linuxgecko | Kobaz: sorry, i don't have a need for that :-( now if it were fxs :):) |
| 20:50.34 | Kobaz | ah |
| 20:50.43 | p3nguin | Well, it's in a few bits, so I have to patch it together. |
| 20:50.53 | CoffeeIV | navaismo: I will add that to the things to try. I'd have to move it to a different hardware to try that, but it's feasiable I think |
| 20:50.57 | *** join/#asterisk bchia (~Adium@nat/digium/x-rpqbharwrserskcu) |
| 20:51.34 | *** join/#asterisk bchia (~Adium@nat/digium/x-muyqbuuojvmwblou) |
| 20:51.43 | navaismo | I installed only a couple of asterisk in Debian, and no crashes too, maybe as you suggest its a hardware problem. |
| 20:52.03 | *** part/#asterisk enroxorz (~enroxorz@65-117-22-130.dia.static.qwest.net) |
| 20:52.12 | Kobaz | what's crashing? |
| 20:52.18 | Kobaz | kernel crash or asterisk crash? |
| 20:53.56 | nny | anyone have a suggestion for which release of asterisk is considered the most stable? Currently have an issue that is fording asterisk to restart/reload in 1.8 for no obvious reason. Can't run a debug on it atm |
| 20:54.17 | Kobaz | if you want solid as can be... 1.4 |
| 20:54.18 | WIMPy | ruied_: Yes sometimes the important things fail silently. I had that with SIP the other day. |
| 20:54.46 | Kobaz | but 1.8 has been really good to me once I fixed the bugs I was hitting |
| 20:54.53 | ruied_ | WIMPy, yeap |
| 20:55.42 | WIMPy | was pretty pissed that a sip debug was the only way to find out what error message was received. |
| 20:57.25 | *** join/#asterisk raker (~raker@cpe-174-102-239-159.wi.res.rr.com) |
| 20:57.30 | CoffeeIV | Kobaz: asterisk crashes, usually with a segfault: https://issues.asterisk.org/jira/browse/ASTERISK-18349 |
| 20:57.58 | p3nguin | Okay, here it is. http://pastebin.com/tAmKxNQz |
| 20:59.21 | Kobaz | CoffeeIV: only happens with conferencing? |
| 21:00.32 | CoffeeIV | Kobaz: a common failure is a crash when a lot of people are leaving a conference. However, I don't think that is the only crash. I'm not sure they are always related to conferencing. |
| 21:00.33 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 21:02.02 | BenC[UK] | When I call Queue() is there anyway to get a variable somehow of which agent the call went to? |
| 21:02.26 | *** join/#asterisk lvlolvlo (~lvlolvlo@unaffiliated/lvlolvlo) |
| 21:02.26 | leifmadsen | when do you want it? |
| 21:02.53 | leifmadsen | BenC[UK]: use DumpChan() to see what channel variables are available to you |
| 21:03.25 | Kobaz | CoffeeIV: using MeetMe or Conf? |
| 21:03.39 | CoffeeIV | Kobaz: I'll check . . . |
| 21:04.55 | CoffeeIV | Kobaz: ConfBridge() is what we're using |
| 21:05.12 | Kobaz | yeah i found that in the backtrace |
| 21:05.19 | Kobaz | well, i would recommend switching to MeetMe in the meantime |
| 21:05.31 | Kobaz | I can try to see if i can reproduce the problem |
| 21:06.20 | CoffeeIV | Kobaz: I can try MeetMe. Would it help if I gave you all or part of our dialplan, scrubbed of passwords and etc of course ? |
| 21:12.00 | Kobaz | wouldn't hurt to paste it up |
| 21:12.22 | CoffeeIV | Kobaz: ok |
| 21:15.13 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 21:17.00 | devil_evoxxx | exi |
| 21:17.18 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 21:18.31 | *** part/#asterisk Bob_Pierce (~WESTMAN\p@216.36.132.162) |
| 21:19.18 | WIMPy | p3nguin: Was your callerID question earlier fully answered or was there any question left? |
| 21:20.03 | CoffeeIV | Kobaz: I put up the relevant chunks of extensions.conf here: http://pastebin.ch/6658 |
| 21:20.05 | p3nguin | I think the subject still needs some attention. |
| 21:21.05 | WIMPy | p3nguin: I certainly wouldn't mind if Asterisk would provide the same functionality as the PSTN did. |
| 21:22.54 | Linuturk | I don't understand why asterisk went from 1.x.x to 10.x.x |
| 21:22.59 | Linuturk | instead of 2.x.x |
| 21:23.25 | Linuturk | mind boggling |
| 21:23.29 | WIMPy | (or that of a plastic PBX) |
| 21:23.42 | WIMPy | ~asterisk10 |
| 21:23.43 | infobot | Asterisk 10 -- http://blogs.digium.com/2011/07/21/the-evolution-of-asterisk-or-how-we-arrived-at-asterisk-10/ |
| 21:23.53 | kn0x | Linuturk: me either |
| 21:24.04 | kn0x | version inflation sucks |
| 21:25.33 | WIMPy | agrees |
| 21:25.46 | Kobaz | it's like java |
| 21:26.02 | *** join/#asterisk omgwtf (hater@lolunix.org) |
| 21:26.07 | Kobaz | java is really 2.7 or whatever but they call it java 7 |
| 21:26.11 | Kobaz | it's marketing |
| 21:26.37 | Kobaz | you know what i like better, is years for versions |
| 21:26.54 | WIMPy | Like Windows '95? |
| 21:26.55 | Kobaz | if you have two releases a year, call it Asterisk 2011.1 and Asterisk.2011.2 |
| 21:27.18 | WIMPy | already voted for the date in ISO format. |
| 21:27.56 | Kobaz | if you do it that way, then no one can complain about version inflation |
| 21:28.23 | ChannelZ | unless you put in minute granularity |
| 21:28.26 | Kobaz | hehe |
| 21:28.43 | Kobaz | Asterisk.2011.1314653315 |
| 21:28.47 | omgwtf | I'm running asterisk 1.4. I have an issue where i'm using addqueuemember and the member is added to the queue but it is marked as dynamic invalid. I have google for people with a similar problem. I found one case from almost 4 years ago that suggested reloading app queue (which doesn't resolve my issue) Does anyone have any idea what else I can try to resolve this? |
| 21:28.49 | ChannelZ | I don't really care, but what really bugs me was when companies went back and forth.. from version numbers into years and then back into version numbers |
| 21:28.54 | Kobaz | use unixtime |
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| 21:32.10 | Kobaz | or just use names |
| 21:32.14 | Kobaz | Asterisk Lion Server |
| 21:34.00 | ChannelZ | Names are worse |
| 21:34.49 | Kobaz | hah yeah |
| 21:34.58 | Kobaz | that was a joke :P |
| 21:35.20 | ChannelZ | At least with Android they use ascending letters so you stand half a chance of knowing what's older than what, the whole Mac OS cat thing is a fail |
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| 21:39.21 | ChannelZ | Of course now they'll have to start naming them "Mac OS 1 A.J." and such |
| 21:52.55 | Eitan | hey guys, what is a good place to find reliable asterisk/freepbx contracters for hire? need some stuff done |
| 21:53.39 | billmania | Eitan: In what part of the world? |
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| 22:12.31 | tyman | I'm seeing this line: " WARNING[20576]: chan_sip.c:19667 handle_response_invite: just did sched_add waitid(2403401) for sip_reinvite_retry for dialog" repeated ~30x in my log many times a day. Googling indicates that this may have something to do with session-timers. Is this something anyone's come across? |
| 22:13.21 | lvlolvlo | @tyman you can try disabling session-timers to see if that is the source of this |
| 22:13.31 | lvlolvlo | although i wouldn't disable session timers for more than a day |
| 22:13.41 | tyman | why is that? |
| 22:14.01 | tyman | session-timers=refuse? |
| 22:14.04 | lvlolvlo | yes |
| 22:14.12 | lvlolvlo | session timers help in the case of an open dialog |
| 22:14.19 | lvlolvlo | err a dialog which is left open |
| 22:14.51 | lvlolvlo | e.g. connection hiccup whereby a BYE is not sent/received successfully |
| 22:15.13 | tyman | so, can session be set for say '5 hours' or something? |
| 22:15.20 | lvlolvlo | yeah it can |
| 22:15.30 | tyman | okâ¦i'll research that... |
| 22:15.31 | lvlolvlo | i would typically set it to 3 hours |
| 22:16.02 | lvlolvlo | so you can do something like session-expires=10800 |
| 22:16.42 | tyman | okâ¦thx. I read people we're having issues at exactly 15mins and deciding to session-timers=refuse which 'fixed' the probâ¦but the thread sounded like a hack.. |
| 22:16.46 | lvlolvlo | so it'll send or expect session refreshes every 3 hours and every 3/2 interval (e.g. 90 minutes) |
| 22:16.49 | tyman | this sounds better |
| 22:17.09 | lvlolvlo | yeah -- was the 15 min thing causing call drops? |
| 22:17.45 | tyman | i haven't personally experienced itâ¦i've only seen it in my logs |
| 22:17.51 | tyman | and caught it on the console |
| 22:18.04 | lvlolvlo | do you know if they were on outbound calls? |
| 22:18.23 | tyman | noâ¦i'll look into it more |
| 22:18.44 | lvlolvlo | kk -- but yeah, putting in the 10800 value should fix it |
| 22:18.55 | lvlolvlo | unless if you have calls which are >3 hours |
| 22:18.58 | tyman | been too busy to do some forensicsâ¦just saw it happen on my console and decided to see if this was a trivial issue |
| 22:19.05 | lvlolvlo | well i'm sorry |
| 22:19.11 | lvlolvlo | it won't fix it |
| 22:19.16 | lvlolvlo | it'll band-aide it |
| 22:19.27 | tyman | ok⦠|
| 22:19.30 | lvlolvlo | the true fix would be to patch the underlying bug |
| 22:20.07 | tyman | still don't understand what's causing that errorâ¦is this a standards compliant implementation issue for the renegotiation? |
| 22:21.00 | WIMPy | What are you talking to when it fails? |
| 22:21.04 | lvlolvlo | well i haven't studied it too much but from what i've seen it looks like it's just a bug in the way the session refreshes |
| 22:21.39 | tyman | WIMPy: flowroute upstream... |
| 22:21.51 | lvlolvlo | inbound call? |
| 22:21.55 | WIMPy | What are they using? |
| 22:22.11 | tyman | here's the relevant sip config currently in place for my flow route stance: session-expires=3600 |
| 22:22.12 | tyman | session-timers=refuse |
| 22:22.12 | tyman | session-refresher=uas |
| 22:22.12 | tyman | session-minse=90 |
| 22:22.45 | lvlolvlo | tyman -- if you're using session-timers=refuse then you don't need the additional configuration parameters |
| 22:22.56 | tyman | Had some problems with them a year ago or so and they told me to put that in there.. |
| 22:23.11 | tyman | yeahâ¦.that makes senseâ¦it's irrelevant if i'm refusing them |
| 22:23.28 | lvlolvlo | if you're using session-timers=originate, then you should have the other values |
| 22:23.34 | WIMPy | only had to disable them when talking to a CCM so far. |
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| 22:25.00 | eja | hey guys, on the READ() call. the 2nd paramater is the filename. how do i found out where that file is being played from? |
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| 22:25.22 | tyman | lvlolvlo: so, although those superfluous session related statements are there, this brings up the point that it may not be timers at all thenâ¦as we're disabling them |
| 22:25.26 | nny | eja /var/lib/asterisk/sounds iirc |
| 22:25.54 | eja | thanks! |
| 22:31.33 | Eitan | billmania: PM |
| 22:32.25 | lvlolvlo | very true |
| 22:32.56 | lvlolvlo | if you already have the refuse statement, then this may not be an issue with session timers |
| 22:37.48 | dan__t | Hrm. can I have a dynamically created context, like [contextname_${CALLERID(num)}] ? |
| 22:38.00 | dan__t | Just used ${CALLERID(num)} as an example here |
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| 22:45.43 | WIMPy | p3nguin: http://www.voip-info.org/wiki/view/RDNIS |
| 22:47.31 | WIMPy | I guess that should have been fixed when it was restructured in 1.8 anyway. |
| 22:50.20 | WIMPy | wonders how much of the normal PSTN functionality the different channeltypes could implement in theory. |
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