| 00:02.29 | *** join/#asterisk Mhaddog_Mac (~anonymous@z65-50-116-17.ips.direcpath.com) |
| 00:07.42 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 00:15.00 | *** join/#asterisk ariel_ (~chatzilla@99-1-236-49.lightspeed.miamfl.sbcglobal.net) |
| 00:37.56 | *** join/#asterisk high-rez (~gus@2001:470:8115::2) |
| 00:38.07 | high-rez | Does dahdi itself compile against libpri, or does just asterisk? |
| 00:38.34 | *** join/#asterisk chopp (~chopp@unaffiliated/chopp) |
| 00:40.17 | WIMPy | Asterisk |
| 00:43.04 | *** join/#asterisk coppice (~chatzilla@60.157.17.210.dyn.pacific.net.hk) |
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| 00:52.22 | high-rez | Thanks! |
| 00:52.23 | *** part/#asterisk high-rez (~gus@2001:470:8115::2) |
| 00:54.17 | leifmadsen | order of compilation: libpri --> dahdi --> asterisk |
| 01:05.20 | mzb | is it possible to enable the extended repository from the command line? ( *without* installing a module from the extended repo first ) |
| 01:09.05 | *** join/#asterisk kobaz (~kobaz@24.145.23.232) |
| 01:10.23 | phix | hey |
| 01:10.37 | Kobaz | it wasnt me |
| 01:15.17 | phix | should I update to latest stable release of asterisk or not? |
| 01:15.48 | Kobaz | sure |
| 01:15.51 | phix | I am currently using 1.4.24.2 |
| 01:16.28 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 01:16.33 | Kobaz | do you need new features? |
| 01:16.52 | Kobaz | as usual, don't put anything into production you haven't tested in the lab yet |
| 01:17.05 | Kobaz | behavior changes, maybe a bug or two, you neverk now |
| 01:17.10 | Kobaz | everyones use case is different |
| 01:17.15 | nestAr | anyone used the snom m3 stuff? |
| 01:17.25 | phix | I don't know what new features are available |
| 01:17.49 | phix | I have a snom 300 :) |
| 01:18.43 | Kobaz | it's mostly stuff to make dialplan developers more happy |
| 01:18.44 | phix | My setup consists of a TDM400p, one land line one internal phone line one VOIP / SIP provider |
| 01:18.54 | Kobaz | like more control over voicemail via minivm |
| 01:18.57 | *** part/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
| 01:18.57 | *** part/#asterisk eskp (~eskp@c877.ie.com.au) |
| 01:19.01 | Kobaz | Originate() from dialplan |
| 01:19.11 | Kobaz | stuff like that... there's human readable changelog somewhere |
| 01:19.14 | phix | I would like to de-americanize my VM system :) |
| 01:19.26 | phix | ok nice |
| 01:19.29 | Kobaz | you can build you own vm system starting in 1.6.2 |
| 01:19.43 | phix | Difficult or straight forward? |
| 01:19.52 | Kobaz | just read core show applications |
| 01:19.57 | Kobaz | it has all the stuff right there |
| 01:21.24 | phix | ok |
| 01:22.34 | *** join/#asterisk Kumbang (~kumbang@180.245.137.5) |
| 01:24.25 | phix | well is 1.4 branch going to be dropped / deprecated any time soon? |
| 01:24.49 | Kobaz | not for another... umm, it's in the wiki |
| 01:25.24 | russellb | in a few months |
| 01:25.28 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 01:25.40 | russellb | https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions |
| 01:25.41 | phix | haha really? |
| 01:25.54 | russellb | just over 2 months actually |
| 01:25.57 | russellb | wow, sooner than i thought |
| 01:26.07 | phix | so should I upgrade to 1.6 or 1.8 ? |
| 01:26.20 | Kobaz | i'm on 1.6.0.26 |
| 01:26.22 | Kobaz | but that's me |
| 01:26.24 | russellb | 1.8 |
| 01:26.33 | phix | can I still use extensions.conf or do I need to use ael? |
| 01:26.40 | phix | or should I be using ael? |
| 01:26.42 | russellb | you can still use .conf |
| 01:26.42 | Kobaz | you can use basically everything |
| 01:26.48 | Kobaz | you should use ael for readability sake |
| 01:26.48 | phix | oh ok |
| 01:26.54 | russellb | please read UPGRADE.txt and the other similarly named files |
| 01:27.01 | phix | I use zaptel atm, it is called something else in 1.6 hey |
| 01:27.05 | Kobaz | there are some changes |
| 01:27.12 | phix | I would rather discuss it in here with you guys :D |
| 01:27.20 | Kobaz | like breaking changes with AGI... you have to use , instead of | |
| 01:27.40 | russellb | yes, zaptel has not been maintained in a few years |
| 01:28.16 | phix | ok I used , out of habbit |
| 01:28.40 | phix | hmmm I wonder if Debian Lenny has 1.6 yet |
| 01:29.00 | phix | hmmm nope |
| 01:29.08 | russellb | the _new_ release of debian (the one from the last week) has 1.6.2 |
| 01:29.24 | phix | wow a new debian was released last week? |
| 01:30.01 | nestAr | wonders never cease |
| 01:30.07 | nestAr | we'll be 6 years before the next |
| 01:30.10 | nestAr | ;) |
| 01:30.43 | *** join/#asterisk adeel (~adeel@c-67-174-36-109.hsd1.ca.comcast.net) |
| 01:31.27 | phix | yay! squeeze! |
| 01:31.47 | phix | nestAr: I know, I am suprised a new one was released so soon :) |
| 01:32.43 | phix | hmmm any tips on making a TDM400p card behave and not complain about IRQs? |
| 01:40.33 | *** join/#asterisk dyer (~dyer@unaffiliated/voxeojohn) |
| 01:41.31 | dyer | Hello, I have a question... in our invite we are Aasterisk is setting the contact IP to the internal address, so the far end appears to be sending media the that address, how do we override the contact with the external IP? |
| 01:42.18 | *** join/#asterisk Mhaddog (~Mhaddog@z65-50-116-17.ips.direcpath.com) |
| 01:42.32 | nestAr | love this CLEC. I have blanket authorization to change anything related to their phone service... things that will cost them money.. |
| 01:42.46 | nestAr | but they wouldn't let me add a PTR record for the IP address |
| 01:43.22 | nestAr | correction, the move lady would do it for me.. during the move.. but if i wanted it done right now, i had to call tech support, and they said no.. change order authorization doesn't apply to DNS |
| 01:43.33 | nestAr | rolls eyes |
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| 02:19.24 | mzb | nm about my previous question re 3rd party ... I've figured it out |
| 02:26.38 | *** join/#asterisk bjhaid (~abejide@41.220.69.11) |
| 02:27.18 | devmod | Is there a dialplan cmd that will play a file on a loop? |
| 02:27.35 | nestAr | forever? |
| 02:29.27 | nestAr | exten => x,1,Play(jdjd) |
| 02:29.28 | nestAr | exten => x,2,Goto(1) |
| 02:29.29 | nestAr | ? |
| 02:30.01 | bjhaid | please can anyone tell me what is wrong with this line in my dialplan? exten => 2,n,Record(/tmp/a.gsm,3,30) |
| 02:30.17 | devmod | nestAr, right I could do that, but Im doing it over AMI |
| 02:30.27 | phix | bjhaid: is that the first line for 2? |
| 02:30.30 | Kobaz | paste more of it along with your console output of execution to pastebin.com |
| 02:30.46 | phix | an extension needs to start at 1 not n |
| 02:31.02 | bjhaid | phix: its not the first line, its a continuation of an extension |
| 02:31.05 | WIMPy | devmod: Then put that into your dialplan and redirect the call there. |
| 02:31.22 | *** join/#asterisk hobodave (~hobodave@pdpc/supporter/professional/hobodave) |
| 02:32.32 | devmod | WIMPy, oh I guess that is an option thx |
| 02:37.07 | phix | ok |
| 02:37.33 | phix | I couldn't see that from that one line you pasted :) you should pastebin I reckon |
| 02:38.02 | bjhaid | phix: is that synopsis for the record command accurate? |
| 02:38.42 | bjhaid | phix: can i record to gsm? because i have tried this and it doesnt work |
| 02:39.20 | phix | I have created a fallback dialer however it isn't working as expected. If my SIP provider cannot be contacted it should try the next SIP provider or landline, however I get a CONGESTED message instead which is the same as engaged right? |
| 02:41.00 | WIMPy | It can be. |
| 02:41.19 | WIMPy | You can check HANGUPCAUSE for additional information. |
| 02:42.15 | WIMPy | An engaged subscriber should give BUSY but depending on the technology used could give CONGESTION as well. |
| 02:42.27 | phix | hold on, I am trying to paste stuff in pastebin.ca for you bjhaid |
| 02:42.34 | phix | It is from my working config |
| 02:42.57 | bjhaid | okay |
| 02:43.07 | phix | WIMPy: ok, good, because I dont want to redial out if the other end was engaged but I do if the VoIP / IP provider cannot be contacted |
| 02:43.21 | phix | you know another pastebin site that doesn't take yrs to load? |
| 02:43.31 | phix | ~pastebin |
| 02:43.31 | infobot | [~pastebin] 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/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 02:43.47 | phix | nm |
| 02:43.48 | phix | thnx infobot |
| 02:44.22 | WIMPy | A HANGUPLOCATION would be extremely helpful for that kind of decision. |
| 02:44.43 | *** join/#asterisk guilhermebr (~Guilherme@189.63.54.51) |
| 02:44.54 | phix | http://asterisk.pastey.net/145948 |
| 02:45.07 | Kobaz | hanguplocation as in... who hung up? |
| 02:45.14 | bjhaid | thanks |
| 02:45.19 | phix | WIMPy: Is that available in 1.4.24.2? |
| 02:45.31 | WIMPy | No where the call was terminated. |
| 02:46.00 | WIMPy | Such as the intormation you get in the PSTN. |
| 02:46.19 | Kobaz | like on t1? or ss7? |
| 02:46.36 | WIMPy | yes |
| 02:46.42 | phix | WIMPy: that yes to me? |
| 02:46.58 | WIMPy | yes to Kobaz |
| 02:47.12 | Kobaz | i want WHOHUNGUP |
| 02:47.20 | Kobaz | oej has a branch for that |
| 02:47.31 | WIMPy | phix: hanguplocation is not available in any version so far. |
| 02:49.22 | *** join/#asterisk Elijah_ (~elijah@184-98-247-124.phnx.qwest.net) |
| 02:57.18 | leifmadsen | Kobaz: +1 |
| 02:57.35 | leifmadsen | would be nice to get it in for 1.10 |
| 02:57.55 | Kobaz | yeah |
| 02:58.18 | Kobaz | i forgot where his branches are |
| 02:58.34 | leifmadsen | http://svn.asterisk.org/svn/asterisk/team ? |
| 02:58.42 | Kobaz | i thought he had his own spot |
| 02:58.44 | Kobaz | i can check |
| 02:58.58 | leifmadsen | don't think so |
| 02:59.02 | leifmadsen | pretty sure he uses svn.asterisk.org |
| 02:59.08 | leifmadsen | http://svn.asterisk.org/svn/asterisk/team/oej/ |
| 02:59.26 | Kobaz | hmm, dont see his hangup stuff |
| 02:59.34 | Kobaz | i remember at astricon it was something like whodoneit |
| 03:00.05 | Kobaz | right along side of pine majo and cherry delight |
| 03:01.22 | leifmadsen | :) |
| 03:03.42 | Kobaz | maybe it was someone else workng on it and oej had mentioned it |
| 03:04.20 | *** join/#asterisk iceyp (~icepick@srs.affordabledomains.co.nz) |
| 03:04.31 | iceyp | hey guys, how can I tell asterisk where my license files are |
| 03:04.54 | iceyp | when i register a license it stores to /var/lib/asterisk/licenses but my asterisk seems to be looking somewhere else |
| 03:05.08 | iceyp | as it's installed in /home/servers/pbxware/asterisk |
| 03:05.30 | fauxalliance | ln -s |
| 03:11.31 | leifmadsen | that's probably the easiest -- I don't see a way of changing it in asterisk.conf which is where I'd expect to see it (if anywhere) |
| 03:21.50 | *** join/#asterisk Nugget (nugget@carrera.macnugget.org) |
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| 03:27.41 | Kobaz | whew, on new colo box |
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| 03:38.01 | iceyp | Well my asterisk license is in the default location |
| 03:38.03 | phix | WIMPy: oh ok |
| 03:38.07 | iceyp | so I'd expect it to find it in that folder |
| 03:38.19 | phix | so what should i Use instead then mr WIMPy ? |
| 03:38.44 | iceyp | i also tried copping it to /home/servers/pbxware/pw/var/lib/asterisk/licenses/ |
| 03:38.51 | WIMPy | You can only check HANGUPCAUSE for additional information. |
| 03:42.01 | leifmadsen | huh.... |
| 03:42.07 | leifmadsen | it's like my mouse just locked up.... |
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| 04:00.43 | drmessano | leifmadsen, may be a Windows update doing that |
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| 04:01.20 | *** mode/#asterisk [+oo Corydon76-home file] by hubbard.freenode.net |
| 04:01.40 | leifmadsen | I'm on Linux :) |
| 04:01.40 | leifmadsen | it really was the mouse |
| 04:01.45 | leifmadsen | took the batteries out for a few mins, and now it's back working again |
| 04:12.45 | *** join/#asterisk Bridge| (bridgelife@netburst.org) |
| 04:19.34 | drmessano | lol |
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| 04:38.53 | *** join/#asterisk CrashSys (~james@65344hfc124.tampabay.res.rr.com) |
| 04:39.32 | CrashSys | Is it possible to have 1 asterisk server register 4 account to 1 carrier? Every time I have more then 1 registered, asterisk seems to think all the calls are from the last entry |
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| 04:50.38 | leifmadsen | CrashSys: that's because it's matching based on IP address and not username |
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| 07:06.51 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
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| 08:17.38 | schmidts | good morning |
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| 08:27.56 | ChannelZ | happy Friday! |
| 08:27.59 | ChannelZ | whips his top off |
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| 09:11.29 | burnout1540 | Hi... i have what i hope is a basic question, yet the answer escapes me |
| 09:11.37 | burnout1540 | I have a dialplan and need to take some sort of action (write to the console, write to a file, move to a file, etc.) if I am not able to connect via my SIP trunk |
| 09:11.46 | burnout1540 | the h extension works to catch regular hangups |
| 09:12.20 | burnout1540 | but not issues like receiving a busy signal or a SIP trunk "Other Call Failure" response |
| 09:12.37 | burnout1540 | how can i catch these issues in my dial plan? |
| 09:13.09 | *** join/#asterisk heffer (~felix@fedora/heffer) |
| 09:16.12 | burnout1540 | i've read something about the 100 plus priority but i'm not quite sure if that's what i need |
| 09:18.39 | schmidts | burnout1540 take a look at the DIALSTATUS variable |
| 09:19.35 | burnout1540 | will do, thanks! |
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| 09:32.21 | Amnesia | hm, for some reason the dahdi module failed to build during set up |
| 09:32.35 | Amnesia | how can I rebuild it and if it has errors again where's the log file? |
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| 09:37.07 | phix | hi gang |
| 09:37.26 | schmidts | hi phix |
| 09:37.47 | phix | sup? |
| 09:38.30 | phix | i am at an indian restruant waiting for my rogan josh and naan |
| 09:38.46 | phix | om nom nom |
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| 10:25.12 | angryuser | Any sangoma guys here ? |
| 10:26.55 | iogan18tm | hi, need help guys. I've written in sip.conf register = user@10.255.10.10:pass@1.2.3.4 where 1.2.3.4 - is providers ip, 10** - ip i'm going to send as id. but in sip set debug on i see "from : sip:user@1.2.3.4" not "from: sip:user@10.**> as it should be. is that right? asterisk is 1.6.2 |
| 10:33.18 | *** join/#asterisk manji (~manjiki@penicuik.physics.upatras.gr) |
| 10:33.35 | manji | according to the RFC |
| 10:33.51 | manji | when the callee response with the 415 message |
| 10:34.05 | manji | (Unsupported Media Type.) |
| 10:34.25 | manji | the message should include the "accept" header |
| 10:34.32 | manji | with the supported media |
| 10:34.36 | manji | is that correct? |
| 10:34.38 | Bridge| | anyone know any services (like how skype used to offer free land line calls) where i could hook a small PBX to and be able to make free landline calls when needed? |
| 10:34.53 | Bridge| | using Trixbox if that helps |
| 10:35.26 | *** join/#asterisk Jasnejac (kvirc@81.91.107.236) |
| 10:37.07 | schmidts | Bridge| AFAIK can you use googletalk for free landline calls in USA and you can use googletalk with asterisk, but i dont think you can use it with trixbox |
| 10:37.58 | Bridge| | k |
| 10:40.05 | Bridge| | nope: The downloadable Google Talk client offers free PC-to-PC calling only - dialing out to landlines or mobile phones isn't supported at this time. |
| 10:42.14 | *** join/#asterisk gerhard7 (~gerhard7@212-123-146-122.ip.telfort.nl) |
| 10:43.47 | *** join/#asterisk Vec (~Vec@84.12.253.146) |
| 10:44.37 | Vec | I seem to have found a bug in Asterisk 1.4.39.1 when running Chanspy on a SIP channel from another SIP channel it always results in the channel being spied on hanging up. |
| 10:48.14 | schmidts | Vec this is a allready known bug, and let me check if there is allready a solution for this (read something) |
| 10:49.37 | Vec | thanks, looking in 1.4.40 to see if its fixed, but can't find any reference to it |
| 10:51.48 | schmidts | vec cant find it by now, but i am sure there was something about this, maybe you can serach at issues.asterisk.org |
| 10:52.28 | Vec | schmidts : I have seen this issue, but in much older versions of 1.4 and it was resolved but it seems as though its come back |
| 10:52.54 | schmidts | vec thats bad |
| 10:53.00 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
| 10:53.51 | Vec | schmidts : found it here, https://issues.asterisk.org/view.php?id=18647 but does not refer to version 1.4.39.1 |
| 10:54.31 | Vec | there is a comment there that refers to it in 1.4.39.1 |
| 10:55.18 | Vec | also bug in 1.8.2.2. |
| 10:56.18 | WIMPy | has not experienced that problem with current Asterisk, yet. |
| 10:58.35 | wdoekes2 | iogan18tm: try to encode the amp as %40 |
| 10:59.37 | wdoekes2 | unless you don't want 1.2.3.4 at all.. in which case you're probably out of luck |
| 10:59.41 | schmidts | vec i found something similar but with 1.6 branch issue 18647 |
| 11:02.09 | Vec | schmidts : its the exact same bug, if you read the comments it has be found in 1.4 and 1.8 branch |
| 11:03.00 | schmidts | vec i have read this too now ;) |
| 11:03.42 | Vec | really need a patch for this :O |
| 11:04.08 | schmidts | vec you can try the svn checkout for 1.4 everything after rev 303552 would be fine |
| 11:05.39 | schmidts | vec even 1.4.40-rc2 doesnt have this fix included AFAIK :( |
| 11:06.38 | schmidts | vec http://svnview.digium.com/svn/asterisk/branches/1.4/main/pbx.c?view=diff&r1=298176&r2=303552&pathrev=303552 here you can create a patch |
| 11:13.05 | schmidts | wdoekes2 i am testing the patch from rgagnon and it looks nice so far: 20015 sip peers [Monitored: 19999 online, 1 offline Unmonitored: 15 online, 0 offline] |
| 11:13.59 | schmidts | load average: 0,08, 0,15, 0,12 :D |
| 11:16.46 | schmidts | ok not so good |
| 11:18.07 | schmidts | do you think sending out 180mbit of sip options messages is a good idea over a 100mbit switch? |
| 11:18.13 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 11:21.17 | Vec | schmidts : how did you find out the bug was resolved in rev 303552 ? |
| 11:21.38 | schmidts | vec thats the revision russell bryant meant in the issue |
| 11:22.02 | iogan18tm | hi, need help guys. I've written in sip.conf register = user@10.255.10.10:pass@1.2.3.4 where 1.2.3.4 - is providers ip, 10** - ip i'm going to send as id. but in sip set debug on i see "from : sip:user@1.2.3.4" not "from: sip:user@10.**> as it should be. is that right? asterisk is 1.6.2 |
| 11:22.24 | wdoekes2 | iogan18tm: how many rows of irc scrollback do you have? |
| 11:22.40 | iogan18tm | many |
| 11:23.04 | iogan18tm | ohh |
| 11:23.08 | iogan18tm | sry |
| 11:23.18 | Vec | schmidts : ok, I guess I'm confused because he was referring to an issue on the 1.6 branch, I would think the revisions would have a diffirent numbering sequence |
| 11:23.59 | schmidts | vec he has submitted patches for 1.4, 1.6, 1.8 and branch which all solves the same problem and the rev i told you was the one from 1.4 |
| 11:24.08 | iogan18tm | wdoekes2: there is no amp in register string |
| 11:24.12 | wdoekes2 | schmidts: great that you're working on it :) |
| 11:24.29 | iogan18tm | wdoekes2: @ encode? |
| 11:25.11 | wdoekes2 | yes.. but that'd yield user%4010.x.y@1.2.3.4 probably.. which means that the username is user@10.x.y |
| 11:25.32 | wdoekes2 | s/username/user-part/ |
| 11:25.37 | wdoekes2 | infobut-- |
| 11:26.05 | schmidts | wdoekes2 registration test looks fine so far, but we will see if i really put it on load. 333 poke peer messages per second is not really much |
| 11:26.25 | *** join/#asterisk Kedare (~Kedare@onyxia.netyxia.net) |
| 11:30.11 | *** join/#asterisk sourcode (~code@ppp-58-8-161-125.revip2.asianet.co.th) |
| 11:30.37 | wdoekes2 | iogan18tm: ah.. I see now that the docs hint that something like that is possible.. perhaps someone else can shed some light on that |
| 11:32.02 | wdoekes2 | it works fine over here in 1.6.2 |
| 11:32.19 | iogan18tm | wdoekes2: as for me i see a bug here becouse fromdomain part is not used in sip connection and in register it fails also |
| 11:32.42 | iogan18tm | wdoekes2: oops. yeah, asterisk 1.6.0 |
| 11:32.52 | wdoekes2 | hm.. unsupported versions |
| 11:41.12 | schmidts | wdoekes2 i am not sure now but it looks like there is some slow locking handling with this patch, i have to take a deeper look at it :( |
| 11:42.50 | schmidts | 20143 active SIP dialogs ups :D |
| 11:46.54 | WIMPy | wonders if the packet size option to allow= is actually documented somewhere. |
| 11:49.02 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 11:53.37 | schmidts | Wimpy what do you mean? |
| 11:54.18 | WIMPy | Things like allow=alaw:10, but I just found it on the wiki. |
| 11:55.48 | schmidts | Wimpy i dont understand what you meant by packet size ;) do you know whats the default value is, asterisk use? 20 ms right? |
| 11:56.02 | WIMPy | yes |
| 11:56.07 | schmidts | ok |
| 11:57.45 | WIMPy | Except for ilbc which has a default of 30ms. |
| 11:58.42 | schmidts | Wimpy why? |
| 11:58.56 | schmidts | would this increase the transcoding time? |
| 11:59.03 | schmidts | s/would/wouldnt |
| 11:59.04 | WIMPy | It's the minimum allowed there. |
| 11:59.34 | schmidts | interesting, didnt know this ;) |
| 11:59.45 | WIMPy | https://wiki.asterisk.org/wiki/display/AST/RTP+Packetization |
| 11:59.52 | WIMPy | Neither did I. |
| 12:04.12 | schmidts | wdoekes2 call handlng looks good so far, even a little bit higher than the last time i have tested it. 640 calls per second |
| 12:04.28 | schmidts | i will try it without this patch to see if it depends on this changes or something else ;) |
| 12:05.50 | *** join/#asterisk OlafsenM (~mark.olaf@193.198.31.85) |
| 12:06.01 | OlafsenM | guys |
| 12:06.03 | OlafsenM | need help |
| 12:06.05 | OlafsenM | anyone here? |
| 12:06.18 | WIMPy | ~ask |
| 12:06.18 | 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. |
| 12:07.24 | schmidts | Welcome OlafsenM how can we make you a better day, even its friday :D |
| 12:07.26 | OlafsenM | I get PRI_EVENT_HANGUP_ACK message from telco for a call, it stays alive in Asterisk, but telco says it's hungup |
| 12:07.46 | schmidts | OlafsenM which asterisk version, dahdi version and hardware do you use? |
| 12:07.59 | OlafsenM | 1.6.2.11 asterisk |
| 12:08.19 | OlafsenM | DAHDI Version: 2.2.0.2 |
| 12:08.37 | OlafsenM | sangoma card inside |
| 12:08.52 | *** join/#asterisk sourcode (~code@ppp-58-8-161-125.revip2.asianet.co.th) |
| 12:09.06 | OlafsenM | hmm |
| 12:09.07 | OlafsenM | DAHDI Version: 2.2.1.1 |
| 12:10.03 | schmidts | OlafsenM can you hangup the call with softhangup or channel request hangup or does it still stay |
| 12:10.23 | WIMPy | What is PRI_EVENT_HANGUP_ACK? I only know RELEASE ACK. |
| 12:10.37 | OlafsenM | I can hangup the call |
| 12:11.05 | schmidts | OlafsenM then its not a deadlock problem ;) hmm |
| 12:11.49 | OlafsenM | PRI_EVENT_HANGUP_ACK -> chan_dahdi.c:13516 |
| 12:11.52 | *** join/#asterisk kaii (~kh@ciphron.de) |
| 12:12.36 | OlafsenM | what exactly HANGUP_ACK means? As I understand it, it's a confirmation that my HANGUP_REQ went through at telco side |
| 12:12.40 | OlafsenM | am I right? |
| 12:13.13 | schmidts | olafsenm i am not sure, the warning says Hangup ACK requested on unconfigured channel number |
| 12:13.25 | OlafsenM | I get to bottom line |
| 12:13.37 | WIMPy | o.O |
| 12:15.02 | WIMPy | What warning? |
| 12:15.21 | schmidts | <PROTECTED> |
| 12:16.14 | schmidts | OlafsenM can you reproduce this easily or just happens sometimes? |
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| 12:16.30 | OlafsenM | it happens sometimes |
| 12:16.32 | schmidts | OlafsenM a pri debug whould help to see if you maybe send the hangup |
| 12:16.36 | schmidts | bad |
| 12:19.00 | OlafsenM | :/ |
| 12:19.14 | OlafsenM | i'm not hanging up the call |
| 12:19.17 | OlafsenM | i answer it |
| 12:19.24 | OlafsenM | and after 18 secs HANGUP ACK comes |
| 12:19.27 | OlafsenM | :/ |
| 12:19.32 | schmidts | OlafsenM sorry i meant asterisk not you ;) |
| 12:19.46 | OlafsenM | :P |
| 12:19.51 | OlafsenM | i know what u meant |
| 12:19.52 | OlafsenM | :D |
| 12:22.18 | schmidts | wdoekes2 ok there is not really an improvment of the cps amount with or without this patch, the only thing which really makes things better is the remove of udptl allocation in sip_alloc but i allways use this in my tests too |
| 12:24.46 | wdoekes2 | schmidts: I wasn't expecting an increase in the cps |
| 12:24.53 | wdoekes2 | we're expecting a decrease in mem/udp leaks |
| 12:25.10 | wdoekes2 | in my case udp leaks.. don't know if there is memory leakage as well |
| 12:26.35 | wdoekes2 | I'll be deploying it tonight. hopefully I can remove the daily restart |
| 12:27.31 | schmidts | wdoekes2 yes i know ;) but i wouldnt be happy if this would cause a decrease in cps |
| 12:27.52 | wdoekes2 | ah.. in that case s/.*/ok good!/ |
| 12:28.02 | schmidts | :D |
| 12:28.43 | schmidts | wdoekes2 my target is to reach 1000 cps before the next astricon with the normal trunk version |
| 12:29.40 | schmidts | and atleast nobody take care about the performance at all, thats why before 1.8 every version is slower then the one before ;) |
| 12:29.40 | *** join/#asterisk wonderworld (~ww@port-92-201-223-111.dynamic.qsc.de) |
| 12:32.35 | wonderworld | hehe, the xkcd is sooooooo true |
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| 12:45.58 | *** join/#asterisk nunne (~nunne@static-213-115-116-75.sme.bredbandsbolaget.se) |
| 12:47.11 | nunne | Does anyone know a good way to get prescene/devstate to a cellphone. for example a specific called number over for example a SIP-trunk/account or E1/T1 line. I'm looking into the extension state function. But i'm getting nowhere on how to map that to the hint-priority |
| 12:47.23 | nunne | and obviously hint:ing SIP/trunk/number for example wont work |
| 12:47.53 | nunne | by this trying to get the state of cellphones that always use callthrough through the pbx |
| 12:48.07 | WIMPy | You can't |
| 12:48.43 | WIMPy | Maybe you can use a local cahnnel to at least find out if you have a call to that phone active. |
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| 12:50.10 | *** join/#asterisk sourcode (~code@ppp-58-8-217-24.revip2.asianet.co.th) |
| 12:50.15 | kaii | you could use custom states as kind of a workaround.. set "Custom:mobileXY" state to RINGING or INUSE before the dial and to "NOT_INUSE" on hangup .. additionally, you can use an answer macro (dial option M) to set the custom state to "INUSE"... the hint then references "Custom:mobileXY" |
| 12:50.42 | wdoekes2 | or "${CHANNEL(SIP/trunk/number)}"!="" ? |
| 12:51.08 | kaii | wdoekes2: you can not map that to a hint priority |
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| 13:07.14 | schmidts | wdoekes2 atleast the sip channels and dialogs are released faster ;) |
| 13:10.59 | wonderworld | does anyone have experience with PATTON media gateways? are they as hassle free as promoted? |
| 13:16.16 | *** join/#asterisk garymc (~chatzilla@host81-148-15-59.in-addr.btopenworld.com) |
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| 13:46.23 | schmidts | wonderworld its a pain to get a running config, but then it works very nice |
| 13:47.07 | wonderworld | better than an easy to setup system thats a pain when running :) |
| 13:47.20 | wonderworld | i'd love to test one cos they are quiet expensive to buy |
| 13:50.01 | schmidts | wonderworld which model are you interested in? |
| 13:51.01 | wonderworld | i just wanted it so ee in general. i have customers on E1s but also on EURO ISDN |
| 13:51.20 | wonderworld | digium cards caused many problems for me in long time stability |
| 13:51.36 | wonderworld | so i am looking for somthing that gives me peace of mind |
| 13:52.29 | nunne | WIMPy: have been thinking of a sollution similar to that. But it's not optimal. not in my case anyway :( would be so convinient if you could hint an extension state in the dialplan |
| 13:53.18 | schmidts | wonderworld then patton would be a nice choice. we use the small Bri models with only two Bris (one Network, one User) and the Big 4xE1 model and we have around 200 devices in use without any problems at all |
| 13:55.38 | nunne | first time configuring an E1 card, how do you properly set up extended callerid? been looking at the callerid function, but dont quiet get what should be what :P. Caller A calls -> PBX. PBX then Calls B and the CallerID of number A is shown. This is easy to do with SIP with the operators that support Diversion. But how to hell do I do it with Euro ISDN? The operator supposodly have it turned |
| 13:55.39 | nunne | on. But i cant just change the CallerID to what i wont :/ So i guess i have to set a valid callerid fist and add the original callerid somewhere |
| 13:55.57 | wonderworld | schmidts: nice. you just configure the pstn channel(s) and some sip channels that go directly into asterisk, right? |
| 13:56.23 | schmidts | wonderworld we use a sip proxy in front of two asterisk but yes ;) |
| 13:57.09 | wonderworld | cool. i think i'll buy a single BRI one and test it out. |
| 13:57.16 | wonderworld | thanks |
| 13:57.20 | schmidts | your welcome |
| 13:57.35 | WIMPy | nunne: You don;t have to do anything, just like with any other call. If you can't set it, 1) it's not formatted correctely, 2) no configured correctely in chan_dahdi.conf or 3) not enabled by your te |
| 13:57.40 | schmidts | i guess the smartnode 4552 should be very cheap and a good point to start |
| 13:58.18 | schmidts | wonderworld something really great about pattons you can reuse most of the config like routing and mappings, only the isdn port configuration may change between bri and PRI |
| 14:02.01 | nunne | WIMPy: i dont have to set it? it will get through automatically? even if I have an Answer() in the dialplan? Well, I can set the CallerID to one of "their" numbers. No problem. But when i try something else (using the same syntax) ofcourse their main number is shown. I have talked to the teleco and they say everything is good. And their old ericsson pbx managed to do it. So i dont think i |
| 14:02.01 | nunne | can blame the teleco on this :) |
| 14:03.21 | WIMPy | Unless you change the calller ID it will stay what it was. |
| 14:07.27 | nunne | feel a bit stupid. found useincomingcalleridondahditransfer = yes was commented out in chan_dahdi.conf |
| 14:07.35 | nunne | kinda sounds like the thing i'm looking for ;) |
| 14:09.36 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 14:09.39 | nunne | but didnt work |
| 14:09.58 | schmidts | nunne do you touch the callerid somewhere in your extensions.conf? |
| 14:12.02 | *** join/#asterisk MrTelephone (~MrTelepho@h697179-171.picriverisp.net) |
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| 14:13.18 | WIMPy | Check the number type. |
| 14:14.12 | nunne | schmidts: i read it. but i have tried both setting it to the number i want to show out and not setting it at all |
| 14:14.26 | nunne | maybe i should avoid having an Answer() there as well? |
| 14:14.36 | MrTelephone | I have a dual port t1 card and one port goes to a channel bank and the other goes to the telco. The telco PRI is intermittenantly losing carrier. It shows by doing an ifconfig. The PRI comes into a fiber to copper conversion box and then there is a little CSU/DSU card with a RJ-45 port on it that goes to my asterisk machine. Anyone experience carrier loss from the telco? |
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| 14:18.01 | schmidts | nunne answer only has to do with media handling not with the callerid |
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| 14:20.04 | *** part/#asterisk NixGeek (~NixGeek@216.240.70.66) |
| 14:20.44 | WIMPy | MrTelephone: ifconfig? |
| 14:23.22 | MrTelephone | I just swapped the cable |
| 14:23.40 | MrTelephone | <PROTECTED> |
| 14:23.52 | MrTelephone | uptime is 644 days |
| 14:24.03 | WIMPy | What kind of interface is that? |
| 14:24.11 | MrTelephone | sangoma w1g1 |
| 14:24.44 | MrTelephone | if it was the cable it should be more consistant |
| 14:25.32 | MrTelephone | sometimes it will go for a week without a hitch. I have a feeling its a bad connection from the telco. |
| 14:25.54 | WIMPy | NFI what the Sangoma drivers tell you with the carrier transitions. |
| 14:25.58 | MrTelephone | port 2 goes to an adit 600 about 500 feet away and it has 0 carrier errors. |
| 14:26.31 | MrTelephone | carrier would just be the physical link from the card to the dsu |
| 14:26.37 | WIMPy | Does it ever stay down for any amout of time that lets you take a look? |
| 14:26.57 | nunne | schmidts: with the useincomingcalleridondahditransfer = yes i should not have to set caller id, yes? all i do is basically Dial(DAHDI/G1/numtocall) .. Should i not use Dial? Should i use another app to tell asterisk it's an "transfer"? I know the Transfer() app doesnt work for DAHDI |
| 14:26.58 | MrTelephone | it just stays down long enough to drop calls |
| 14:27.32 | WIMPy | nunne: No, just Dial. |
| 14:28.14 | nunne | and after changing useincomingcalleridondahditransfer = yes a module reload should suffice? or do i have reload the dhadi modules and restart asterisk? |
| 14:28.16 | MrTelephone | I've been ignoring the problem but there are complaints now that people are getting cut off |
| 14:29.08 | WIMPy | nunne: Not sure about a reload, but a module unload, then load will do. |
| 14:30.01 | schmidts | MrTelephone i had this problem just some weeks ago but with 8 Pri lines, all of them got disconnected and get back online after 1 or 2 secs but my telco also had seen this problem on their switch |
| 14:30.32 | MrTelephone | If you take your machine offline that still kills the circuit |
| 14:30.37 | WIMPy | only knows the well known 6 second outages. |
| 14:30.44 | MrTelephone | lol |
| 14:30.48 | MrTelephone | is this a common thing? |
| 14:31.30 | WIMPy | Yes, if your PRI is delivered on a DSL line, but as you're on fiber that can't be your problem. |
| 14:32.13 | MrTelephone | I thought it might be the M6 DSU framing box or whatever it is. |
| 14:33.03 | schmidts | MrTelephone in my case it was an interface card which was killed slowly |
| 14:33.06 | WIMPy | Could even be a shitty PSU. Do you have heavy machinery in the neighbourhood? |
| 14:33.09 | MrTelephone | THere is 20km of fiber and we have an oc-3 coming out of it. I get intermitten packet loss on that circuit as well so I dunno. It would be impossible to get someone to look at it thoroughly I think |
| 14:33.44 | WIMPy | Ah, that will probably have the same cause then. |
| 14:33.45 | MrTelephone | schmidts, a digium/sangoma card or telco card? |
| 14:34.01 | *** join/#asterisk omani (~hasan@33.37.69.80.in-addr.net-lab.net) |
| 14:34.09 | schmidts | MrTelephone on the fiber connection not on my side |
| 14:34.39 | MrTelephone | I'm gonna have to compare the carrier drops to the packet loss and see if I can make a case. The telephone people look at me like I'm nuts when I talk about packet loss. |
| 14:35.32 | MrTelephone | I should reboot the machine after 2 years. Maybe that will help :( |
| 14:35.56 | WIMPy | If the framing on the virtual PRI gets lost, it's likely singnalled by setting the physical i/f down. |
| 14:36.51 | WIMPy | So I'd try to get to the ground of the packet losses. |
| 14:36.54 | MrTelephone | if you lose a few frames out of a call will it drop or just get silence? |
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| 14:37.25 | WIMPy | Depends on where in the data stream it's missing. |
| 14:37.32 | omani | I want to migrate my asterisk project to an existing company communication solution based on conventional telephining with an old pbx system over isdn. we want to expand to native voip communication. I chose a voip concept with asterisk (for voip/sip), ldap (centralized phonebook) and sipgate (as a sip provider/registrar) |
| 14:37.43 | omani | now my question is, how can I take the secure step of migration |
| 14:38.10 | omani | with fallback strategies to the old system, if the we have the case that nothing works anymore. |
| 14:38.11 | WIMPy | If it's missing in the user data you'd get silence, but if it's in the framing signals that might well trigger an interface deactivation. |
| 14:39.13 | omani | in detail, what could we do, to provide connectivity to the outworld in a worst-case scneraio without keeping our old pbx system running |
| 14:40.09 | devmod | Once 1.6.2 becomes Security Fix Only, would there be a 1.6.3? .4 .5 etc till EOL? or how is that determined? |
| 14:40.15 | omani | the whole time. because that's what we want to replace for future. |
| 14:41.27 | *** join/#asterisk b14ck (~b14ck@cpe-72-129-70-245.socal.res.rr.com) |
| 14:41.31 | schmidts | devmod 1.6.2 is the latest branch for 1.6 there will not be a 1.6.3 |
| 14:43.53 | devmod | schmidts, ohh ok got it. How would one know that? is there a site somewhre? |
| 14:44.26 | schmidts | devmod there is, give me a moment to find it |
| 14:45.07 | schmidts | devmod here you go: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions |
| 14:45.47 | devmod | schmidts, that is the one I was looking at but how would I know 1.6.2 is the last branch for 1.6? |
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| 14:46.39 | schmidts | devmod cause 1.8 allready exists |
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| 14:46.59 | Polysics | hello |
| 14:47.03 | devmod | schmidts, ohh ok got it. thanks |
| 14:47.14 | Polysics | is there a "timeout" setting on SIP registrations, please? |
| 14:47.24 | Polysics | i have some clients that "fall off" after a couple hours |
| 14:47.29 | schmidts | and the work on 1.10 allready started which will be a version like 1.6 ;) |
| 14:47.38 | Polysics | no reinvites, nothing |
| 14:47.51 | MrTelephone | poly, that is usually on the endpoint configuration? |
| 14:47.54 | schmidts | polysics yes, its called the registration timeout and normaly is set on the clients phone |
| 14:48.10 | Polysics | then i need to find out where it is on Zoiper Web |
| 14:48.13 | Polysics | ok, thanks |
| 14:48.19 | schmidts | devmod: The type of release defines how long it will be supported. A Long Term Support (LTS) release will be fully supported for 4 years, with one additional year of maintenance for security fixes. Standard releases are supported for a shorter period of time, which will be at least one year of full support and an additional year of maintenance for security fixes. |
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| 14:54.31 | asteriskmonkey | heya im running asterisk 1.4.39.1, and im having an odbc voicemail issue, i have everything setup correctly however im not seeing the box make any attempt to access the db.. I see the odbc is listed as connected, just no voicemail users are showing and no quries from ngrep.. |
| 14:58.40 | nunne | schmidts: one "funny" thing is that, what is not working is call through to the cellphones. ie. someone calls the landline number and is directed to the phone. BUT what is working is when a cellphone calls into the mobile extension number and i after that set the callerid to a land number.. THEN it works.. and i can manipulate it exactly how i want. i can even set it to 123 if i want.. so.. |
| 14:58.40 | nunne | how come it works in one case, and not the other? :/ can it be because it's to smart? when i'm testing now i'm trying to call myself. so it recognizes that the callerid is the same as the called number and forces the "main number" on? |
| 14:59.01 | MrTelephone | asteirskmonkey: did you make the edits to voicemail.conf? odbcstorage = asterisk |
| 14:59.01 | MrTelephone | odbctable = voicemail_messages |
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| 15:00.42 | WIMPy | nunne: Maybe your cellphone provider doesn't accept user provided numbers? |
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| 15:05.29 | asteriskmonkey | MrTelephone: yes |
| 15:05.49 | asteriskmonkey | MrTelephone: odd thing is i cant see any users either... |
| 15:06.20 | asteriskmonkey | MrTelephone: everything connects great from the command line, but it seems asterisk dosnt want to query the db :P ... |
| 15:07.33 | MrTelephone | i'm still using 1.2.24 |
| 15:07.44 | MrTelephone | I mean 1.4.24 |
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| 15:09.25 | MrTelephone | are you using ngrep on the asterisk box? |
| 15:09.41 | asteriskmonkey | MrTelephone: yes ive tried it locally and remote |
| 15:10.08 | MrTelephone | try to login remotely with manually with the mysql -h voicemailserver -u asterisk -p |
| 15:10.23 | MrTelephone | my english is terrible :( |
| 15:10.32 | asteriskmonkey | MrTelephone: login to db is fine |
| 15:11.06 | MrTelephone | was it working in earlier versions? |
| 15:11.58 | MrTelephone | did you select OBDC under voicemail build options when doing your 'make menuselect'? |
| 15:12.16 | drift- | hrm |
| 15:17.29 | MrTelephone | if that doesn't work try and compile 1.4.24.1 or something just to see if you can get it working. I was going to upgrade to 1.4.39.1 and now you have me freaked out |
| 15:17.30 | MrTelephone | I have 1.4.33 on my backup machine and it seems to be working |
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| 15:24.49 | wonderworld | is there a more simple voicemail app for asterisk than the included one? |
| 15:25.36 | wonderworld | perfect would be to just being able to hear messages. go forward and backward and delete them. all with keys. |
| 15:25.51 | wonderworld | without long anouncements |
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| 15:30.09 | omani | wonderworld: that's what im searching for too |
| 15:30.33 | omani | a nice short voicemail menu |
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| 15:30.50 | omani | just for hearing messages, delete and step for/backward |
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| 15:52.50 | puzzled | hi |
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| 16:38.59 | *** join/#asterisk ccesario (~ccesario@189-29-59-116-ac.cpe.vivax.com.br) |
| 16:39.20 | *** part/#asterisk sekil (~sekil@80.93.247.26) |
| 16:41.15 | *** join/#asterisk FlashDeluxe (~benedict@static-87-79-94-28.netcologne.de) |
| 16:42.15 | *** join/#asterisk TheDavidFactor (~chatzilla@nc-71-52-235-146.dhcp.embarqhsd.net) |
| 16:42.57 | FlashDeluxe | hi@ all! How can i find out which file is used for my voicemail? There is an old server which is not running anymore and which is not connected and my customer wants to have his old announcement for his voicemail and i don`t know how the soundfile is named :( |
| 16:49.05 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 16:49.09 | *** join/#asterisk b14ck (~b14ck@cpe-72-129-70-245.socal.res.rr.com) |
| 16:53.57 | kaii | :q |
| 16:55.43 | *** join/#asterisk tvc123 (~tcameron@host-148.pl1071220.fiber.net) |
| 16:58.37 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-84-109-243-175.red.bezeqint.net) |
| 17:10.48 | *** join/#asterisk pp07 (~fedup@182.71.111.126) |
| 17:11.29 | pp07 | hello |
| 17:11.31 | pp07 | <PROTECTED> |
| 17:11.50 | pp07 | help me plz... |
| 17:13.03 | *** join/#asterisk rubberneck (~chatzilla@ext-52.sagetelecom.net) |
| 17:13.33 | rubberneck | Is there a way to bridge 2 existing calls via AMI or the console? |
| 17:13.52 | pp07 | <PROTECTED> |
| 17:14.49 | pp07 | <PROTECTED> |
| 17:14.50 | pp07 | [Feb 10 14:03:59] > Channel Local/8600051@default-6ed6,1 was answered. |
| 17:14.50 | pp07 | [Feb 10 14:03:59] == Starting Local/8600051@default-6ed6,1 at default,91148279 1785,1 failed so falling back to exten 's' |
| 17:14.50 | pp07 | [Feb 10 14:03:59] == Starting Local/8600051@default-6ed6,1 at default,s,1 stil l failed so falling back to context 'default' |
| 17:14.50 | pp07 | [Feb 10 14:03:59] -- Sent into invalid extension 's' in context 'default' on Local/8600051@default-6ed6,1 |
| 17:14.50 | pp07 | [Feb 10 14:04:00] -- Executing [i@default:1] Playback("Local/8600051@default -6ed6,1", "invalid") in new stack |
| 17:14.51 | pp07 | [Feb 10 14:04:00] -- <Local/8600051@default-6ed6,1> Playing 'invalid' (langu age 'en') |
| 17:14.52 | pp07 | [Feb 10 14:04:00] WARNING[25394]: file.c:1292 waitstream_core: Unexpected contro l subclass '-1' |
| 17:14.53 | pp07 | [Feb 10 14:04:04] == Manager 'sendcron' logged off from 127.0.0.1 |
| 17:14.54 | pp07 | [Feb 10 14:04:05] == Auto fallthrough, channel 'Local/8600051@default-6ed6,1' status is 'UNKNOWN' |
| 17:14.54 | pp07 | [Feb 10 14:04:05] -- Executing [h@default:1] DeadAGI("Local/8600051@default- 6ed6,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0-------- -------") in new stack |
| 17:14.57 | pp07 | CLI output |
| 17:15.08 | pp07 | plz help me out anyone |
| 17:15.24 | pp07 | why it dails number with 91 prefix |
| 17:15.25 | pp07 | ?? |
| 17:15.31 | rubberneck | Are you kidding me |
| 17:15.35 | rubberneck | use pastebin |
| 17:15.43 | pp07 | my dialplan starts with _944. |
| 17:16.00 | *** join/#asterisk ketas-l (~ketas@ketas-laptop6.si.pri.ee) |
| 17:17.50 | *** join/#asterisk b14ck (~b14ck@cpe-72-129-70-245.socal.res.rr.com) |
| 17:18.14 | pp07 | no one ?? |
| 17:18.23 | pp07 | <PROTECTED> |
| 17:18.23 | pp07 | [Feb 10 14:03:59] > Channel Local/8600051@default-6ed6,1 was answered. |
| 17:18.24 | pp07 | [Feb 10 14:03:59] == Starting Local/8600051@default-6ed6,1 at default,91148279 1785,1 failed so falling back to exten 's' |
| 17:18.24 | pp07 | [Feb 10 14:03:59] == Starting Local/8600051@default-6ed6,1 at default,s,1 stil l failed so falling back to context 'default' |
| 17:18.24 | pp07 | [Feb 10 14:03:59] -- Sent into invalid extension 's' in context 'default' on Local/8600051@default-6ed6,1 |
| 17:18.24 | pp07 | [Feb 10 14:04:00] -- Executing [i@default:1] Playback("Local/8600051@default -6ed6,1", "invalid") in new stack |
| 17:18.24 | pp07 | [Feb 10 14:04:00] -- <Local/8600051@default-6ed6,1> Playing 'invalid' (langu age 'en') |
| 17:18.24 | pp07 | [Feb 10 14:04:00] WARNING[25394]: file.c:1292 waitstream_core: Unexpected contro l subclass '-1' |
| 17:18.25 | pp07 | [Feb 10 14:04:00] WARNING[25394]: file.c:1292 waitstream_core: Unexpected contro l subclass '-1' |
| 17:18.26 | pp07 | [Feb 10 14:04:04] == Manager 'sendcron' logged off from 127.0.0.1 |
| 17:18.26 | pp07 | [Feb 10 14:04:05] == Auto fallthrough, channel 'Local/8600051@default-6ed6,1' status is 'UNKNOWN' |
| 17:18.27 | pp07 | [Feb 10 14:04:05] -- Executing [h@default:1] DeadAGI("Local/8600051@default- 6ed6,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0-------- -------") in new stack |
| 17:18.33 | *** kick/#asterisk [pp07!~north@pdpc/sponsor/digium/Qwell] by Qwell (pastebin.com) |
| 17:18.51 | WIMPy | Stop flooding the channel |
| 17:19.08 | *** join/#asterisk pp07 (~fedup@182.71.111.126) |
| 17:19.37 | pp07 | hey is u can't asnswer do kick me off |
| 17:19.39 | pp07 | ... |
| 17:19.49 | pp07 | i am just asking... solution.. |
| 17:19.52 | pp07 | <PROTECTED> |
| 17:19.52 | pp07 | [Feb 10 14:03:59] > Channel Local/8600051@default-6ed6,1 was answered. |
| 17:19.52 | pp07 | [Feb 10 14:03:59] == Starting Local/8600051@default-6ed6,1 at default,91148279 1785,1 failed so falling back to exten 's' |
| 17:19.52 | pp07 | [Feb 10 14:03:59] == Starting Local/8600051@default-6ed6,1 at default,s,1 stil l failed so falling back to context 'default' |
| 17:19.52 | pp07 | [Feb 10 14:03:59] -- Sent into invalid extension 's' in context 'default' on Local/8600051@default-6ed6,1 |
| 17:19.52 | pp07 | [Feb 10 14:04:00] -- Executing [i@default:1] Playback("Local/8600051@default -6ed6,1", "invalid") in new stack |
| 17:19.53 | pp07 | [Feb 10 14:04:00] -- <Local/8600051@default-6ed6,1> Playing 'invalid' (langu age 'en') |
| 17:19.53 | pp07 | [Feb 10 14:04:00] WARNING[25394]: file.c:1292 waitstream_core: Unexpected contro l subclass '-1' |
| 17:19.54 | pp07 | [Feb 10 14:04:04] == Manager 'sendcron' logged off from 127.0.0.1 |
| 17:19.55 | pp07 | [Feb 10 14:04:05] == Auto fallthrough, channel 'Local/8600051@default-6ed6,1' status is 'UNKNOWN' |
| 17:19.55 | pp07 | [Feb 10 14:04:05] -- Executing [h@default:1] DeadAGI("Local/8600051@default- 6ed6,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0-------- -------") in new stack |
| 17:19.55 | *** kick/#asterisk [pp07!~north@pdpc/sponsor/digium/Qwell] by Qwell (pastebin.com) |
| 17:19.59 | *** join/#asterisk pp07 (~fedup@182.71.111.126) |
| 17:20.01 | Qwell | If you can't follow our rules, leave. |
| 17:20.05 | Qwell | ~pb |
| 17:20.06 | infobot | [~pb] 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/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 17:20.08 | chazzam | pp07 stop flooding |
| 17:20.19 | *** join/#asterisk csnook (~chris@c-76-19-64-161.hsd1.ma.comcast.net) |
| 17:20.36 | pp07 | i am not flooding just posting CLI output... |
| 17:20.59 | pp07 | if u don't know then shut up ok.. |
| 17:21.10 | chazzam | that is flooding |
| 17:21.24 | pp07 | ohh ya... |
| 17:21.30 | pp07 | then plz tell me |
| 17:21.33 | ketas-l | where's water |
| 17:21.38 | pp07 | how to post CLI output??? |
| 17:21.42 | Qwell | ~pb |
| 17:21.42 | infobot | [~pb] 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/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 17:21.51 | Qwell | for the 4th time. |
| 17:23.21 | pp07 | i am in windows.. so can't use that.. |
| 17:23.27 | chazzam | ... |
| 17:23.36 | pp07 | plz tell me the solution if u can.... |
| 17:23.41 | chazzam | you don't have a browser or internet access because you are in windows? |
| 17:24.04 | chazzam | if you don't have internet then I can't see what you're writing. :) |
| 17:24.15 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 17:24.29 | *** join/#asterisk Tim_Toady (~moi@178.128.24.211) |
| 17:27.06 | pp07 | don't try to be oversmart... now i am 100% sure u know nothing about ....asterisk.... |
| 17:27.21 | _Sam-- | pp07: that wont get you much help. |
| 17:27.28 | chazzam | I am half amazed now that you are here, since you apparently don't have a browser to download mIRC with |
| 17:27.51 | _Sam-- | what they are saying is that if you want help, you have to paste your CLI output into something like pastebin.com |
| 17:27.55 | chazzam | psss, windows ships with a browser, its called Internet Explorer |
| 17:27.57 | _Sam-- | then anyone here can check it out and help you. |
| 17:28.15 | Qwell | _Sam--: but nobody will now, because he's an ass. :) |
| 17:28.30 | Qwell | welcome to free support forums. people don't have to help you. |
| 17:31.04 | *** join/#asterisk linuxplatform (~centoslin@143-77-9.connect.netcom.no) |
| 17:31.05 | *** join/#asterisk chasing`Sol (~rc4@smtp.master-zone.net) |
| 17:33.51 | *** join/#asterisk Tim_Toady (~moi@178.128.24.211.dsl.dyn.forthnet.gr) |
| 17:36.48 | pp07 | how do i get reply after pasing my error in pastbin? |
| 17:36.51 | pp07 | http://paste.lisp.org/display/119584 |
| 17:37.17 | pp07 | reply me if anyone... want to help me... out |
| 17:37.29 | *** join/#asterisk KamikazeMicrowav (~contegix@seraph.contegix.com) |
| 17:38.55 | chazzam | looks like you are sending dialplan to an extension that doesn't exist |
| 17:40.16 | chazzam | 8600051@default-6ed6,2 doesn't appear to exist, so it is failing whatever you are trying to do |
| 17:41.10 | *** join/#asterisk ront (~Adium@194-230-89-225.static.adslpremium.ch) |
| 17:41.55 | *** join/#asterisk Defraz (~Defraz@tim.spudnik.com) |
| 17:42.20 | *** part/#asterisk omani (~hasan@33.37.69.80.in-addr.net-lab.net) |
| 17:43.38 | pp07 | ok but manual is working fine.... how? |
| 17:55.43 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 17:55.43 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.2.3 (2010/01/26), 1.6.2.16.1 (2010/01/18), 1.4.39.1 (2010/01/18), *-Addons 1.6.2.3, 1.4.13 (2010/01/14), dahdi-linux 2.4.0 (2010/09/01), dahdi-tools 2.4.0 (2010/09/01), libpri 1.4.11.5 (2010/11/22) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
| 17:55.46 | *** join/#asterisk Jasnejac (kvirc@81.91.107.236) |
| 18:00.00 | *** join/#asterisk flujan (~flujan@189.111.254.251) |
| 18:01.18 | flujan | hello guys. I got a core.dump and now I am using gdb and trying to figure out what cause the dump... |
| 18:01.20 | flujan | I am running the command : thread apply all bt |
| 18:01.41 | flujan | How can I identify which thread cause the dump? |
| 18:04.32 | *** join/#asterisk McBoingbo (~mcboingbo@mail.hrsg.ca) |
| 18:05.54 | McBoingbo | we have 8 lines available to us with some pay as you go line toll free numbers that redirect to our landline number, my problem is trying to filter out which lines coming in originated from the toll free line so that I can properly monitor our 8 internal lines, clear as mud? thanks! |
| 18:13.19 | *** join/#asterisk s519 (~steve@119.42.87.193) |
| 18:15.06 | *** join/#asterisk Sipster (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
| 18:16.05 | *** join/#asterisk Defraz (~Defraz@tim.spudnik.com) |