| 00:03.06 | SeRi | p3nguin: ouch |
| 00:03.34 | SeRi | are this avail under the subscription? or just limited in general? |
| 00:03.43 | p3nguin | Strriiiiike twooooo! |
| 00:03.57 | p3nguin | It seems to be a general limitation. |
| 00:04.16 | *** join/#asterisk tekoholic (~quassel@97-118-242-155.hlrn.qwest.net) |
| 00:04.31 | SeRi | ouch in general than |
| 00:04.45 | p3nguin | What's worse is that I am configuring it exactly the way one of the Vyatta guys says to configure it, and it still isn't working for me. |
| 00:05.28 | p3nguin | I should probably examine the firewall to make sure that's not stopping it. |
| 00:06.17 | SeRi | +1 |
| 00:12.26 | *** join/#asterisk voipeng (~tom@c-69-242-127-126.hsd1.pa.comcast.net) |
| 00:15.19 | *** join/#asterisk meest (~alpaca@155-23-88-118.dcwest.net.au) |
| 00:18.32 | *** part/#asterisk CheeseMouse (~cheddar@2001:470:1f10:1023::2) |
| 00:19.00 | pdtpatrick | is this valid ? ... NoOp(Incoming from ${CALLERID}) ?? thanks |
| 00:20.04 | *** join/#asterisk godmachine-x6 (~poorelanc@funtoo/user/godmachine-x6) |
| 00:20.06 | [TK]D-Fender | pdtpatrick: No, that is a 1.0 var. Unless you are running such a tragically old system you should be using the function that was created with 1.2 "core show function CALLERID" |
| 00:20.26 | SeRi | Verbose(0,${CALLERID(num)} is calling your home) |
| 00:20.37 | pdtpatrick | sweet thanks |
| 00:20.58 | SeRi | n/p |
| 00:21.25 | *** join/#asterisk godmachine-x6 (~poorelanc@funtoo/user/godmachine-x6) |
| 00:21.38 | *** join/#asterisk eharris (~eharris@99-179-7-82.lightspeed.austtx.sbcglobal.net) |
| 00:24.50 | pdtpatrick | Another question .. which is better route in terms of improvements to the dialplan? AEL or LUA? I see AEL has better document in asterisk wiki whereas LUA is spotty |
| 00:25.53 | *** part/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 00:27.17 | [TK]D-Fender | pdtpatrick: AEL offers less that standard dialplan. |
| 00:28.04 | pdtpatrick | interesting |
| 00:28.10 | pdtpatrick | thanks for that |
| 00:28.53 | *** part/#asterisk Neptu (~jromero@213-67-244-97-no100.tbcn.telia.com) |
| 00:30.16 | [TK]D-Fender | Not sure about LUA. I thought I saw something about it being an alternative direct method to access things at a lower level than dialplan but never really followed up on it. |
| 00:31.53 | pdtpatrick | was just asking because i noticed with LUA and AEL u could do something like switch statements which are similar to GotoIfs ..but personally i perfer the current asterisk format. Was just asking if that's the direction asterisk heading (more LUA and/or AEL). |
| 00:34.11 | [TK]D-Fender | AEL only gets parsed back to extensions.conf logic so it can only do less |
| 00:34.23 | [TK]D-Fender | Can't speak for LUA though |
| 00:34.30 | *** join/#asterisk pdtpatrick_ (~pdtpatric@ip72-211-207-15.oc.oc.cox.net) |
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| 00:57.33 | *** join/#asterisk IsUp (IsUp@unaffiliated/isup) |
| 00:57.44 | IsUp | hello |
| 00:59.01 | SeRi | whats going on IsUp |
| 01:00.11 | dijib | howcome this dont worky? echo "select 1" | isql -v asterisk-connector |
| 01:00.35 | dijib | [28000][unixODBC]FATAL: Ident authentication failed for user "asterisk" |
| 01:00.35 | dijib | [ISQL]ERROR: Could not SQLConnect |
| 01:00.58 | dijib | Starting postgresql service: [ OK ] |
| 01:01.15 | dijib | DB has been created, and connect though console |
| 01:01.21 | dijib | can connect |
| 01:01.40 | IsUp | SeRi: problems, works. nothing much |
| 01:02.01 | SeRi | same shit different day ;) |
| 01:04.13 | IsUp | exactly |
| 01:04.23 | SeRi | lol |
| 01:05.41 | SeRi | dijib: -vvv for more output |
| 01:14.51 | *** join/#asterisk wesphillips (~wphill04@adsl-75-53-136-233.dsl.hstntx.sbcglobal.net) |
| 01:15.01 | *** part/#asterisk wesphillips (~wphill04@adsl-75-53-136-233.dsl.hstntx.sbcglobal.net) |
| 01:18.32 | *** join/#asterisk Jenkens (48b605d5@gateway/web/freenode/ip.72.182.5.213) |
| 01:21.12 | *** join/#asterisk clarkster48 (6303ad20@gateway/web/freenode/ip.99.3.173.32) |
| 01:21.25 | clarkster48 | Hello, I need a little bit of quick help on an Asterisk and Cisco SPA525G phone question |
| 01:21.48 | Jenkens | jabs clarkster48 |
| 01:24.09 | clarkster48 | so I have a SPA525G phone that has 2 extensions on it, I have them joined by 1 voicemail account (Phone 1 at my office has x701 for example, Phone 2 at home has x751, but I want voicemailbox 701 to work for 751). I have it setup in FreePBX for the system to display a message notification on my cisco, but what I am trying to do now is setup the voicemail button on my SPA525G to directly connect to voicemail box 701 at home. |
| 01:25.51 | *** join/#asterisk ferdna (~yup@cpe-67-10-220-35.elp.res.rr.com) |
| 01:25.57 | WIMPy | ~freepbx |
| 01:25.58 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 01:29.14 | clarkster48 | Thanks guys |
| 01:29.24 | *** part/#asterisk Jenkens (48b605d5@gateway/web/freenode/ip.72.182.5.213) |
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| 02:28.53 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 03:01.54 | SeRi | p3nguin: any visio like for linux? |
| 03:02.41 | GGD | question for anyone here can asterisk be used for a VOIP trunk provider? |
| 03:05.51 | [TK]D-Fender | SeRi: Dia |
| 03:06.00 | [TK]D-Fender | GGD: Sure |
| 03:07.22 | GGD | it can [TK]D-Fender? |
| 03:07.40 | SeRi | [TK]D-Fender: I used dia. to basic... well just not so up to date... ill give it a run again. |
| 03:08.22 | SeRi | Guys any easy way to look up a number? I recived a call and there is no cid anem on it and I want to see who those the number belong too. |
| 03:11.36 | *** join/#asterisk hariom (~hariom@117.195.160.115) |
| 03:12.47 | hariom | I am trying to create an AGI application. In that i am recording few recording few answers. I find that after 3 recordings, on the 4th recording, I am getting this error: 200 result=-1 (randomerror) endpos=0 |
| 03:12.58 | hariom | What could be the reason? |
| 03:14.24 | hariom | Anybody who can put insights on this problem? |
| 03:18.50 | [TK]D-Fender | SeRi: http://freecnam.org/ |
| 03:19.01 | hariom | Playing 'beep.slin' (language '/xxxx/xxxx/xxxx/xxxx') Retransmission timeout reached on transmission qhxmadgslntzmtd@ubuntu for seqno 702 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions |
| 03:19.01 | hariom | Packet timed out after 32000ms with no response |
| 03:19.01 | hariom | Hanging up call qhxmadgslntzmtd@ubuntu - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions) |
| 03:19.06 | SeRi | [TK]D-Fender: Thanks! |
| 03:19.13 | p3nguin | seri: There are several places to look up numbers. If you'll give me the numbers, I'll search, too. |
| 03:20.14 | *** part/#asterisk GGD (~deberle@pool-173-72-204-39.clppva.fios.verizon.net) |
| 03:22.18 | hariom | can anybody help? |
| 03:25.50 | hariom | My problem is same as mentioned here: http://www.safisystems.com/index.cfm?pageMode=messages&threadid=3AB15C02-DA2B-798F-8174A1281FEE5EAD |
| 03:28.21 | hariom | How to upgrade my asterisk 1.8 to the latest |
| 03:28.41 | hariom | latest in 1.8 release. I have 1.8.5 |
| 03:49.57 | *** join/#asterisk kikohnl (~kotis@udp019436uds.hawaiiantel.net) |
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| 04:05.48 | Sean-Der | Does anyone know of any sort of reporting frontend to CEL? |
| 04:08.12 | *** join/#asterisk LiENUS (quake@209.168.144.219) |
| 04:16.57 | dijib | anybody run pgsql? |
| 04:28.18 | *** join/#asterisk voipeng (~tom@c-69-242-127-126.hsd1.pa.comcast.net) |
| 04:28.28 | SeRi | dijib: whats going? |
| 04:29.19 | voipeng | Getting this error when trying to install zaptel, all yum updates show i have the latest You do not appear to have the sources for the 2.6.18-274.12.1.el5PAE kernel installed. |
| 04:31.47 | voipeng | i ran through this process fine in our lab, went to upgrade this after hours tonight and i cant get through the install, since i get that error everytime i make zaptel 1.4.12.1 |
| 04:38.00 | p3nguin | Lots of people use postgresql. |
| 04:46.42 | voipeng | could anyone lend a hand pleas? really would like to get production back up before tomorrow morning |
| 04:53.11 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 04:53.11 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 04:59.02 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 04:59.02 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 05:01.01 | SeRi | p3nguin: Tomorrow is run day. I am going to sketch which route I am going to take for teh univ. So far a real time db looks like what we will do for the long term solution. |
| 05:01.45 | SeRi | for now a small test env arch/* will do to let the fac/staff/students test |
| 05:02.28 | voipeng | was able to get past that error, but now asterisk wont start at all.. ERROR[4572]: asterisk.c:3087 main: Asterisk has detected a problem with your Zaptel configuration and will shutdown for your protection. status 1 code 1 |
| 05:08.26 | SeRi | http://www.voipjet.com/ <---- another sunk ship :/ |
| 05:12.14 | p3nguin | You've just about got to use real time when you're going to be adding and removing new phones with the frequency you'll be doing it. |
| 05:16.39 | SeRi | p3nguin: you are right. The web dev will look in to my request during the new year. the date is not set but is goign to be withing Jan 2012. |
| 05:16.57 | SeRi | s/withing/within/ |
| 05:18.08 | SeRi | so digium will be selling their own sip phones soon? |
| 05:18.48 | SeRi | Wait a couple of months and you will see a line of phones from Digium. Not just Asterisk compatible, but optimal means of integration with Asterisk. <---- from mgraves |
| 05:23.26 | WIMPy | So they found out why everybody else builds their own phones with their own protocols? |
| 05:24.15 | SeRi | lol |
| 05:26.42 | SeRi | is there a way to find out if I have fax detect turn on in asterisk cli? |
| 05:26.59 | SeRi | Sometimes the calls come in get answered by a fax reciving tone.... |
| 05:34.25 | p3nguin | It's not the astcli, but: grep faxdetect /etc/asterisk/sip.conf|grep -v ^\; |
| 05:36.36 | SeRi | I did that and si not in place. |
| 05:36.55 | SeRi | I am just wondering why I am getting those random faxt like tones when calling in |
| 05:44.31 | *** join/#asterisk cerberus_za (~coert@41-134-110-162.dsl.mweb.co.za) |
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| 06:59.37 | *** join/#asterisk joren (~joren@97-115-71-224.ptld.qwest.net) |
| 07:00.10 | joren | Hey, is there an asterisk guru in here that could spare something like probably a half hour of their time to just do this for me? |
| 07:00.22 | joren | I think it could work, but I'm running down the wire right now. |
| 07:00.39 | joren | it's for a wireless audio system for a portland occupy event on monday. |
| 07:00.47 | *** join/#asterisk voipeng (~tom@70.44.203.146.res-cmts.brd2.ptd.net) |
| 07:00.53 | joren | If anyone has time to donate, it'd be appreciated. |
| 07:01.52 | voipeng | heh not sure i can help, having problems getting my zaptel drivers back online |
| 07:03.21 | joren | What I want is a very simple config: I want a meetme room, with all the "mics" but one muted by the server. All of the users should be able to connect anonymously if possible (not important) the server will have a sip client running that auths with a real user/password. and it's the only one that's allowed to transmit on the channel. |
| 07:03.24 | joren | Is this possible? |
| 07:03.49 | joren | My brain is a bit fried right now, and I need to work on the wireless hardware. |
| 07:04.02 | joren | I also don't have that good of a grasp on asterisk |
| 07:04.13 | joren | I also played with mumble. it worked well-ish |
| 07:04.20 | joren | I could play with that some more |
| 07:04.34 | joren | voipeng, what probably are you having? |
| 07:04.40 | joren | I *might* be able to help.. |
| 07:05.02 | voipeng | ERROR asterisk.c: Asterisk has detected a problem with your Zaptel configuration and will shutdown for your protection. |
| 07:05.02 | voipeng | You have options: |
| 07:05.15 | voipeng | can only start asterisk once i stop zaptel first |
| 07:05.18 | joren | if somone wanted to just do this for me, you would be my hero, and your work would get tested/used, and published. |
| 07:05.31 | *** join/#asterisk oej (~olle@87.96.134.129) |
| 07:05.32 | joren | no kernel errors? |
| 07:05.37 | joren | in dmesg? |
| 07:05.44 | voipeng | whats the command to check? |
| 07:06.06 | joren | dmesg |grep zaptel I would imagine |
| 07:06.07 | joren | second |
| 07:06.12 | joren | is oej my hero? :) |
| 07:06.36 | voipeng | nothing from that command dmesg | grep zap |
| 07:06.42 | joren | voipeng, Mine's using something called dahdi |
| 07:06.51 | joren | you might just dissable zaptel in your config |
| 07:06.58 | joren | dahdi might be a replacement |
| 07:07.11 | joren | I'm running on fumes, so take it with a grain of salt |
| 07:07.14 | voipeng | i need virtual timing to work, our software repo installs it off the bat |
| 07:07.34 | voipeng | ive had more problems tryin to go from one to the other than making zaptel work |
| 07:07.53 | voipeng | i ran through updating the driver on my lab a few times and it worked fine |
| 07:08.05 | joren | zaptel mostly always just worked for me. |
| 07:08.10 | voipeng | i tried production and yea... it loaded just something screwed up |
| 07:08.18 | joren | are your sources solid? |
| 07:08.33 | joren | version missmatch maybe? |
| 07:08.39 | joren | somone else probably knows. |
| 07:09.07 | joren | This config of mine has to be stupid simple |
| 07:09.14 | joren | asterisk is too much for what I want |
| 07:09.15 | joren | maybe |
| 07:09.29 | joren | I wonder if mumble can mute remotes |
| 07:09.35 | joren | that |
| 07:09.41 | joren | 's the only thing that was broken |
| 07:09.47 | joren | turning on the mic on an android phone |
| 07:09.57 | joren | multiple mics" |
| 07:10.03 | joren | but I only want one input |
| 07:10.25 | joren | I must be high or something |
| 07:10.33 | joren | my brain is racing :D |
| 07:10.42 | joren | isn't high |
| 07:10.45 | *** join/#asterisk LiuYan (~liu.yan@211.154.128.171) |
| 07:10.47 | joren | afaik |
| 07:11.36 | joren | Anyone local to pdx with a 1-5w fm transmitter would be VERY appreciated too. We're looking to buy |
| 07:11.56 | joren | this is long term. meet me is something I'm trying to cobble together for tomorrow |
| 07:11.59 | joren | on short order |
| 07:12.02 | joren | I have the hardware. |
| 07:12.13 | *** join/#asterisk AmirBehzad (~behzad@87.248.136.180) |
| 07:12.46 | joren | I'll get up on the mic and mention your name :D if you want |
| 07:12.47 | joren | :P |
| 07:13.20 | joren | is mic shy |
| 07:14.38 | joren | no takers? I need like 3 config files I think. What 3 do I need? |
| 07:14.52 | joren | sip.conf, I figured I would put the users in |
| 07:15.13 | joren | but putting users in there seems to make everything anonymous |
| 07:16.03 | joren | voipeng, see what the module is called, and grep for that. |
| 07:16.12 | joren | /var/log/syslog too |
| 07:18.13 | voipeng | nadda/./ |
| 07:19.27 | joren | I've got nada then |
| 07:19.29 | joren | sorry |
| 07:19.39 | joren | check your versions. |
| 07:19.48 | joren | are you using an actual zaptel card? |
| 07:19.55 | joren | what os? |
| 07:19.56 | voipeng | nope |
| 07:19.59 | voipeng | cent os 5 |
| 07:20.15 | voipeng | just need virtual timing for meetme |
| 07:20.20 | joren | my brain is too add for this, I should sleep :D |
| 07:20.26 | joren | uhm |
| 07:20.42 | voipeng | hah me 2 but if i dont get the server up by morning... ill have a lot more time to sleep hah |
| 07:20.51 | joren | core... redhat? |
| 07:21.13 | voipeng | hmm? centos 5.7 |
| 07:21.52 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-wkatyedqshuumcbp) |
| 07:21.56 | joren | rum --help |
| 07:22.08 | joren | find the package that installed zaptel |
| 07:22.21 | joren | I have no idea, I might be leading you down a wrong road. |
| 07:22.37 | joren | goes back to work |
| 07:23.27 | joren | god, it does NOT help that my fucking tab complete inserts a sp after every DIRECTORY |
| 07:23.29 | joren | wtf |
| 07:24.08 | joren | vodka might not be helping either. |
| 07:26.04 | joren | root@j-laptop:/etc/asterisk# man ../asterisk-old/festival.conf why did that command succeed? |
| 07:26.11 | joren | weird. |
| 07:28.15 | *** join/#asterisk mandla (~mandla@168.167.180.161) |
| 07:33.58 | voipeng | oh forgot to mention then errors Asterisk ended with exit status 1 |
| 07:33.59 | voipeng | Asterisk died with code 1. |
| 07:39.30 | *** join/#asterisk ollii (~risker@2001:470:1f15:1384:62eb:69ff:fe31:b4) |
| 07:39.53 | *** join/#asterisk jkroon (~jkroon@196.25.195.42) |
| 07:40.22 | jkroon | hi guys, how can I get hold of the ConnectedLineIDNum value from DumpChan in my dialplan? |
| 07:40.24 | *** join/#asterisk timahvo1 (~rogue@41.80.124.113) |
| 07:47.06 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 07:47.08 | schmidts | good morning |
| 07:49.01 | ollii | g'ornin |
| 07:50.08 | joren | good fucking night. |
| 07:50.32 | joren | is shmidts or ollii an asterisk guru? :D |
| 07:50.39 | joren | I need an occupy favor |
| 07:50.53 | joren | I don't think it would take much |
| 07:54.15 | schmidts | joren it depends what you need ;) |
| 07:54.34 | joren | I need a couple dozen lines of config |
| 07:54.38 | joren | by tomorrow morning |
| 07:54.45 | joren | lemmy copy paste |
| 07:54.48 | schmidts | ok |
| 07:54.50 | joren | not so sober. |
| 07:55.53 | joren | <joren> What I want is a very simple config: I want a meetme room, with all the "mics" but one muted by the server. All of the users should be able to connect anonymously if possible (not important) the server will have a sip client running that auths with a real user/password. and it's the only one that's allowed to transmit on the channel. |
| 07:56.21 | joren | it's not a big deal, but it would be cool to have |
| 07:56.26 | joren | with the hardware that I have. |
| 07:57.02 | joren | (laptop(server+source), plugged into a bullet, with a 20ish db antenna. it'd be helpful in our efforts. |
| 07:58.58 | joren | phones as voip clients |
| 07:59.17 | joren | your work will get used, published, creddited, if you want |
| 07:59.35 | joren | I can give you ssh into my "server" and remote hands. |
| 07:59.53 | joren | sorry for everyone in here that's heard me repeat the same shit |
| 08:00.18 | joren | is hating on softer right now |
| 08:00.25 | joren | loving working with hardware |
| 08:03.07 | joren | or, I need you to tell me which files are mandatory. |
| 08:03.30 | joren | which files I should focus on, which sections I should focus on changing. |
| 08:03.32 | *** join/#asterisk Wiretap7 (~Wiretap@unaffiliated/wiretap) |
| 08:03.49 | *** join/#asterisk waschtl (~waschtl@HSI-KBW-078-043-090-014.hsi4.kabel-badenwuerttemberg.de) |
| 08:03.53 | joren | or do you have any aderol? :D |
| 08:03.58 | joren | that's a joke. |
| 08:04.20 | *** join/#asterisk s[X] (~mark@ppp118-208-26-44.lns20.bne1.internode.on.net) |
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| 08:09.52 | *** join/#asterisk e7e5 (~rudenko@188.134.2.33) |
| 08:10.36 | IsUp | hi |
| 08:12.37 | *** join/#asterisk freeedrich| (friedrich@2a01:4f8:130:2023:1:151:0:babe) |
| 08:13.20 | joren | hi |
| 08:13.34 | joren | IsUp, want's to help occupy? |
| 08:13.50 | joren | I'm sorry, I shoudln't be spamming |
| 08:14.03 | IsUp | joren: whats up |
| 08:14.21 | joren | "What I want is a very simple config: I want a meetme room, with all the "mics" but one muted by the server. All of the users should be able to connect anonymously if possible (not important) the server will have a sip client running that auths with a real user/password. and it's the only one that's allowed to transmit on the channel. " |
| 08:14.33 | joren | I have hardware. I need lines of config |
| 08:14.39 | joren | on probably too short order |
| 08:15.39 | joren | mics refers to receivers, cell phones. |
| 08:15.46 | joren | the server wants to transmit |
| 08:16.01 | joren | Tomorrow. |
| 08:16.39 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 08:18.32 | IsUp | joren: do 'core show application MeetMe' in your CLI. probably you need 'l, t, q' options |
| 08:19.47 | IsUp | joren: adding a meetme room is easy, go to your meetme.conf, and see examples. 'conf => 1111', this will create a room called "1111". so you can run MeetMe(1111,q) on your dialplan. |
| 08:20.43 | joren | I don't have a brain for this right now, unfortunately. but within the next month I'd like to getit going |
| 08:20.57 | joren | unless you're willing to hack a bit tonight :P |
| 08:20.59 | IsUp | joren: also, you need a timing source for MeetMe. if you dont have any timing source (hardware), you'll need dahdi_dummy. so make sure you have installed dahdi first. i dont have much experience on conference, so i cant help you :> |
| 08:21.16 | joren | I have dahdi. |
| 08:21.32 | *** join/#asterisk TimeRider (~steve@92.40.247.208.threembb.co.uk) |
| 08:21.46 | joren | a timer, I'm not sure if it's needed |
| 08:22.07 | joren | We're expecting delay to the outputs. |
| 08:22.24 | joren | I'm not sure that syncing that delay relay matters |
| 08:22.32 | joren | but that's usefull info info |
| 08:22.40 | joren | I'll try to cram it in somewhere :P |
| 08:23.45 | joren | is there a cross platform timing source? |
| 08:24.08 | IsUp | joren: what you mean? |
| 08:24.12 | ChannelZ | pthreads |
| 08:24.41 | joren | something that would time on linux as well as android. and account for wifi delay |
| 08:24.49 | ChannelZ | 'modern' versions of asterisk will build timerfd and pthreads as timing sources and use what it can |
| 08:24.51 | joren | honestly, a delay might be usefull |
| 08:24.56 | *** join/#asterisk bulkorok (~bulkorok@85.183.36.36) |
| 08:25.17 | joren | to prevent feedback |
| 08:25.42 | joren | if somone could tell me what configs to focus on, I could do the work, or stumble through it |
| 08:26.36 | ChannelZ | for what now? just doing a meetme conf? |
| 08:27.47 | joren | I know I need: 1 verified user (sip.conf) , unlimited anonymous users (?), the meetme chan (meetme.conf+extensions.conf,) the server itself (asterisk.conf) I need a flag to mute the anon guys as well. |
| 08:28.07 | ChannelZ | oh so you have nothing setup? |
| 08:28.34 | joren | ChannelZ, yes, mostly just a meetme conf. I've got a phone connected :P I've focused too much on hardware |
| 08:29.15 | ChannelZ | IsUp pretty much gave you all you need above. |
| 08:29.43 | joren | ok, I'll try. |
| 08:29.53 | joren | I have more hardware to build too thouhg. |
| 08:30.36 | joren | I have no "core" command |
| 08:30.56 | ChannelZ | uhmm.. what? |
| 08:31.05 | IsUp | joren: try without 'core', which version youre running? |
| 08:31.12 | joren | idk, the second thing he said, I'll ignore it :) |
| 08:32.28 | *** join/#asterisk Nasga (~Nasga@112.4.118.78.rev.sfr.net) |
| 08:32.33 | *** join/#asterisk CaneToad (~CaneToad@c122-108-198-52.rochd4.qld.optusnet.com.au) |
| 08:32.49 | joren | ubuntu 11.10 |
| 08:33.03 | joren | or 11.04 |
| 08:33.04 | IsUp | joren: * version? |
| 08:33.20 | joren | asterisk version. |
| 08:33.44 | joren | 1.6.2.9 |
| 08:33.47 | CaneToad | I've got a situation with asterisk where it keeps sending REGISTER requests, and getting a 401 Unauthorized, but none of the REGISTER requests include an Authorization field. Any ideas why it would not contain that...the password is definitely specified in the register command |
| 08:34.05 | CaneToad | I've check the password with the server using an ATA and it works fine |
| 08:34.20 | IsUp | CaneToad: enable SIP debugging and pastebin your output |
| 08:36.22 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
| 08:41.22 | CaneToad | IsUp: http://fpaste.org/Ajrb/ |
| 08:41.55 | IsUp | CaneToad: looks like a NAT issue |
| 08:42.02 | CaneToad | IsUp: I have snooped the traffic and I can see the 401 Unauthorized reply coming back |
| 08:42.11 | joren | I'm over working myself right now. I'll come back to it, and probably come back to the chan. thanks for the knowledge I got. |
| 08:42.32 | IsUp | CaneToad: umm i cant see 401 replies in your paste |
| 08:42.35 | *** join/#asterisk domi (~domi@mail.tas.de) |
| 08:42.50 | joren | how are you fukin snnoopin trafic |
| 08:42.56 | joren | that's kinda cool :P |
| 08:43.00 | CaneToad | joren: tcpdump |
| 08:43.12 | CaneToad | then loading it into wireshark |
| 08:43.13 | joren | easy enough. |
| 08:43.44 | IsUp | CaneToad: do you have externip= and localnet= set in your sip.conf? |
| 08:46.30 | *** join/#asterisk asterisk-Tester (~RAMYT@193.227.170.247) |
| 08:46.32 | wdoekes2 | CaneToad: if you see the 401 coming back, is that on the same port as the one you sent from? do you see it on the same host? or on the nat router? |
| 08:47.04 | CaneToad | yup it is... take a look at http://www.aaa.net.au/campbell/snoop.png |
| 08:48.12 | wdoekes2 | CaneToad: do you have a firewall on your host? |
| 08:48.54 | CaneToad | wdoekes2: you'd think that must be it wouldn't you.... |
| 08:48.56 | CaneToad | I'll go look |
| 08:49.01 | CaneToad | what is set up |
| 08:49.07 | wdoekes2 | iptables -nvL |
| 08:50.18 | CaneToad | <PROTECTED> |
| 08:50.40 | CaneToad | on chain INPUT |
| 08:50.43 | *** join/#asterisk bjornts (~BTS@it010226.klientdrift.uib.no) |
| 08:51.59 | wdoekes2 | (1) you haven't told use whether the tcpdump was made on the same host or the router and (2) your iptables says 0 packets.. so a different rule hits it earlier (if (1) is same-host), pb the whole thing |
| 08:52.42 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
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| 08:53.41 | CaneToad | the asterisk is running on the router (dd-wrt) and the tcpdump was on that device |
| 08:54.04 | CaneToad | getting called to dinner... |
| 08:54.14 | CaneToad | thanks guys, be back later |
| 08:54.52 | wdoekes2 | hm.. that changes things.. eth1 on INPUT will not see packets from outside, since it's natted |
| 08:56.15 | wdoekes2 | (a) if you're on the world-device, why are you sending out an internal IP to the outside.. ast_ouraddrfor() should detect the right source IP |
| 08:57.20 | wdoekes2 | (b) if you're not fixing (a), you'll see the packets on different chains (FORWARD and -t nat PREROUTING) |
| 08:57.28 | wdoekes2 | (I think) |
| 09:00.31 | *** join/#asterisk kriegerod (~krieger@chaf.tenet.odessa.ua) |
| 09:00.38 | jkroon | on SendFAX - is it supposed to sit for minutes (even hours) waiting for the remote end to initiate fax tone? |
| 09:01.12 | jkroon | channel is Up, and has been for 227 seconds, remote end is sitting in Wait(300), after which it will hang up, but still, feels odd that SendFAX hasn't already given up |
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| 09:09.54 | mandla | [Dec 12 10:52:06] WARNING[3344]: app_dial.c:1098 wait_for_answer: Unable to forward voice |
| 09:10.36 | mandla | Whats the possible cause of this problem? |
| 09:10.49 | schmidts | Does anyone of you use Patton smartnode devices? |
| 09:13.03 | *** join/#asterisk krotos (~d3v1l@host211-3-dynamic.5-87-r.retail.telecomitalia.it) |
| 09:13.13 | krotos | hi all |
| 09:20.57 | *** join/#asterisk Azrael808 (~peter@cpc17-walt12-2-0-cust657.13-2.cable.virginmedia.com) |
| 09:20.58 | IsUp | hi krotos |
| 09:21.30 | IsUp | mandla: i have same messages everyday, if you find a solution, tell me :p |
| 09:21.36 | IsUp | mandla: are you running 1.4? |
| 09:22.10 | mandla | 1.6 |
| 09:22.20 | mandla | IsUp, 1.6 |
| 09:23.00 | *** join/#asterisk jacc0 (~jacc0@D522448D.static.ziggozakelijk.nl) |
| 09:23.08 | mandla | IsUp, But can you make and receive calls with that message present? |
| 09:23.09 | krotos | Hi IsUp :) |
| 09:23.29 | IsUp | mandla: yes, all calls are working fine. |
| 09:23.31 | jacc0 | hi all |
| 09:23.52 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 09:24.06 | jacc0 | I have this astdb that is causing a segfault on asterisk startup |
| 09:24.39 | krotos | try to start * with core-dump? |
| 09:24.46 | krotos | for debugging |
| 09:26.36 | *** join/#asterisk s[X] (~mark@ppp118-208-26-44.lns20.bne1.internode.on.net) |
| 09:29.37 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 09:30.09 | jkroon | res_fax.c:1200 set_fax_t38_caps: channel 'Local/27867209000@colt45-outbound-2f01;1' is in an unsupported T.38 negotiation state, cannot continue. |
| 09:30.09 | jkroon | is there anybody that might be able to assist with that? |
| 09:30.39 | jkroon | remote peer goes into ReceiveFAX, on this end WaitFAX() succeeds with T38 and then the immediately following SendFAX fails with that. |
| 09:31.14 | mandla | IsUp, Channel 0/1, span 1 got hangup, cause 18. |
| 09:31.48 | mandla | IsUp, And that one ever run across it? How did you resolve it? |
| 09:31.57 | jkroon | mandla, version of asterisk? |
| 09:32.17 | jkroon | has seen that crap with 1.6.1, not with >=2.6.2 |
| 09:32.30 | jkroon | *1.6.2 |
| 09:32.50 | mandla | jkroon, Asterisk 1.6.2.11 |
| 09:32.55 | jkroon | odd ... |
| 09:33.25 | mandla | jkroon, very odd... |
| 09:33.42 | jkroon | channel types involved? |
| 09:34.30 | mandla | jkroon, channel types?? |
| 09:34.45 | jkroon | sip? dahdi? iax/2? |
| 09:36.01 | IsUp | im having this on DAHDI |
| 09:36.52 | mandla | Its dahdi. |
| 09:36.56 | IsUp | here is an output: http://pastebin.com/raw.php?i=yh89ZEXY |
| 09:38.11 | IsUp | my scenerio: customer (ss7/sip) -> calling my main pbx for gsm gateway (iax2) -> dialing agent (dahdi) |
| 09:38.39 | IsUp | mandla: whats your traffic load? are you processing too much calls? |
| 09:39.29 | *** join/#asterisk s[X] (~mark@ppp118-208-26-44.lns20.bne1.internode.on.net) |
| 09:39.34 | mandla | IsUp, Cant get calls and cant make calls. |
| 09:40.15 | mandla | http://pastebin.com/6DbYzek5 |
| 09:40.40 | IsUp | oh, ive noticed that also my calls are not working as you see in pastebin. i didnt really care it. this only happens 1-2 times in a day. |
| 09:40.45 | IsUp | strange |
| 09:41.49 | IsUp | mandla: are you using PRI? was it working before? |
| 09:41.58 | mandla | yes. |
| 09:42.10 | mandla | And yes it was working before. |
| 09:42.42 | *** join/#asterisk timahvo1 (~rogue@41.80.73.70) |
| 09:43.09 | mandla | I dont know, whether the problem is with the Telkom service provider or it with my asterisk server, or with the NT boxex. |
| 09:43.58 | IsUp | mandla: you can enable PRI debugging with 'pri intense debug span <span>', and pastebin your output maybe we can find something |
| 09:44.24 | IsUp | mandla: but first of all, i suggest you to talk to your telco. they can reset their side. |
| 09:45.35 | jkroon | mandla, version of libpri? |
| 09:45.37 | jkroon | and of dahdi |
| 09:48.23 | mandla | http://pastebin.com/Ru4zZqUd |
| 09:49.40 | IsUp | mandla: did you place a call after enabling debug? |
| 09:49.53 | mandla | No. |
| 09:50.11 | IsUp | mandla: okay place a call, these are normal ACK messages. place a call and paste output |
| 09:52.02 | *** join/#asterisk s[X] (~mark@ppp118-208-26-44.lns20.bne1.internode.on.net) |
| 09:54.54 | mandla | http://pastebin.com/V8MiH5zK |
| 09:58.14 | krotos | hey, somehone here use opensips/other sip router with ast? |
| 09:58.41 | *** join/#asterisk maxhbp204 (~chatzilla@122.170.71.104) |
| 09:58.44 | krotos | i'm little bit curious in this possible solutions |
| 09:59.44 | maxhbp204 | Hi All, can any one provide me chan_ss7.conf or ss7.conf example for Digium TE420B, I want to configure SS7 with this device and want to send calls with that line |
| 10:01.22 | *** join/#asterisk s[X] (~mark@ppp118-208-26-44.lns20.bne1.internode.on.net) |
| 10:02.11 | IsUp | maxhbp204: hello, im running chan_ss7, i'll pm u in 2 mins |
| 10:02.37 | mandla | IsUp, ?? |
| 10:02.43 | IsUp | mandla: ive cmopared a normal call with your debug. i think problem is telco side. |
| 10:03.04 | IsUp | mandla: also, did you changed any config? |
| 10:03.28 | mandla | IsUp, No, never changed the configs. |
| 10:04.35 | *** join/#asterisk turtlefence (~turtlefen@c58-111-144-182.thorn2.nsw.optusnet.com.au) |
| 10:05.44 | maxhbp204 | IsUP: yes thanks, i am waiting for your help |
| 10:07.47 | IsUp | mandla: okay and your caller id looks empty, "Presentation permitted, user number not screened (0) '' ] ---> this looks strange. |
| 10:08.34 | mandla | IsUp, So the problem its the telco side. |
| 10:08.51 | IsUp | mandla: yes, it looks so. |
| 10:09.08 | mandla | IsUp, Ok, thanx man. |
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| 10:46.19 | Ionic` | Shit, I'm mentally retarded and stupid |
| 10:46.33 | Ionic` | Is there any way to get the current extensions.conf out of a running asterisk instance? |
| 10:46.43 | Ionic` | i.e. dump the dialplan or something |
| 10:47.04 | Ionic` | Yes, I actually deleted that file... |
| 10:50.27 | jkroon | asterisk -rx "dialplan show" ? |
| 10:50.52 | jkroon | you will need to clean it up, but at least you'll have something to work from |
| 10:52.39 | Ionic` | Ah |
| 10:52.53 | Ionic` | Wait, thank god for terminal buffers |
| 10:53.24 | jkroon | lol, or redirectors, asterisk -rx "dialplan show" > dialplan.txt |
| 10:53.55 | Ionic` | Nah I found the config in my buffer |
| 11:02.00 | kaii | congrats |
| 11:02.34 | jkroon | Ionic`, lucky :) |
| 11:02.40 | Ionic` | Hmm... |
| 11:02.43 | Ionic` | " |
| 11:02.55 | Ionic` | Uh |
| 11:02.59 | Ionic` | "NOTE:On some ISDN lines if you use isdnmode=MSN, it is mandatory to also set immediate=yes. This applied to me after changing to the German Telko "Hansenet" aka "Alice". I had strange behaviour with immediate=no: Calls from other customers of my Telko where answered by asterisk, while those from mobiles and others Telkos where silently ignored" |
| 11:03.11 | Ionic` | Interesting, found on http://www.voip-info.org/wiki/view/Asterisk+How+to+connect+with+CAPI |
| 11:03.18 | Ionic` | Let's see if I have that |
| 11:04.32 | *** join/#asterisk skrusty (~ksrusty@62.252.24.138) |
| 11:04.41 | *** part/#asterisk skrusty (~ksrusty@62.252.24.138) |
| 11:04.44 | kaii | sounds like a misconfiguration on their ISDN switch .. fun |
| 11:04.49 | Ionic` | Hm ok |
| 11:05.14 | Ionic` | Sounds like it forces Asterisk to process the call even if not all information are gathered yet |
| 11:05.37 | *** join/#asterisk skrusty (~ksrusty@62.252.24.138) |
| 11:05.42 | Ionic` | I am on Hansenet/Alice/O2 but haven't yet noticed such behavior |
| 11:05.58 | skrusty | morning all |
| 11:06.51 | Ionic` | Ok, I hope some comment in capi.conf is enough to remind me in case I see dropped calls |
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| 12:30.05 | jkroon | is there a hard limit on the number of Local/ channels? |
| 12:31.26 | jkroon | doesn't look that way from the code, so going to assume no. |
| 12:32.33 | tzafrir | There's a global limit on the number of channels, though |
| 12:32.53 | tzafrir | What do you need this for? |
| 12:33.00 | jkroon | tzafrir, thanks for that heads up - you wouldn't happen to know what that limit is? |
| 12:33.19 | jkroon | channel origination is messing me around combined with CDRs and the GROUP() functions. |
| 12:33.23 | tzafrir | By default: none. You can set it in asterisk.conf, IIRC |
| 12:34.23 | jkroon | easy fix is to just keep adding Local/ channels however, expecting 150+ "real" calls and at a current cascade level of 4 worst case this can get quite nasty. would prefer to try and get rid of as many of them as I can but there is more urgent stuff that needs to happen first. |
| 12:51.01 | jkroon | what is the difference between $[] and ${MATH()} ?? |
| 12:53.03 | *** join/#asterisk verywiseman (~verywisem@unaffiliated/verywiseman) |
| 12:57.27 | schmidts | jkroon a $[?]?is imho a logical result but MATH() gives you a math result |
| 12:59.10 | jkroon | schmidts, ok ... was wondering about that, core show function MATH indicates it too has some "logical" operators ... |
| 12:59.24 | verywiseman | i install asterisk-gui , and i did all instruction in README file , after i run `make checkconfig`, it said OK , and inform me to open this link "http:/ip:8088/asterisk/static/config/index.html" , and when i opened it , there is no page found , but i noticed that , the asterisk-gui content was installed in /var/lib/asterisk/static-http , so when i opened this link "http://ip:8088/asterisk/static-http/config/index.html" also page not fou |
| 12:59.24 | verywiseman | nd , why? |
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| 13:03.49 | schmidts | jkroon maybe there isnt really a difference but i wouldnt trust a simple $[ ] enough for real complex math operations, but you can use it for something simple like $[ 5 + 3 ] i guess |
| 13:04.23 | jkroon | thanks schmidts - i tend to agree with your assessement |
| 13:04.24 | *** join/#asterisk darkskiez_ (~dz@62-50-207-166.client.stsn.net) |
| 13:04.27 | schmidts | verywiseman is your apache or any other webserver running? |
| 13:04.51 | verywiseman | schmidts, no |
| 13:05.12 | schmidts | verywiseman i havent ever used the GUI but imho you will need a webserver for it |
| 13:05.22 | IsUp | jkroon: MATH() funcs is more complex, thats the difference i think. |
| 13:05.50 | verywiseman | schmidts, as i read , asterisk has one |
| 13:06.12 | verywiseman | schmidts, where i connect to 8088 port , and asterisk bind to it |
| 13:06.14 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 13:06.48 | schmidts | verywiseman ok, as i said i have never used it ;) but btw is your asterisk running or have you restarted it after the installation? |
| 13:09.08 | verywiseman | yes |
| 13:10.55 | schmidts | verywiseman do you have telnet installed on the server where your gui is installed? what do you get back when you try a telnet localhost 8088 |
| 13:11.39 | jkroon | verywiseman, check http[d?].conf - you probably need to explicitly enable it. |
| 13:15.34 | *** join/#asterisk fprior (be0bbc86@gateway/web/freenode/ip.190.11.188.134) |
| 13:15.53 | verywiseman | jkroon, are you mean /etc/httpd/conf/httpd.conf ? |
| 13:16.11 | *** join/#asterisk lcat (~lcat@187.45.254.89) |
| 13:16.44 | jkroon | no, i mean /etc/asterisk/http.conf |
| 13:17.05 | *** join/#asterisk AmirBehzad (~behzad@87.248.136.180) |
| 13:19.27 | fprior | Hi all, I'm playing with odbc. Nos response from test select in func_odbc.conf . No information in CLI, how can I debug odbc ? |
| 13:19.28 | verywiseman | jkroon, it is enabled |
| 13:20.15 | jkroon | ok, so netstat -ntlp shows it running? haven't worked with ast-GUI really, seen it like once, and left it at that. |
| 13:20.18 | verywiseman | jkroon, look, asterisk bind to 8088 , and no firewall , and this message appear "The requested URL was not found on this server." |
| 13:20.41 | jkroon | ok, so your browser is connecting to the server, but asterisk is giving a 404 on the path. |
| 13:20.53 | jkroon | you are using :8088/path right? |
| 13:21.01 | verywiseman | jkroon, yes |
| 13:21.05 | verywiseman | jkroon, but |
| 13:21.42 | verywiseman | in README file it is said go to http://ip:8088/asterisk/static/index.html |
| 13:22.32 | verywiseman | but there is not /var/lib/asterisk/static directory , there is /var/lib/asterisk/static-http , and when i open http://172.31.0.10:8088/asterisk/static-http/index.html same message appeared |
| 13:23.17 | *** join/#asterisk PoWeRKiLL (~powerkill@77.125.245.91) |
| 13:24.05 | verywiseman | jkroon, another issue , i make index.html file in /var/lib/asterisk , and when open http://172.31.0.10:8088/asterisk/index.html , also same message appeared |
| 13:24.20 | jkroon | try dropping the /asterisk/ out of there |
| 13:24.29 | jkroon | ie, just ip:8088/static/index.html |
| 13:24.58 | jkroon | I believe /static/ maps ONTO /var/lib/asterisk/static-http/ |
| 13:25.37 | verywiseman | jkroon, same message |
| 13:25.50 | verywiseman | jkroon, in http.conf i put prefix=asterisk |
| 13:25.55 | jkroon | give me a few seconds. |
| 13:25.59 | verywiseman | ok |
| 13:26.15 | jkroon | you have enablestatic in http.conf? |
| 13:27.13 | jkroon | http show status gives me this line which is crucial: /static/... => Asterisk HTTP Static Delivery |
| 13:30.32 | verywiseman | yes i enabled it |
| 13:32.46 | jkroon | hmm, not working for me either. |
| 13:32.48 | jkroon | very strange. |
| 13:33.35 | jkroon | nm, didn't enable http: http://localhost:8088/static/astman.js works for me. |
| 13:33.52 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 13:34.05 | jkroon | enabled=yes; enablestatic=yes and bindaddr is the settings I've got in [general] of http.conf |
| 13:34.56 | jkroon | and /static/ goes to /var/lib/asterisk/static-http/ |
| 13:36.22 | *** join/#asterisk box (~box@unaffiliated/box) |
| 13:36.23 | box | hi |
| 13:36.36 | box | does anyone know how to connect to mysql? |
| 13:36.47 | box | i made a lot of test calls and want to remove those numbers from the database |
| 13:37.10 | jkroon | box, mysql -u user -p -h hostname dbname |
| 13:39.27 | box | jkroon thanks |
| 13:39.55 | box | according to `show databases` i only have information_schema and test |
| 13:40.18 | jkroon | then you are connecting to the wrong host |
| 13:45.43 | *** join/#asterisk corretico (~luis@190.211.94.6) |
| 13:47.18 | *** part/#asterisk bulkorok (~bulkorok@85.183.36.36) |
| 13:47.40 | box | jkroon thank you bro |
| 13:47.46 | verywiseman | jkroon, then what can we do? |
| 13:49.03 | fprior | my trouble with odbc + asterisk: No records from test "select 1", no errors in CLI, howto debug odbc ? |
| 13:50.36 | [TK]D-Fender | fprior, I'd start by showing us everything you've got... |
| 13:53.46 | IsUp | hey [TK]D-Fender |
| 13:55.08 | verywiseman | [TK]D-Fender, would you please help me in this issue: i install asterisk-gui , and i did all instructions in README file , after i run `make checkconfig`, it said OK , and inform me to open this link "http:/ip:8088/asterisk/static/config/index.html" , and when i opened it , this message appeared "The requested URL was not found on this server." , but i noticed that , the asterisk-gui content was installed in /var/lib/asterisk/static-htt |
| 13:55.09 | verywiseman | p , so when i opened this link "http://ip:8088/asterisk/static-http/config/index.html" also same message appear , what is problem? |
| 13:55.31 | fprior | yes, [TK-D-Fender]: http://pastebin.com/raw.php?i=VCfXqxBj |
| 13:55.34 | [TK]D-Fender | verywiseman, I don't use asterisk-gui |
| 13:57.32 | [TK]D-Fender | fprior, go connect to your dsn directly with isql |
| 13:57.45 | [TK]D-Fender | fprior, also "readsql=SELECT '234'" doesn't loko like legitmate syntax |
| 13:57.50 | [TK]D-Fender | No table name there |
| 14:00.02 | [TK]D-Fender | fprior, and you aren't adding () at the end of your function reference which makes it loko like you are calling a variable, not a function. |
| 14:02.30 | fprior | [TK]D-Fender: yes, "echo "select 234" | isql -v asterisk-connector" response 234 |
| 14:02.58 | [TK]D-Fender | fprior, fix the function reference syntax |
| 14:03.20 | fprior | [TK]D-Fender, where add () ? |
| 14:03.44 | [TK]D-Fender | to your function. |
| 14:03.56 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-zxtukodsnvcijyqb) |
| 14:05.25 | *** join/#asterisk serafie (~erin@nat/digium/x-uaoiazewjxiytkoo) |
| 14:08.26 | *** join/#asterisk voipeng (~voipeng45@75-150-128-2-Philadelphia.hfc.comcastbusiness.net) |
| 14:08.54 | fprior | [TK]D-Fender, correct: the problem was missing (), no problem with a Select without from table. Thanks |
| 14:09.09 | [TK]D-Fender | fprior, You're welcome |
| 14:12.31 | *** join/#asterisk puzzled (~patrick@2001:980:5e31:1:6ef0:49ff:fe50:659d) |
| 14:15.02 | *** join/#asterisk sereal-work (~sereal@unaffiliated/sereal) |
| 14:15.20 | sereal-work | is there a channel for asterisk2billing? |
| 14:15.35 | ollii | contact areski via mail |
| 14:15.59 | sereal-work | I don't need that much response. I was just hopping to get a bit of help |
| 14:21.09 | SeRi | p3nguin: voip.ms fixed the hard limit isue. now when is set to 0.00 or -1.00 call just get fully cut off and no charges will happen. |
| 14:21.18 | SeRi | s/isue/issue/ |
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| 14:54.20 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 15:05.45 | *** join/#asterisk asterisk-Tester (~RAMYT@193.227.170.247) |
| 15:10.55 | *** join/#asterisk asteriskmonkey1 (~darlek@74-51-38-204.telnetcommunications.com) |
| 15:19.09 | IsUp | i cant see ".call" files actions in my log, any ideas? i have this in my logger.conf "121211_full.log => notice,warning,error,verbose,debug,dtmf" |
| 15:19.20 | IsUp | im logging -almost- everything |
| 15:19.51 | *** join/#asterisk zerohalo (~zerohalo@74-94-141-204-ma.hfc.comcastbusiness.net) |
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| 15:21.01 | pabelanger | IsUp: pbx_spool.so loaded? |
| 15:21.43 | IsUp | umm yeah, im processing call files so often. probably 200-250 call files per day. i was looking for something and somehow call file is created but asterisk never tried to process it. |
| 15:22.03 | IsUp | so i know call file is created, but i cant see anything for call progress in log. |
| 15:23.17 | IsUp | also i have "exten => failed" in my context. because im executing a deadAGI there. |
| 15:23.24 | asteriskmonkey1 | is there anyway to make asterisk realtime actually work realtime? ie without having to have pointers in the diaplan context? |
| 15:25.15 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 15:26.21 | *** part/#asterisk AmirBehzad (~behzad@87.248.136.180) |
| 15:32.33 | IsUp | pabelanger: ok somehow asterisk didnt processed call file, but call file is not there. it's deleted. |
| 15:32.58 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 15:33.22 | IsUp | any ideas? |
| 15:34.05 | *** join/#asterisk stope (~stope@sud-cable-cmts3-69-60-247-14.vianet.ca) |
| 15:34.49 | *** join/#asterisk smellis (~smellis@2001:470:c452:2:210:18ff:fe55:c589) |
| 15:35.43 | stope | I keep struggling with this error and it results in one way audio: Call from 'custA-230' (74.59.221.53:29483) to extension 'custA.domain.net:5060' rejected because extension not found in context 'custA' |
| 15:36.03 | stope | or just dead air as they say.... |
| 15:37.10 | Kobaz | stope: dialplan 101 |
| 15:37.47 | Kobaz | custA-230 is calling the extension (phone number, etc) 'custA.domain.net:5060' and it doesn't exist in the context 'custA' |
| 15:38.02 | Kobaz | you can fix it one of two ways |
| 15:38.10 | Kobaz | you can have custA-230 call an extension that actually exists |
| 15:38.31 | Kobaz | or you can add 'custA.domain.net:5060' as an extension in the context 'custA' which probably isn't what you want |
| 15:38.54 | Kobaz | the error message contains everything you need in order to fix the problem |
| 15:42.06 | Ziaeon | drem yol lok |
| 15:43.47 | *** join/#asterisk telnettech (~telnettec@216.49.139.56) |
| 15:43.59 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 15:44.01 | stope | ok, thanks Kobaz |
| 15:44.03 | telnettech | Morning all......... |
| 15:44.15 | *** join/#asterisk wcselby (~wcselby@pdpc/supporter/professional/wcselby) |
| 15:44.16 | wcselby | o/ |
| 15:44.24 | [TK]D-Fender | <PROTECTED> |
| 15:44.48 | Kobaz | coughs |
| 15:45.08 | telnettech | If a person puts a call block for a particular number on their phone, wouldnt there be a SIP message that is out of the ordinary?????? |
| 15:45.23 | telnettech | when that number calls them????? |
| 15:45.26 | Kobaz | ? |
| 15:45.33 | Kobaz | why would it be out of the ordinary? |
| 15:45.57 | telnettech | well it wouldnt be the normal messaging on a simple call |
| 15:46.25 | telnettech | wouldnt there be some type of 486-Busy Here message or something like that |
| 15:47.04 | [TK]D-Fender | telnettech, block for a particular number? Do you mean for a particular individual call? What is the phone talking to? where are you expecting to see a difference? Where is * in this picture? |
| 15:47.29 | *** join/#asterisk anonymouz666 (~anonymouz@189.25.79.187) |
| 15:47.37 | *** part/#asterisk smellis (~smellis@2001:470:c452:2:210:18ff:fe55:c589) |
| 15:47.45 | [TK]D-Fender | telnettech, Is "block" meant to mean "block CallerID"? Or are you implying some sort of selective DND? |
| 15:48.19 | telnettech | person makes a call from 223-555-5555 and they are calling 456-555-5555.... person at 456-555-5555 doesnt want to receive calls from 223-555-5555 so they block any calls from them |
| 15:48.44 | Kobaz | depending how you are blocking them, they would hear ring no answer or get a busy |
| 15:49.08 | Kobaz | but you're not giving any specifics |
| 15:49.13 | telnettech | so in SIP wouldnt there be some message that would be in the call trap to indicate that it wasnt someone on the line and cant receive another call |
| 15:49.24 | Kobaz | busy or indinite ringing |
| 15:49.27 | Kobaz | infinite |
| 15:49.44 | telnettech | it appears to be infinite ringing |
| 15:50.05 | [TK]D-Fender | telnettech, Please clarify exactly what "block" mechanism you are referring to |
| 15:50.48 | telnettech | I dont want to get calls from you so I block your number from calling me. |
| 15:51.11 | [TK]D-Fender | How? |
| 15:51.22 | [TK]D-Fender | On what? |
| 15:51.34 | telnettech | I can do some star code thru the telco that keeps them from sending your call to my phone |
| 15:51.57 | [TK]D-Fender | telnettech, Well if its at a telco end then it should eb invisible to you |
| 15:52.12 | [TK]D-Fender | And God only knows what they choose to make it look like to the other side |
| 15:52.51 | telnettech | I am the Telco.....But I am looking for a SIP message that would indicate that the person put the calling party on a call block list |
| 15:53.04 | [TK]D-Fender | telnettech, Got something to show us? |
| 15:53.33 | Kobaz | telnettech: you can read the rfc for sip response codes |
| 15:53.45 | telnettech | That is the Thing....What would the SIP message be to indicate that scenario |
| 15:54.08 | tzanger | morning |
| 15:54.21 | telnettech | I thought it would be a 603-Call Blocked or 486-Busy Here |
| 15:54.40 | telnettech | something of that nature |
| 15:55.13 | telnettech | sorry 603-Decline |
| 15:57.12 | *** join/#asterisk fisted_ (~fisted@unaffiliated/fisted) |
| 15:58.14 | telnettech | finally something popped up on my google search and it is a 403-Forbidden |
| 15:59.58 | [TK]D-Fender | that is an auth refusal normally |
| 16:00.22 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 16:06.40 | skrusty | quick one if anyone has any pointers, just setup an asterisk 1.8 box and enabled a queue, except i hear no announcements or MOH - call connects fine, and MOH works in an active call... any suggestions? |
| 16:07.33 | wcselby | skrusty how are you calling the Queue() (which options are you using) and what does the Queue definition look like in queues.conf? |
| 16:07.49 | wcselby | I'm gonna be in a meeting here in about 5 minutes |
| 16:09.15 | stope | Kobaz, here is a trace of the problem. And it's for an incoming call, that's the part that I don't get... http://pastebin.com/Zm7JjGCr |
| 16:09.42 | skrusty | it's the standard markq |
| 16:09.52 | stope | If it was an outgoing call, I would understand... |
| 16:10.07 | skrusty | and i am just calling Queue(markq) |
| 16:11.51 | wcselby | as far as I know there is no "standard markq" |
| 16:12.01 | wcselby | there's a commented out example named markq |
| 16:12.08 | wcselby | but you need to let me know what you've uncommented |
| 16:12.13 | *** join/#asterisk brdude (~brdude@c-24-130-20-142.hsd1.ca.comcast.net) |
| 16:12.30 | skrusty | two mo's.... |
| 16:12.45 | skrusty | (and yes, i was refering to the commented out [markq] |
| 16:12.46 | skrusty | ) |
| 16:13.17 | wcselby | two mo's..... ? |
| 16:13.32 | wcselby | do this |
| 16:13.37 | wcselby | in the asterisk cli |
| 16:13.45 | wcselby | type "queue show markq" and pastebin the output |
| 16:13.48 | wcselby | ~pb |
| 16:13.48 | 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.us , 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. |
| 16:14.09 | skrusty | two mo's - two moment |
| 16:14.09 | skrusty | s |
| 16:15.00 | *** join/#asterisk afink (~afink@wsip-184-187-15-226.om.om.cox.net) |
| 16:16.19 | skrusty | wcselby: http://pastebin.com/nQ2KzNYJ |
| 16:17.15 | skrusty | cli shows it's playing MOH and playing announcements, but no audio to handset (until agent pickup, and then two way audio happens as expected) |
| 16:17.38 | *** part/#asterisk hemantvoip (~hemant@203.122.46.246) |
| 16:17.51 | wcselby | what does the dilaplan look like for this call? is there an Answer() anywhere before you make the Queue() call? |
| 16:18.01 | afink | anyone know of any decent free asterisk call reporting packages? I'm not looking for anything earth moving just call totals and such. |
| 16:18.21 | skrusty | ah, you may have just hit the nail on the head! :) two mo(ments) |
| 16:18.41 | *** join/#asterisk Maliuta (~nobusines@kiev.lusan.id.au) |
| 16:18.56 | jrrose | wcselby: You absolutely must use either answer() or progress() before queue. |
| 16:19.01 | skrusty | ta-da ;) |
| 16:19.04 | jrrose | Well, that's not entirely true. |
| 16:19.10 | jrrose | But if you don't, expect horrible things to happen. |
| 16:19.11 | skrusty | thanks wcselby - stupid mistake! |
| 16:19.34 | jrrose | Oh , was that for you skrusty? |
| 16:19.40 | jrrose | In any event, same answer. |
| 16:19.49 | skrusty | thanks jrrose ;) |
| 16:20.12 | joren | Anyone this morning want to give me a couple dozen lines of config? :P |
| 16:20.18 | skrusty | it was late last night when i installed 1.8, so my bad! schoolboy error :) |
| 16:20.44 | *** part/#asterisk Carlos_PHX_ (~Carlos@ip24-56-6-80.ph.ph.cox.net) |
| 16:21.11 | jrrose | skrusty: For more on that, check out this wiki page: https://wiki.asterisk.org/wiki/display/AST/Early+Media+and+the+Progress+Application |
| 16:21.40 | joren | I need a server, with anonymous clients, one not anon, that picks up a signal, all the others I'd like muted by the server |
| 16:21.49 | voipeng | anyone recommend a asterisk consulting company that will do support on a T/M case by case basis instead of a subscription? |
| 16:22.06 | joren | this is for occupy. for whatever that's worth |
| 16:22.32 | jrrose | Long story short, some applications auto answer because it's just practical for them to do so. Some of those applications have the ability to disable auto answer (like playback) in case that isn't desired. Some applications, like Queue, have a lot of ways they can be initiated, so they don't autoanswer at all. |
| 16:22.41 | ChannelZ | What, the flea baggers are going from 'human microphone' to voip? |
| 16:22.49 | joren | I am. |
| 16:23.03 | joren | I'm not a flea bagger. I have hardware |
| 16:23.11 | joren | maybe I'm a fuckin flea bagger :P |
| 16:23.23 | jrrose | Another thing you could do with queue is X,1,Ringing() and then X,2,Queue(blah) in order to make it never answer and just keep ringing until an agent picks up. |
| 16:23.29 | joren | human mic is tiresum |
| 16:23.58 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 16:24.52 | joren | we have a DJ truck w/1000w inverter, sound, dj that the broadcast will go from |
| 16:25.02 | joren | I should just do it |
| 16:25.24 | joren | I have some of the know how |
| 16:25.24 | joren | I'll ask here if I need anything |
| 16:25.47 | joren | I need my damn injector is what I need |
| 16:26.02 | *** join/#asterisk devmikey (~irc@unaffiliated/devmikey) |
| 16:30.07 | wcselby | my bluetooth just died, i guess I gotta reboot.......? |
| 16:38.02 | *** join/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell) |
| 16:38.03 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 16:40.06 | chuckf | so the lazy are looking to get lazier? |
| 16:40.19 | skrusty | jrrose: thanks for the tips! |
| 16:41.51 | *** join/#asterisk r0m|u (~wtf@darkstar.rice.edu) |
| 16:43.08 | jrrose | skrusty: No problem. |
| 16:43.13 | leifmadsen | voipeng: I do |
| 16:43.30 | voipeng | ok |
| 16:43.35 | p3nguin | What's a T/M? |
| 16:43.39 | voipeng | time and material |
| 16:44.09 | leifmadsen | voipeng: http://www.leifmadsen.com |
| 16:45.21 | voipeng | thank you |
| 16:48.03 | Kobaz | i always thought it was toast and mayo |
| 16:50.56 | *** join/#asterisk __OpenGL__ (~alan@87.194.173.43) |
| 16:51.23 | *** join/#asterisk nighty- (~nighty@static-68-179-124-161.ptr.terago.net) |
| 16:51.32 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v016-024.mobile.uci.edu) |
| 16:52.30 | __OpenGL__ | could someone point me in the correct direction for locating a document that might explain the correct format for sip.conf entries in asterisk 1.8 where two asterisk servers should be authenticated to each other using a pre shared pass phrase? |
| 16:52.56 | __OpenGL__ | im a little confused as to if I need secret, remotesecret, both, what happens if one isnt set etc |
| 16:54.27 | __OpenGL__ | ive covered my arse by adding permit and deny rules to limit by ip; however when I changed the secret= line on one side to update the pre shared pass phrase for a stronger one both sides still exchanged calls as if nothing had changed... |
| 16:54.41 | [TK]D-Fender | __OpenGL__, "secret" and if one is set and the other isn't (side that is), then they don't aggree and the call won't match |
| 16:55.16 | __OpenGL__ | all my sip.conf entries for devices (ie hardware phones) use secret and none of those devices can talk with asterisk or mmake calls unless secret is set correctly on the device |
| 16:55.24 | [TK]D-Fender | __OpenGL__, If you have a fixed host, then you don't need permit/deny either |
| 16:55.50 | p3nguin | Although an ACL doesn't hurt. |
| 16:55.59 | [TK]D-Fender | __OpenGL__, "insecure" can also bypass secrets |
| 16:56.14 | __OpenGL__ | ok so one side I have secret set, host set |
| 16:56.16 | [TK]D-Fender | p3nguin, It can if your mask hits the actual host |
| 16:56.20 | *** join/#asterisk SuperNull (~FreeManof@24-148-101-238.ip.mhcable.com) |
| 16:56.22 | __OpenGL__ | on the other side I have both set too |
| 16:57.17 | __OpenGL__ | hmm I shouldnt be using insecure= line for anything but services that require it i dont think? |
| 16:57.28 | [TK]D-Fender | __OpenGL__, Correct |
| 16:57.40 | __OpenGL__ | an asterisk box talking to an asterisk box never seems to need that to work? |
| 16:57.53 | p3nguin | only if it does. |
| 16:58.05 | [TK]D-Fender | __OpenGL__, It works the way you choose for it to work. |
| 16:58.13 | p3nguin | s/only/It only needs it/ |
| 16:58.28 | [TK]D-Fender | __OpenGL__, You have a secret on one side and insecure on the other? Will slide right through regardles. |
| 16:58.39 | __OpenGL__ | it seems I need to use insecure= lines where the other side isnt using asterisk most of the time... |
| 16:59.22 | __OpenGL__ | hmm ugh |
| 16:59.44 | __OpenGL__ | ive got secret set on both sides. im not using insecure= lines on any asterisk-asterisk entry |
| 17:00.30 | [TK]D-Fender | __OpenGL__, Ok well if You'd like a hand, pastebin is your friend... |
| 17:00.30 | [TK]D-Fender | ~pb |
| 17:00.31 | 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.us , 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. |
| 17:00.32 | [TK]D-Fender | ^^^ |
| 17:01.15 | __OpenGL__ | the only insecure= line ive got set is for localphone; they also have a host= line - that means that noone can use that from outside my asterisk box to that sip entry I should think? |
| 17:02.12 | __OpenGL__ | what you said re insecure= meaning that secret is ignored made me concerned that perhaps someone from outside could use my localphone sip entry in ways I didnt expect |
| 17:02.48 | __OpenGL__ | but as I say my localphone entry has host= set so that should be OK? |
| 17:02.52 | *** join/#asterisk chazzam (~chazz@50-81-150-34.client.mchsi.com) |
| 17:04.01 | *** join/#asterisk kamikazemicrow-1 (~kamikazem@seraph.contegix.com) |
| 17:04.57 | [TK]D-Fender | __OpenGL__, Yes, the host will restrict if you set it |
| 17:05.17 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 17:06.55 | __OpenGL__ | http://pastebin.com/zQpDiKSG this for example does not work |
| 17:07.18 | __OpenGL__ | remotesecret= lines were added just half an hour ago |
| 17:07.50 | __OpenGL__ | i get failed to authenticate INVITE on the sending server with that setup |
| 17:10.36 | __OpenGL__ | ive just tried without the remotesecret lines as well which also does not work |
| 17:10.53 | *** join/#asterisk fisted_ (~fisted@unaffiliated/fisted) |
| 17:11.05 | [TK]D-Fender | __OpenGL__, http://pastebin.com/xCpASQsf |
| 17:12.55 | *** join/#asterisk nighty- (~nighty@static-68-179-124-161.ptr.terago.net) |
| 17:13.04 | __OpenGL__ | what are trustrpid=yes |
| 17:13.04 | __OpenGL__ | sendrpid=yes ? |
| 17:14.26 | [TK]D-Fender | lallows you to pass on CID from devices on one side to the other |
| 17:14.46 | [TK]D-Fender | so you can see that <100> Joe is the one calling over to the other side |
| 17:14.55 | __OpenGL__ | ah year |
| 17:14.58 | __OpenGL__ | yeah* |
| 17:15.09 | [TK]D-Fender | So that it isn't just "from box B" |
| 17:15.10 | __OpenGL__ | isnt that the default? |
| 17:15.13 | [TK]D-Fender | nope |
| 17:15.24 | __OpenGL__ | on 1.4 was it the default? |
| 17:15.27 | [TK]D-Fender | CID will interfere and could be the reason for failures to match |
| 17:18.47 | *** join/#asterisk jaux (~jaux@bas3-toronto47-1242433431.dsl.bell.ca) |
| 17:20.48 | *** join/#asterisk wcselby (~wcselby@pdpc/supporter/professional/wcselby) |
| 17:20.49 | wcselby | o/ |
| 17:23.17 | *** join/#asterisk paulc (~paulc@unaffiliated/paulc) |
| 17:24.17 | *** join/#asterisk timahvo1 (~rogue@197.176.168.192) |
| 17:24.28 | __OpenGL__ | ok ive adapted your example but I get a username mis match |
| 17:25.50 | __OpenGL__ | on the remote side its telling me that it failed to authenticate device <name of my hardware phone> |
| 17:26.19 | [TK]D-Fender | PB <- |
| 17:26.38 | jaux | Hi guys, is there a way I can find a voicemail is resulted from which unsuccessful call? For example, a call with unique id 123456 was no answer, so the caller left a voicemail says it was msg0001.wav/txt, in this voicemail info file, is it possible to include that call's unique id? |
| 17:26.40 | __OpenGL__ | ok |
| 17:26.52 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 17:27.33 | __OpenGL__ | http://pastebin.com/qZScw1kh |
| 17:28.04 | [TK]D-Fender | __OpenGL__, get rid of the permit/deny |
| 17:28.35 | [TK]D-Fender | _-and fix the NAT back. * should always take care of itself |
| 17:28.53 | [TK]D-Fender | __OpenGL__, and include the failed attempt. |
| 17:29.00 | __OpenGL__ | one box is behind a nat firewall, one isnt |
| 17:29.31 | [TK]D-Fender | Doesn't amtter. Each box should be taking care of it's own business. You should never need that for * or any provider |
| 17:30.29 | p3nguin | its |
| 17:31.06 | __OpenGL__ | the ports etc are open on each nat and forwarded to each of the boxes, but surely I need the nat= lines as otherwise traffic always seems to end up going to the wrong IP at the other end |
| 17:31.10 | __OpenGL__ | and other spurious issues |
| 17:33.43 | [TK]D-Fender | __OpenGL__, Only if you configured 1 end wrong |
| 17:34.32 | __OpenGL__ | so what are you saying? I dont need nat= lines or? |
| 17:34.43 | [TK]D-Fender | Correct. |
| 17:34.48 | [TK]D-Fender | should be "no" |
| 17:35.05 | [TK]D-Fender | let each sides NAT settings under general take care of thier own business |
| 17:35.21 | *** join/#asterisk umay (~chris@67-6-169-86.hlrn.qwest.net) |
| 17:35.24 | __OpenGL__ | this end has a nat... which has ports forwarded correctly to the internal asterisk box |
| 17:35.49 | __OpenGL__ | the other side the asterisk box has the internet IP, ports opened on its iptables firewall and provides NAT for internal clients |
| 17:36.24 | [TK]D-Fender | Need more that port forwarding : |
| 17:36.26 | [TK]D-Fender | ~sipnat |
| 17:36.26 | infobot | [~sipnat] Quick guide on configuring Asterisk + SIP behind NAT: http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions . Typically, you MUST configure these settings with appropriate values in the [general] section of sip.conf: nat, canreinvite, externhost or externip, and localnet. |
| 17:36.30 | [TK]D-Fender | than* |
| 17:37.08 | __OpenGL__ | ive got directmedia= externip and localnet set |
| 17:37.15 | __OpenGL__ | gimme moment to read |
| 17:37.40 | [TK]D-Fender | nat=yes for [general] |
| 17:37.46 | *** join/#asterisk Pan3D (~Pan3D@63.208.160.190) |
| 17:37.47 | __OpenGL__ | yes got that |
| 17:37.51 | __OpenGL__ | on my side |
| 17:37.51 | *** join/#asterisk _Corey_ (~chatzilla@mail1.infradapt.com) |
| 17:38.23 | __OpenGL__ | remote side doesnt have that set |
| 17:38.26 | wcselby | [TK]D-Fender you don't want to mix your nat settings anymore |
| 17:38.35 | [TK]D-Fender | wcselby, ? |
| 17:38.35 | wcselby | according to the latest security release for * |
| 17:38.46 | wcselby | nat= should be the same in general and per-peer |
| 17:38.57 | [TK]D-Fender | wcselby, that is insane <- |
| 17:39.00 | wcselby | otherwise username enumeration is a possibility |
| 17:39.17 | WIMPy | The short version is not to use nat= in peers at all. |
| 17:39.23 | wcselby | [TK]D-Fender one sec, let me link you to the appropriate mailing list thread |
| 17:39.33 | [TK]D-Fender | wcselby, that breaks a lot of ITSP's whose media and signalling servers are separate |
| 17:39.58 | [TK]D-Fender | Rollback or wait for a fix. Accomodating it is nuts |
| 17:40.03 | __OpenGL__ | both sides are now nat=no in the peer; i get the same asterisk cli output as before now intersprinkled with retransmit failures which keep retrying |
| 17:40.16 | wcselby | [TK]D-Fender -> http://lists.digium.com/pipermail/asterisk-users/2011-December/268426.html |
| 17:40.27 | [TK]D-Fender | __OpenGL__, Setting is wrong, or port forwarding is bad, or implementation is hostile |
| 17:41.19 | __OpenGL__ | port forwarding should be good this side; inbound calls from itsp into my default context work correctly |
| 17:41.27 | __OpenGL__ | UDP 5060 and UDP10000-20000 are forwarded |
| 17:41.53 | __OpenGL__ | other side hasnt got nat, ports are opened in its INPUT chain 5060 and 10000-20000 udp |
| 17:42.04 | __OpenGL__ | how could my setup be 'hostile' ? |
| 17:42.07 | *** part/#asterisk asterisk-Tester (~RAMYT@193.227.170.247) |
| 17:42.44 | [TK]D-Fender | depends on what router is doing your NAT |
| 17:42.52 | __OpenGL__ | iptables masquerade this side |
| 17:43.05 | [TK]D-Fender | Should be good then. |
| 17:43.16 | __OpenGL__ | on the otherside they have nat but the asterisk is running on the public ip |
| 17:43.24 | [TK]D-Fender | __OpenGL__, ok, lets leave NAT out of this for now |
| 17:43.27 | __OpenGL__ | with ports opened in input chain |
| 17:44.09 | [TK]D-Fender | __OpenGL__, change [ycs} for the precise same name as the other side as well |
| 17:45.02 | __OpenGL__ | does it need to be the same as username fromuser etc? |
| 17:45.12 | [TK]D-Fender | make it all match on both sides |
| 17:47.02 | __OpenGL__ | ok I modified the peer on one side, saved and sip reload and the peer was instantly available |
| 17:47.17 | __OpenGL__ | then I went to the other side and made the change and the peer was instantly available |
| 17:47.20 | __OpenGL__ | lets try to make call |
| 17:48.26 | __OpenGL__ | oops needed to edit extensions.conf for the new [name] diaplan reload now to retry |
| 17:49.00 | __OpenGL__ | yes that works including two way audio |
| 17:49.36 | [TK]D-Fender | \o/ |
| 17:49.58 | __OpenGL__ | is it impossible to have the [section] on either side having different names then? |
| 17:50.15 | __OpenGL__ | whats an appropriate name on one side might not be appropriate for the other side thats all... |
| 17:50.58 | [TK]D-Fender | __OpenGL__, well multiple things changed. the username should ahve done it all by itself... and might have depending what other changes arrived at the same time. |
| 17:51.39 | __OpenGL__ | ok wanna bounce some other things off you if thats OK |
| 17:52.04 | __OpenGL__ | way back about 3 years ago I got bitten and stung badly due to a typo in my sip.conf |
| 17:52.05 | [TK]D-Fender | <- highly elastic |
| 17:52.49 | __OpenGL__ | I typoed the secret= line and made that line invalid. there were no permit or deny rules and my [sections] werent named in hard to guess ways |
| 17:53.03 | p3nguin | Here's an example using a different name on each side: http://pastebin.com/Ag7tknm2 |
| 17:53.33 | __OpenGL__ | someone came along registered my extension and made premium rate phone calls untill I hit my credit limit at my provider - ouchies |
| 17:53.55 | __OpenGL__ | if remotesecret or md5secret arnt set would that open me up to the same kind of exploit? |
| 17:54.32 | __OpenGL__ | thanks p3nguin |
| 17:55.48 | [TK]D-Fender | __OpenGL__, I have never heard of any use of "remotesecret" before |
| 17:55.50 | __OpenGL__ | I know nowadays technically noone could come and register an extension like that because I have a multi tiered security approach but its been a concern of mine in the badck of my head |
| 17:56.25 | __OpenGL__ | if secret is set a pass phrase is required provided that insecure= is not set? |
| 17:56.52 | *** join/#asterisk MarKsaitis (~MarKsaiti@88.96.60.78) |
| 17:56.53 | [TK]D-Fender | You could have a secret on one side and insecure on the other. |
| 17:57.10 | [TK]D-Fender | Particularly when you want them to auth to register but not when sending calls |
| 17:58.14 | __OpenGL__ | not something id want in any of my setups tbh |
| 18:00.07 | __OpenGL__ | http://pastebin.com/d7iAGZCR I have the odd extension configured like this dotted around, assuming noone sniffed and grabbed that pass phrase some how there should be no other way to auth that extension and make calls? |
| 18:00.37 | __OpenGL__ | in these cases I cant use permit and deny rules and firewalling to add an additional layer of security |
| 18:00.50 | __OpenGL__ | because the users are road warriors |
| 18:01.40 | __OpenGL__ | we *do* use vpn for these people but sometimes they go to places that dont pass the vpn right and I have to open something like that paste up... |
| 18:05.09 | *** join/#asterisk Lunarion (~Lunarion@LOrleans-167-37-42-184.w193-250.abo.wanadoo.fr) |
| 18:05.38 | __OpenGL__ | http://asteriskfaqs.org/tag/remotesecret is my concern btw |
| 18:06.09 | __OpenGL__ | im concerned that if remotesecret isnt set does it act like if secret isnt set and open a passwordless hole? |
| 18:06.44 | *** join/#asterisk kikohnl (~kotis@ext-dip-171.hnl.cdsinc.com) |
| 18:08.00 | IsUp | turn off your server |
| 18:08.44 | [TK]D-Fender | __OpenGL__, Set up aternative means. Complex secrets, fail2ban, etc |
| 18:09.26 | [TK]D-Fender | __OpenGL__, port-knocking, web-driven firewall access, etc |
| 18:09.54 | [TK]D-Fender | __OpenGL__, remember security is a process, not a product |
| 18:10.27 | ChannelZ | shakes his fists at Norton Internet Security |
| 18:10.30 | __OpenGL__ | make sense |
| 18:10.34 | __OpenGL__ | makes* sense |
| 18:12.44 | __OpenGL__ | since being stung before I have everything locked down in a multi tiered fashion |
| 18:14.18 | *** join/#asterisk singler (~singler@84.15.129.49) |
| 18:14.30 | __OpenGL__ | ipt firewall; permit deny rules; complex passphrases; non standard [sip entry names]; lines in extensions.conf which match and block premium rate prefixes in the outbound config; all destinations over 30p/minute blacklisted at the itsp |
| 18:14.56 | __OpenGL__ | dont think there is much else I can do except f2b which I keep meaning to implement but need time to get my head around it |
| 18:15.46 | __OpenGL__ | oh and a more sensible credit limit with the itsp so if somehow all the above fails me I dont end up with a 4 figure bill |
| 18:16.10 | WIMPy | Is your RTP secure? |
| 18:16.23 | __OpenGL__ | no :( |
| 18:16.49 | __OpenGL__ | I trust my isp tho and where phones are at remote sites permanently they have permit deny rules and go via vpn |
| 18:17.11 | __OpenGL__ | thats permit deny to a localip - 172.x |
| 18:17.38 | WIMPy | But that's only for SIP, not for RTP, or is it? |
| 18:18.01 | __OpenGL__ | rtp = media? ie the audio data... ? |
| 18:18.12 | WIMPy | yes |
| 18:18.38 | __OpenGL__ | for sites that are remote and where the phones are a permanent fixture the signalling and the rtp goes via the vpn |
| 18:19.12 | __OpenGL__ | for roadwarriors where the other end passes vpn correctly they vpn in and voip goes over the vpn |
| 18:19.41 | __OpenGL__ | very occasionally I have to open a passworded extension to the internet but only last resort |
| 18:20.55 | __OpenGL__ | inbound calls come direct from a bit telco... rtp and signalling for those isnt secured but I trust my ISP |
| 18:22.53 | WIMPy | Securing rtp ports is jst as important as securing the sip port. |
| 18:23.18 | __OpenGL__ | what avenues of attack are we talking of here? asterisk vulns? |
| 18:24.02 | WIMPy | Always set strictrtp=yes |
| 18:24.43 | WIMPy | In theory that should be enough. |
| 18:24.58 | __OpenGL__ | thats the default and ive never found the need to change it |
| 18:25.27 | WIMPy | But the firewall might be of use there, too. |
| 18:25.32 | WIMPy | It used to default to no. |
| 18:25.50 | WIMPy | Godd that that has been fixed. |
| 18:26.02 | __OpenGL__ | unfortunately due to the way bit itsp delivers the calls I cant firewall the inbound rtp or sip signalling as I would like |
| 18:26.21 | __OpenGL__ | big* itsp |
| 18:27.40 | __OpenGL__ | however I have been careful in my extensions config not to take variables that are externally influenced and use that data without sanity checks |
| 18:27.56 | p3nguin | What is the name of this "big ITSP" that you're talking about? |
| 18:28.01 | __OpenGL__ | magrathea |
| 18:28.44 | *** join/#asterisk _Corey_ (~chatzilla@mail1.infradapt.com) |
| 18:30.32 | __OpenGL__ | the only undesirable behaviour that I can think of that may happen to my setup is if someone guesses /sniffs the extension they deliver phone calls to they could create nuisance calls on my internal extensions or to my (hard coded) mobile number |
| 18:31.07 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 18:37.20 | __OpenGL__ | the only variable I use or manipulate in externally facing ael code that is externally provided is callerid name and number; I couldnt think of any way this could be manipulated maliciously but I do some sanity checking anyway |
| 18:40.31 | [TK]D-Fender | __OpenGL__, Sounsd like you've long since learned your lesson and already have ides for when you an get around to them. |
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| 18:47.09 | __OpenGL__ | yeah 4 figure phone bill made me probably overcautious |
| 18:51.48 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 19:05.23 | Kobaz | mm, lots of little birdds out pecking in the grass |
| 19:05.55 | Kobaz | time for lunch |
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| 19:22.40 | MrTelephone | Windows event logging is poopy |
| 19:23.14 | tzafrir | If I have a newer asterisk with CAPABILITIES2 and FORMAT2 support, will it fail to connect to an older asterisk without those? |
| 19:23.25 | __OpenGL__ | adds another recorded message CID to his blocklist |
| 19:23.27 | *** join/#asterisk navaismo (~navaismo@201.123.84.9) |
| 19:24.35 | __OpenGL__ | but question to throw out to the more knowledable masses here... could I somehow read a text file from compare the callerid against each line in that; then if mmatch go to?something |
| 19:25.10 | navaismo | hi everyone, one quick question: If i want to remove the sound auth-thankyou in the app voicemail when we press the # i need to remove this ast_play_and_wait(chan, "auth-thankyou"); from the app.voicemail.c t |
| 19:25.10 | MrTelephone | you should use func_odbc |
| 19:25.16 | navaismo | and compile again? |
| 19:25.16 | __OpenGL__ | as opposed to atm where I have a gotoif callerid num = 901233456 ? something - im getting quite a collection of these lines |
| 19:25.20 | MrTelephone | opengl, you have alocal mysql? |
| 19:25.26 | __OpenGL__ | yes |
| 19:25.43 | MrTelephone | I just setup func_odbc a couple weeks ago and it works good |
| 19:25.47 | navaismo | and most important i can do that? |
| 19:25.55 | p3nguin | __opengl__: Use the database. |
| 19:26.04 | p3nguin | astDB or any other DB of your liking. |
| 19:26.22 | __OpenGL__ | thanks for the suggestions, which is the simplest to use for storing arbitary values? |
| 19:26.33 | MrTelephone | do you have to pull a value and use it? |
| 19:26.35 | p3nguin | I use astDB for everything, it seems. |
| 19:26.49 | MrTelephone | isn't astdb just field,value pairs? |
| 19:27.00 | p3nguin | Yes. |
| 19:27.08 | __OpenGL__ | erm i need to match callerid num against a list and if match go to something that doesnt ring my phones |
| 19:27.45 | p3nguin | database put noring 901233456/numbers 1 |
| 19:27.55 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 19:27.59 | MrTelephone | make a table with columns called id,callerid and use the func_odbc statement SELECT COUNT(*) from where callerid=$ARG1.. In the dialplan if it returns 1 count then goto something |
| 19:28.07 | __OpenGL__ | if I gain a good enough understanding of how to do it all perhaps I could create a special extension which we can transfer calls to if they are recorded message calls that will take their callerid num and add it to the list?! |
| 19:28.12 | MrTelephone | if 0 or the database is down it will continue as normal |
| 19:28.27 | MrTelephone | I use that same thing for my blacklist script |
| 19:28.35 | MrTelephone | blocking incoming calls like your doing |
| 19:28.56 | __OpenGL__ | sweet |
| 19:29.10 | __OpenGL__ | ill have a play after ive filled my gut with grub :D |
| 19:29.11 | MrTelephone | readsql=SELECT COUNT(*) FROM blacklist WHERE callerid='${ARG1}' and accountcode='${ARG2}' |
| 19:29.15 | p3nguin | GotoIf($[${DB_EXISTS(noring/${CALLERID(num)}/numbers)}]?something) |
| 19:29.33 | MrTelephone | exten => s,1,Set(BL=${ODBC_BLACKLIST(${CALLERID(num)},807${CALLERID(dnid)})}) |
| 19:29.34 | MrTelephone | exten => s,n,GotoIf($["${BL}" > "0"]?busy:continue) |
| 19:29.54 | __OpenGL__ | MrTelephone, is that using mysql? |
| 19:30.04 | __OpenGL__ | no |
| 19:30.07 | MrTelephone | odbc |
| 19:30.13 | __OpenGL__ | alan read it before opening mouth :P |
| 19:30.21 | p3nguin | obdc is just a connector. |
| 19:31.13 | MrTelephone | astdb would work too but with mysql it is easy to make a web frontend to add numbers or something |
| 19:31.22 | MrTelephone | whatever your familiar with |
| 19:32.00 | __OpenGL__ | one last thing to throw into the conversation before I go feed my face... llast time my isp had a failcake my asterisk 1.4 install fell to bits and stopped working even for devices that should need no internet... I was thinking this was caused by the single threaded nature of * blocking retrying the many external peers ? |
| 19:32.33 | leifmadsen | __OpenGL__: open issue |
| 19:32.34 | MrTelephone | upgrade |
| 19:32.48 | MrTelephone | leifmadsen whats going on buddy |
| 19:32.54 | __OpenGL__ | note ive since moved to 1.8 due to needing g722 and as part of a general upgrade and ive yet to try blocking internet connectivity to the pbx... |
| 19:34.15 | leifmadsen | __OpenGL__: https://issues.asterisk.org/jira/browse/ASTERISK-18930 |
| 19:34.30 | leifmadsen | MrTelephone: just trying to stay focused on work :) |
| 19:34.36 | leifmadsen | oh look a squirrel! |
| 19:36.49 | MrTelephone | squirrels are in hibernation |
| 19:36.56 | MrTelephone | battlefield 3 tonight? |
| 19:37.02 | leifmadsen | I wish :) |
| 19:37.05 | leifmadsen | I don't have BF3 |
| 19:37.12 | MrTelephone | it will ruin your life |
| 19:37.18 | leifmadsen | hence.... not having it ;) |
| 19:37.47 | MrTelephone | better than doing crack I suppose |
| 19:38.38 | leifmadsen | I suppose... |
| 19:39.21 | MrTelephone | How come the hard drive companies are not starting up some quick factories in america. Is it really that expensive? |
| 19:39.27 | *** join/#asterisk file (~file@asterisk/developer-and-muffin-lover/file) |
| 19:39.27 | *** mode/#asterisk [+o file] by ChanServ |
| 19:39.46 | MrTelephone | I'm starting to get a little worried here |
| 19:40.09 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
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| 19:51.11 | _Corey_ | <PROTECTED> |
| 19:56.43 | __OpenGL__ | hdd situation is making me sad :( its inflating the costs of my core products and the fact im struggling to provide quotes valid for more than 5 days is putting people off |
| 19:56.50 | __OpenGL__ | prices seem to be trending down tho... |
| 20:02.19 | wcselby | ... |
| 20:05.30 | __OpenGL__ | leifmadsen, i hope they find a fix for this issue soon. I dont recall this hapenning on 1.6 |
| 20:05.32 | __OpenGL__ | sorry |
| 20:05.33 | __OpenGL__ | 1.2 |
| 20:06.45 | __OpenGL__ | in my experience with 1.4.x my internet had to be down for a number of minutes before asterisk appeared to become blocked; 2-3 minutes seemed OK |
| 20:06.56 | __OpenGL__ | but my testing isnt exhaustive |
| 20:10.00 | leifmadsen | __OpenGL__: yes sounds like the issue that is opened against 1.8 -- not sure when it'll get resolved. If you have the resources or skills to triage the issue and provide a patch, it will certainly help move the issue along faster. Other than that, you'll just need to be patient or work around the issue. |
| 20:11.23 | *** join/#asterisk bn-7bc (bjarne@pdpc/supporter/active/bn-7bc) |
| 20:15.40 | __OpenGL__ | leifmadsen, unfortunately my skillset runs out at very basic php mysql coding... my skills are mainly sys admin related |
| 20:16.03 | wcselby | __OpenGL__ what is the issue? |
| 20:16.03 | leifmadsen | __OpenGL__: then the latter part of my msg applies :) |
| 20:16.43 | wcselby | your internet crapped out and your phones stopped working? |
| 20:16.48 | wcselby | even the local ones? |
| 20:16.57 | __OpenGL__ | wcselby, its the 'if asterisk looses internet connectivity and has internet peers defined via dns names asterisk gets blocked trying to do something and services no new requests - bug |
| 20:17.02 | __OpenGL__ | yes |
| 20:17.12 | wcselby | install a local caching dns server |
| 20:17.15 | wcselby | on the asterisk box |
| 20:17.29 | wcselby | it's a work around, but the only one that I know of at the moment |
| 20:17.53 | __OpenGL__ | I have a linksys spa3102 adaptor at my house (& home office) specifically for the time when the internet broke... so I could say redirect my voip numbers to another box and forward them to the landline and make calls in/out using land line in emergency |
| 20:18.21 | __OpenGL__ | however when I came to need it as my internet was down it turned out asterisk totally failed and I wound up finding a POTS phone and plugging it in to get my calls :P |
| 20:18.30 | *** join/#asterisk timahvo1 (~rogue@197.176.91.101) |
| 20:18.36 | leifmadsen | wcselby: seems like a reasonable work around |
| 20:18.43 | __OpenGL__ | my internet went down cus a fibre got cut somewhere in betheres network and there was no redundant path :@ |
| 20:19.17 | __OpenGL__ | I have a local caching dns server on my network but when the caches expire asterisk fails |
| 20:19.21 | r0m|u | waz up wcselby |
| 20:20.25 | __OpenGL__ | that might be why my asterisk box seems to weather a internet lost sync and it took a while to come back up event without issue; however a internet was down for over an hour event causes everything to fail miserably |
| 20:21.40 | __OpenGL__ | wcselby, luckilly its somewhat a non issue for me because my internet is reliable... im more concerned for 'disaster planning' scenarios at my customers places |
| 20:21.55 | wcselby | heh |
| 20:21.56 | wcselby | yeah |
| 20:22.20 | wcselby | i've worked around the issue by installing the caching dns server ont he asterisk box itself, I seem to recall having some issue using a different server on the network |
| 20:22.37 | wcselby | and then I think you set the cache to something really long |
| 20:22.39 | wcselby | like a week |
| 20:22.45 | wcselby | sup r0m|u |
| 20:23.07 | __OpenGL__ | id rather not have to say to my clients: if the internet breaks expect to have no phone system at all; id rather be able to say itll degrade and youll have your backup pots line and internall calling only |
| 20:23.08 | __OpenGL__ | or something |
| 20:23.18 | wcselby | leifmadsen i can't take credit for it, I don't recall where I saw it first, but it's worked for me |
| 20:23.27 | __OpenGL__ | wcselby, could that backup dns run on a non standard port or something? |
| 20:23.28 | leifmadsen | well it certainly makes sense ;) |
| 20:23.37 | wcselby | :) |
| 20:23.57 | wcselby | __OpenGL__ that's exactly what installing the local caching name server on the asterisk box allows you to do |
| 20:24.07 | __OpenGL__ | all customers with asterisk also have their own caching dns server... but what if the provider in question sets a low stale ttl thingy? |
| 20:24.36 | *** join/#asterisk Neptu (~jromero@213-67-244-97-no100.tbcn.telia.com) |
| 20:25.16 | __OpenGL__ | wcselby, what I mean is if it required me to set it up in a way that wouldnt be sensible as a general resolver... ie overriding the dns origins suggested ttls... that might cause issues for other stuff? |
| 20:26.00 | __OpenGL__ | disclaimer: the asterisk boxes at my customers sites are all serving small amounts of users and acting as an internet gateway and other stuff in addition to being asterisk boxes |
| 20:26.10 | wcselby | only have the asterisk box use the cache, and you don't have to worry about anything else getting messed up |
| 20:26.22 | wcselby | oh |
| 20:26.33 | wcselby | you're using the asterisk box itself as the dns server for all the users? |
| 20:26.40 | __OpenGL__ | nods |
| 20:27.22 | __OpenGL__ | in almost every case the server provices: NAT, dhcp, dns, asterisk, samba to small amounts of users - max is about 25 |
| 20:27.31 | wcselby | there may be a way to run both a caching dns server for just the asterisk stuff and general bind for everything else, but I've never had to look at that, so I'm not sure. maybe with a VM? |
| 20:27.49 | __OpenGL__ | yeah |
| 20:27.57 | wcselby | setup a tiny VM just for BIND |
| 20:28.19 | wcselby | update the DHCP entry to point to the new DNS server |
| 20:28.28 | wcselby | and the client's shouldn't have to do anything other than reboot |
| 20:29.02 | __OpenGL__ | I was thinking of another route... have the bind on the box for everyone to use; tell bind only to bind (lol) to the local ip address which is what I do anyway... then create say a dnsmasq caching dns server on a virtual ip... ie create eth0:1 and get just the asterisk box to look at that ip |
| 20:29.41 | wcselby | __OpenGL__ yeah that may work too, that was my first suggestion that I said I'd never tried to do so I didn't know how or if it would work :) |
| 20:30.13 | __OpenGL__ | useful to bounce ideas around :) |
| 20:30.25 | wcselby | indeed |
| 20:30.59 | __OpenGL__ | weather here is vile :( |
| 20:31.59 | __OpenGL__ | shoot its forcast up to 10cm of snow by morning |
| 20:32.01 | __OpenGL__ | eurgh!!! |
| 20:32.07 | wcselby | snow? |
| 20:32.18 | wcselby | is that that white stuff that falls on the ground for 30 minutes once every 5 years? |
| 20:32.22 | __OpenGL__ | http://www.metoffice.gov.uk :P |
| 20:32.53 | __OpenGL__ | snow caused some interesting issues over the past 3 years ive been working for myself |
| 20:33.22 | __OpenGL__ | like everyone at an organisation thinking 'oh ill work from home ive got vpn access' then the companies 1mbit upload falling on its face |
| 20:33.32 | wcselby | hahahahaha |
| 20:33.41 | wcselby | yeah I have a client that still uses a T1 as their internet service |
| 20:33.47 | wcselby | it's miserable trying to download anything there |
| 20:33.49 | wcselby | or upload |
| 20:34.08 | __OpenGL__ | then them moaning my solution wasnt fit for purpose untill I pointed out that id suggested changing it |
| 20:34.24 | wcselby | it's always our fault |
| 20:34.25 | wcselby | :) |
| 20:35.34 | __OpenGL__ | the problem for this client was they wanted vpn but their internet connection is on the end of a very long telephone line so the speed sucks |
| 20:35.49 | __OpenGL__ | anything but adsl was too expensive |
| 20:37.19 | __OpenGL__ | the best one is the construction company MD that got a network with road warrior vpn from me... he said he wanted to open and edit documents, use the internet securely and 'upload some photos' |
| 20:37.27 | __OpenGL__ | so I say yeah itll be find for all those purposes |
| 20:38.22 | __OpenGL__ | then I find he wants to 'upload some photos' but its taking ages... just a few gigs of photos of a finished construction project over a weak gprs connection ;) |
| 20:38.49 | wcselby | yeah, whenever anyone says anything about photos, get more details |
| 20:38.55 | __OpenGL__ | customers mind set: buy product; expect miracles |
| 20:40.04 | __OpenGL__ | customers like to twist things around too... sold someone incremental backups which generally work perfectly fine... been doing incremental backups over the net for 3 years with just 1 case of needing to manually take a disc from one server to the other |
| 20:40.09 | wcselby | a buddy had one client flip out because they wanted to store 500 gigs of photos on a 250 gig share he had set up. they didn't tell him they were storing raw uncompressed web design psd files, with multiple copies and versions of each design, when he purchased their server and set everything up |
| 20:40.45 | __OpenGL__ | then they wonder why i say I cant sync the changes after someone copied 'some photos' to their network shares... just 16 gigs of holiday movies shot on the digital camera... |
| 20:41.57 | __OpenGL__ | 'fun' times anyhow |
| 20:42.20 | wcselby | relocating to our server room, afk for a bit |
| 20:42.38 | __OpenGL__ | just a shame when you fall out with a customer because they have unrealistic demands |
| 20:47.09 | *** join/#asterisk voipeng (~voipeng45@75-150-128-2-Philadelphia.hfc.comcastbusiness.net) |
| 20:57.41 | *** join/#asterisk jetlag (jetlag@pool-71-168-248-89.cmdnnj.east.verizon.net) |
| 20:58.03 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 21:00.24 | *** join/#asterisk bipolar (~bipolar@offsitesysadmin.com) |
| 21:00.50 | *** join/#asterisk vetal (~vitaliy.c@16-94-134-95.pool.ukrtel.net) |
| 21:03.51 | *** join/#asterisk edgeleb (c84b8cf2@gateway/web/freenode/ip.200.75.140.242) |
| 21:04.23 | vetal | hi, I need your help. After I upgrade Asterisk from 1.6 to 1.8 I've got an issue with Call connect, I receive connect from voip provider, but my asterisk do not receive it, so I hear 1 word and then RTP goes to my server, but one way. I can provide full dump and all need information. I think that is was bug, but in issues said that it is configuration. |
| 21:04.54 | __OpenGL__ | one way audio after 1.6 yo 1.8 upgrade? |
| 21:04.55 | __OpenGL__ | to* |
| 21:05.15 | *** join/#asterisk citywok (~citywok@67-134-194-33.dia.static.qwest.net) |
| 21:05.31 | __OpenGL__ | vetal, ^^^ |
| 21:05.34 | vetal | yes |
| 21:05.42 | __OpenGL__ | vetal, nat? |
| 21:05.47 | vetal | no NAT |
| 21:05.58 | vetal | one moment I axplain... |
| 21:06.04 | *** join/#asterisk brettnem (~brett@76-216-205-249.lightspeed.austtx.sbcglobal.net) |
| 21:06.09 | edgeleb | i have an issue with a digium tdm400p....i setup a server with elastix all is rightly configured...but asterisk cant answer the call from the card fxo |
| 21:06.12 | brettnem | hey all |
| 21:06.19 | edgeleb | hi all |
| 21:06.56 | __OpenGL__ | edgeleb, it just rings and rings when you dial in from outside? |
| 21:06.56 | edgeleb | on the CLI i see asterisk treating the call as if it was answered but the incoming call keeps ringing |
| 21:07.03 | edgeleb | yep |
| 21:07.04 | vetal | this I get in console with verbose:– Executing [74957410001@test:1] Dial("SIP/00002-00000000", "SIP/ez/01#7495 |
| 21:07.05 | vetal | 7410001") in new stack |
| 21:07.05 | vetal | == Using SIP RTP CoS mark 5 |
| 21:07.05 | vetal | – Called SIP/ez/01#74957410001 |
| 21:07.05 | vetal | – SIP/ez-00000001 is ringing |
| 21:07.05 | vetal | – SIP/ez-00000001 is making progress passing it to SIP/00002-00000000 |
| 21:07.05 | vetal | [Nov 30 22:43:50] NOTICE[27741]: chan_sip.c:20192 handle_response_peerpoke: Peer |
| 21:07.06 | vetal | 'ez' is now Lagged. (1ms / 0ms) |
| 21:07.06 | vetal | – No one is available to answer at this time (1:0/0/0) |
| 21:07.07 | vetal | – Auto fallthrough, channel 'SIP/00002-00000000' status is 'NOANSWER' |
| 21:07.31 | __OpenGL__ | vetal, use www.pastebin.com for pastes or youll get barred from the irc by the anti flood bot (possibly) |
| 21:07.54 | vetal | sorry, first use |
| 21:08.08 | vetal | can I give you link to issue? |
| 21:08.16 | __OpenGL__ | vetal, you have qualify=1 in your sip.conf set it to qualify=yes or some number larger than 1000 |
| 21:08.29 | __OpenGL__ | and the issue will go away |
| 21:08.54 | __OpenGL__ | vetal, if you dont want qualify set qualify=no not 0 |
| 21:09.33 | vetal | i get qualify=no, if i put qualify=yes i receive Peer 'ez' is now Lagged. (56929ms / 0ms) |
| 21:10.30 | __OpenGL__ | edgeleb, sorry I dont know a lot about the tdm400p or other similar cards however it sounds like asterisk is presenting the incorrect tones to the line to signify answered |
| 21:10.36 | __OpenGL__ | or possibly a site wiring issue... |
| 21:10.50 | __OpenGL__ | vetal, thats strange... who is ez? |
| 21:11.04 | r0m|u | can I set Verbose(0,${CALLERID(num)} to catch numbers been dialed out? |
| 21:11.17 | vetal | ez is operator with softswitch MERA MVTS PRO |
| 21:11.18 | wcselby | edgeleb pastebin us the CLI of a call coming in |
| 21:11.19 | p3nguin | ~pb |
| 21:11.20 | 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.us , 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. |
| 21:11.28 | edgeleb | ummm wiring...that could be...the tones are correct |
| 21:11.38 | __OpenGL__ | vetal are you running virtualised? |
| 21:11.41 | wcselby | edgeleb pastebin us the CLI of a call coming in |
| 21:11.46 | brettnem | hey all, is there a good way to tell which end of a call terminated a call when one side is sip and the other PRI? I've got reports of dropped calls and I'm trying to figure out who's fault it is |
| 21:12.04 | vetal | no, i ve got this issue on 2 my servers, freebsd and centos |
| 21:12.07 | vetal | problem is same |
| 21:12.24 | edgeleb | ok give me a second |
| 21:12.40 | vetal | calls with such problem is 8 of 10, 2 of 10 is going - good |
| 21:12.42 | wcselby | brettnem there is an option you can add to the Dial() commadn that will force it to continue in the dialplan if the callee hangs up first |
| 21:12.52 | *** join/#asterisk kodomo (~gregor@g225037247.adsl.alicedsl.de) |
| 21:12.58 | *** join/#asterisk jkroon (~jkroon@dsl-241-253-186.telkomadsl.co.za) |
| 21:13.01 | wcselby | I don't remember it off the top of my head, but that's an option to look for |
| 21:13.02 | __OpenGL__ | vetal, your pasted output is odd because the second number in the lagged message should be the time limit for the peer poke response |
| 21:13.10 | __OpenGL__ | yet yours is saying 0ms which is weird... |
| 21:13.14 | r0m|u | p3nguin: can I set Verbose(0,${CALLERID(num)} to catch numbers been dialed out? |
| 21:13.45 | brettnem | wcselby: Yeah, I know that option, this is on a freepbx based system, I'm just trying to debug the asterisk bits |
| 21:13.57 | __OpenGL__ | vetal, here is an example from my vanilla asterisk install Peer 'localphone-ycs' is now Lagged. (6971ms / 5000ms) |
| 21:13.58 | p3nguin | r0m|u: Are you wanting to see who is calling, or who is being called? |
| 21:14.02 | wcselby | r0m|u if you want outbound numbers being dialed you would use ${EXTEN} |
| 21:14.10 | vetal | i've got 0ms, when qualify is "no" |
| 21:14.25 | vetal | when yes, i've got 124363467ms / 2000ms |
| 21:14.30 | r0m|u | p3nguin: "who is been called" |
| 21:14.33 | edgeleb | the inconming route is to a IVR all the time the line keep ringing... http://pastebin.com/PMMe8YRA |
| 21:14.45 | p3nguin | You could do both, even. Verbose(${CALLERID(num)} is calling ${EXTEN}) |
| 21:14.57 | r0m|u | wcselby: ooo ok so just change calleris for exten |
| 21:14.59 | __OpenGL__ | vetal, it sounds like a problem with a device at the other end of the connection; its being very slow to respond to pokes from asterisk |
| 21:15.04 | r0m|u | ah! |
| 21:15.25 | helen_ | WIMPy? |
| 21:15.46 | r0m|u | p3nguin: Thats perfect |
| 21:15.54 | __OpenGL__ | vetal, 50000ms is like 50 seconds to respond to a simple poke that should be a high priority to respond to... i cant imagien network latency causing that... |
| 21:16.06 | wcselby | edgeleb asterisk is seeing the call and passing it into an IVR, what you're saying is the caller doesn't hear anything other than ringing? |
| 21:16.25 | edgeleb | exacly just ringing |
| 21:16.33 | kodomo | hi folks... did anyone around here get the snom m9 to work with asterisk 1.8.7.1 (connecting via TLS)? |
| 21:16.40 | edgeleb | and on outgoing calls is the same |
| 21:16.50 | wcselby | edgeleb sounds like you dont' have the config set just right for the card |
| 21:17.04 | vetal | when I setup Asterisk 1.6... All calls go thru - with no problems. |
| 21:17.10 | edgeleb | asterisk shows as it makes the call but the line is never picked up |
| 21:17.15 | wcselby | like maybe you're using hookswitch and the other end is expecting flash or something (i don't recall all the terms off the top of my head) |
| 21:17.30 | wcselby | edgeleb confirm with your telco what the settings should be |
| 21:17.36 | __OpenGL__ | vetal, please prefix your responses with my name, most irc clients will tab complete it for you |
| 21:17.46 | kodomo | I've got the weird situation that everything works out with the registration, but asterisk closes the TLS connection when the snom tries to dial (more specifically: immediately after the ACK for the 401 - not waiting for an authenticated INVITE) |
| 21:17.56 | vetal | __OpenGL__, ok |
| 21:17.59 | __OpenGL__ | pastebin your sip.conf for 1.6 and sip.conf for 1.8 please |
| 21:18.10 | *** join/#asterisk krotos (~d3v1l@host211-3-dynamic.5-87-r.retail.telecomitalia.it) |
| 21:18.23 | edgeleb | wcselby already confirmed with them and the config is ok |
| 21:18.57 | wcselby | edgeleb well show us what you've got for the config |
| 21:19.08 | wcselby | and what they've told you to use |
| 21:19.35 | vetal | __OpenGL__, I make new install for asterisk 1.6 and 1.8, and to samples add only http://pastebin.com/pKsSdpys |
| 21:19.52 | wcselby | kodomo give us a pastebin of the sip debug of the failure, along with another one showing us the CLI output |
| 21:20.20 | edgeleb | wcselby http://pastebin.com/Twsv935u |
| 21:20.25 | vetal | __OpenGL__, what is interesting, that I've got this issue only with MERA sodtswitch and Asterisk 1.8 |
| 21:20.49 | wcselby | edgeleb i meant your dahdi settings or whatever, not the indications |
| 21:21.49 | kodomo | wcselby: sure - thx for the quick response :) : http://pastebin.com/w7byNZhP |
| 21:21.57 | *** join/#asterisk pdtpatrick1 (~ptaylor@12.249.4.226) |
| 21:22.15 | __OpenGL__ | vetal, i hope you've obfuscasted that pass and its not really 12345 |
| 21:22.42 | edgeleb | wcselby http://pastebin.com/gMJmRjAj |
| 21:22.49 | __OpenGL__ | vetal with qualify=no what happens to your calls via the ez? |
| 21:22.56 | *** join/#asterisk eduzimrs (~eduzimrs@201.22.86.124.static.gvt.net.br) |
| 21:23.43 | wcselby | edgeleb and your /etc/dahdi/system.conf file? |
| 21:24.31 | eduzimrs | hi, im having a problem with moh in queues, when i dial to a queue, i do not listen the moh, but its weird because it just happen in sip phone, in xlite for example i hear the moh. |
| 21:24.47 | wcselby | eduzimrs xlite is a sip phone |
| 21:25.02 | __OpenGL__ | eduzimrs, sounds like a transcode issue? |
| 21:25.04 | edgeleb | wcselby http://pastebin.com/gyjs27Mw |
| 21:25.05 | eduzimrs | yeap, i mean physical ones |
| 21:25.14 | eduzimrs | yeap |
| 21:25.18 | *** join/#asterisk FLeiXiuS (~FLeiXiuS@unaffiliated/fleixius) |
| 21:25.21 | vetal | __OpenGL__, when qualify no, i ve got lagged with 1ms/0ms |
| 21:25.55 | FLeiXiuS | Is there a way to initiate a phone call with no audio using something like AGI or AMI (a scriptable interface) |
| 21:26.10 | __OpenGL__ | eduzimrs, could it be a natting issue or something? are the hardphones connected toa different net to your softphone? did you try echotest? |
| 21:26.16 | wcselby | FLeiXiuS Originate() |
| 21:26.26 | kodomo | wcselby: since it's a quite long output: the disconnection happens after line 484 |
| 21:26.32 | __OpenGL__ | vetal, is that the entire contents of your sip.conf file ? |
| 21:26.43 | wcselby | edgeleb what happens if you plug a normal analog phone into the jack from the wall and try to make a phone call or receive a phone call? does it work? |
| 21:26.44 | eduzimrs | __OpenGL__: theire in the same network, echotest no |
| 21:26.50 | FLeiXiuS | wcselby, Thanks! |
| 21:27.01 | kodomo | basically, asterisk seems to state that it doesn't know (anymore about the CSeq used for the invite dialog) |
| 21:27.06 | __OpenGL__ | eduzimrs, try echotest to asterisk to check bidirectional audio is working |
| 21:27.17 | edgeleb | wcselby yes it works without issues when i connect a analog phone to the line |
| 21:27.27 | kodomo | and the actual invite receives only a TCL RST packet, as the socket is already closed |
| 21:27.36 | vetal | __OpenGL__, no I've got full sip.conf what asterisk samples provide |
| 21:27.37 | kodomo | no further error/warning/whatever message |
| 21:27.47 | wcselby | kodomo it's hard for me to read this, can you repaste it without the blocks? |
| 21:27.51 | __OpenGL__ | vetal, strip it to basics |
| 21:28.08 | kodomo | wcselby: I'll try... sec... |
| 21:28.28 | wcselby | just dump it into notepad++ (or similiar) and do a search and replace on whatever that block thing is |
| 21:29.06 | FLeiXiuS | wcselby, Is there a difference between AGI and AMI originate? |
| 21:29.13 | eduzimrs | __OpenGL__: echotest ? |
| 21:29.20 | vetal | __OpenGL__, it is basic, after install I only add to the end of file http://pastebin.com/pKsSdpys |
| 21:29.28 | __OpenGL__ | vetal, http://pastebin.com/wbDaSPkx |
| 21:29.53 | wcselby | edgeleb FLeiXiuS yes, AGI is called from the dialplan while AMI acts from without. |
| 21:30.10 | FLeiXiuS | wcselby, Ahh, that makes much more sense. |
| 21:30.23 | wcselby | so if you don't want to have to dial an extension first, I would suggest using AMI |
| 21:30.29 | __OpenGL__ | vetal, eduzimrs, http://pastebin.com/2kDGPTEk example echo test for extensions.conf |
| 21:31.14 | vetal | __OpenGL__, You mean to setup this configuration in sip.conf? |
| 21:31.25 | __OpenGL__ | vetal, no extensions.conf |
| 21:31.51 | __OpenGL__ | however tho trying to troubleshoot with the incredibly large demo files there is hard... |
| 21:32.25 | wcselby | edgeleb did you write this dialplan yourself or is it generated by a gui like freepbx? |
| 21:32.34 | __OpenGL__ | when i started with asterisk with 1.2 a few years back I had to strip mmy configs to the bare minumums so I could get my head around it and avoid weird problems that became impossible to diagnose |
| 21:32.49 | eduzimrs | __OpenGL__: i heard normal the sound "demo-echotest" |
| 21:33.09 | *** join/#asterisk WiretapSeven (~Wiretap@unaffiliated/wiretap) |
| 21:33.22 | edgeleb | wcselby the dialplan is generated by freepbx on a elastix install |
| 21:33.28 | __OpenGL__ | eduzimrs, you you hear the output of your mic looping back to your earpiece |
| 21:33.38 | vetal | __OpenGL__, I do not understand, you want me to put onle string from pastbin, and try? |
| 21:33.53 | wcselby | so r0m|u I am playing server roulette...trying to see which of these ancient servers will boot so I can migrate that crashed server to a new one. and the one I pull up is a linux box with hostname localhost.localdomain |
| 21:33.57 | wcselby | i'm afraid to find out what's on this box |
| 21:34.05 | FLeiXiuS | wcselby, Yeah I just need to dial an extension for automation...AMI seems like the way to go. |
| 21:34.11 | wcselby | edgeleb hmmm |
| 21:34.29 | wcselby | FLeiXiuS cool, there are lots of examples out there on the web oh originating with different programming languages |
| 21:34.34 | wcselby | php perl python, etc |
| 21:34.40 | __OpenGL__ | vetal, eduzimrs unfortunately my experience lies with vanilla asterisk installs from either source or rpm; not gui based things which ive always found more complex and hard to fault find unfortunately |
| 21:34.53 | wcselby | edgeleb i'm almost tempted to say it's because you have no Answer() anywhere in your dialplan |
| 21:34.55 | FLeiXiuS | wcselby, Yep, I'm going to merge this into a rails app...thanks for the help. |
| 21:35.06 | wcselby | edgeleb but if freepbx is doing the config, it should handle that |
| 21:35.07 | edgeleb | wcselby it is very strange...i'm leaning to the card been faulty but i had to make sure |
| 21:35.07 | carrar | peppermint Asterisk is much better |
| 21:35.14 | eduzimrs | __OpenGL__: humm the dial to the queue continues without audio |
| 21:35.28 | wcselby | edgeleb well I wouldn't say that, I would think if the card was faulty you'd not see the call on the CLI at all |
| 21:35.51 | wcselby | edgeleb are you calling in from a cell phone / land line, have you tried calling in from more than one ? |
| 21:36.14 | eduzimrs | __OpenGL__: it happend after i upgrade * from 1.4 to 1.8 ... |
| 21:36.31 | edgeleb | wcselby i have tried from all kinds of sources land lines cel lines voip severce...all the same |
| 21:36.39 | wcselby | edgeleb gotcha |
| 21:36.55 | __OpenGL__ | eduzimrs, you need to confirm that you have two way audio to the server... this confirms traffic is passing correctly from your devices to the server... i always use echotest for this as per my code snippet which I literally took from my production extensions.conf file |
| 21:36.56 | wcselby | check with your telco and see if they recognize the call as being answered at all? |
| 21:37.28 | *** join/#asterisk MarKsaitis (~MarKsaiti@cpc4-rdng22-2-0-cust932.15-3.cable.virginmedia.com) |
| 21:37.28 | wcselby | edgeleb the only other thing I would suggest, other than trying on the elastix forums or the #freepbx channel, would be to write some customer dialplan in extensions_custom.conf that answers the call first |
| 21:37.36 | __OpenGL__ | eduzimrs, it could be the audio files arnt in place; why upgrade from 1.4 to 1.8 blindly?! a lot of things have changed and your old configs might not work any longer... |
| 21:38.06 | wcselby | i.e in extensions_custom.conf, try adding a new context that is called [from-pstn-custom], and have it say be two lines: |
| 21:38.15 | wcselby | s,1,Answer() |
| 21:38.17 | eduzimrs | __OpenGL__: i read about changes, but this one is the only that not work |
| 21:38.26 | wcselby | s,2,Goto(from-pstn,${EXTEN},1) |
| 21:38.29 | edgeleb | wcselby i'll try that...and i already talked to my telco i they didnt see any pickups |
| 21:38.51 | __OpenGL__ | eduzimrs, dont forget about the changes youd of had to make to go from 1.4 to 1.6... |
| 21:38.55 | wcselby | and then point your dahdi channel to the [from-pstn-custom] contet |
| 21:38.58 | wcselby | context* |
| 21:39.09 | eduzimrs | __OpenGL__: all files are normal and in the right place, it just dont work at physical sip phones |
| 21:39.29 | edgeleb | wcselby ok let me try and i'll get back |
| 21:39.31 | eduzimrs | __OpenGL__: in xlite all goes normal |
| 21:39.40 | wcselby | eduzimrs you're haviung issues with MOH on queue calls only when the call is coming from another sip phone, correct? |
| 21:39.47 | kodomo | wcselby: http://pastebin.com/209JVSJT |
| 21:39.48 | edgeleb | wcselby thanks |
| 21:39.50 | wcselby | from inside the network, not coming in from a provider? |
| 21:39.53 | wcselby | edgeleb good luck |
| 21:40.05 | __OpenGL__ | eduzimrs, sorry the complexities involved with taking a 1.4 install to 1.8 are beyond what i can troubleshoot here. do yourself a favour go back to 1.4 and dont reinstall 1.8 untill you are ready to totally redo your configs |
| 21:40.24 | eduzimrs | __OpenGL__: yes, but physical sip phones...like a said before |
| 21:40.27 | kodomo | wcselby: sorry - took me a moment to figure how to replace Ctrl-whatever in vim ;) |
| 21:40.36 | wcselby | lol no worries |
| 21:40.47 | leifmadsen | wait, someone did an upgrade from 1.4 to 1.8 on a production machine without testing? tsk tsk |
| 21:41.00 | jaytee | harumph! |
| 21:41.34 | kodomo | wcselby: the Match Not Found line is now at 949 |
| 21:41.50 | __OpenGL__ | i dont understand why you would go through the pain of a major version upgrade just for the hell of it |
| 21:42.39 | __OpenGL__ | I recently went from 1.4 to 1.8 and as part of that move totally redid all my config files and still had teething issues. changing versions without redoing your configs is a recipe for disaster |
| 21:43.37 | wcselby | kodomo i'm seeing a lot of "Unable to connect SIP socket to 92.225.46.160:4413: Connection refused" in here |
| 21:43.42 | vetal | __OpenGL__, with your sip.conf - same issue |
| 21:43.44 | eduzimrs | im not in product system yet... |
| 21:43.49 | __OpenGL__ | absolutly tons of config parameters have changed; stuff that worked one way in 1.4 didnt work the same in 1.8, audio files were expected to have different names, extensions.conf syntax and variable names were altered, all sorts of hurdles I had to overcome |
| 21:44.18 | kodomo | wcselby: yes, at the beginning, before the snom booted |
| 21:44.20 | __OpenGL__ | vetal, eduzimrs what purpose is the upgrade to 1.8 serving for you? |
| 21:44.21 | wcselby | __OpenGL__ the issue not much of a misconfiguration with asterisk since when he calls in from xlite, the MOH is heard |
| 21:44.30 | wcselby | kodomo oh okay, I'm still in the first 200 lines lol |
| 21:44.33 | __OpenGL__ | do you need a new feature thats only in 1.8? |
| 21:44.41 | jkroon | hi all, i find some mention of t38 support for IAX/2, however, i'm unable to find documentation indicating that it has been done, or of proposed patches. Does anybody know if this is in the pipeline? Am I missing something somewhere? Would someone be willing to point me at what I need to code it? |
| 21:44.45 | vetal | yes, textsupport |
| 21:45.15 | leifmadsen | no t.38 support on iax2 in the pipeline |
| 21:45.34 | jkroon | leifmadsen, ok, so it's not going to be supported? |
| 21:45.55 | leifmadsen | I didn't say that |
| 21:45.55 | kodomo | wcselby: most is regular stuff... as described above... the snom registers... becomes 'AVAILABLE'... and then the dialing starts... at line... 776 |
| 21:45.58 | vetal | __OpenGL__, one more thing, i forgot about it. When one of good calls go, when i press hangup, I got Lagged |
| 21:46.08 | leifmadsen | jkroon: I said it's not in the pipeline |
| 21:46.32 | wcselby | kodomo thanks :) looking farther down now |
| 21:46.43 | jkroon | leifmadsen, just that no one has yet indicated interest in doing it then. ok, if I'd like to volunteer to make an attempt - where do I start? |
| 21:46.57 | kodomo | wcselby: asterisk prepares a response around 879 (401) |
| 21:47.09 | __OpenGL__ | vetal, ive had similar problems in the past where cisco themselves gave up troubleshooting the issue and took all the phones back |
| 21:47.23 | kodomo | wcselby: the snom responds by ACK at 911 |
| 21:47.41 | leifmadsen | jkroon: I suggest with the code |
| 21:47.52 | kodomo | wcselby: asterisk acknowledges receipt of the ACK at 948 |
| 21:47.52 | __OpenGL__ | vetal, that was wip310 phones which were generally problematic everywhere, so im not even gonna begin to try and figure out why 1.6 worked and 1.8 didnt :( |
| 21:47.59 | kodomo | ...and then ... closes the connection |
| 21:48.02 | leifmadsen | jkroon: probably need to learn whatever is involved with t.38, then determine how best to implement it |
| 21:48.12 | leifmadsen | could discuss on the mailing list when you have done your due diligence |
| 21:48.13 | kodomo | as a result, the snom is deregistered after a timeout |
| 21:48.19 | kodomo | and then reregisters |
| 21:48.39 | leifmadsen | jkroon: it'll be non-trivial |
| 21:48.40 | *** join/#asterisk Kyosh (~whoa@pool-74-108-19-39.nycmny.fios.verizon.net) |
| 21:48.55 | jkroon | lol, figured @ code. but usually i find it's better to first understand the design of the channel driver you're working with and also that which you're trying to add. |
| 21:49.07 | wcselby | kodomo and shortly thereafter the snom becomes unreachable |
| 21:49.15 | wcselby | is the snom staying online? |
| 21:49.16 | jkroon | leifmadsen, i didn't ask for easy, if it was easy someone would have done it already. |
| 21:49.21 | kodomo | wcselby: starting from line 468, you can see the tcpdump |
| 21:49.50 | *** join/#asterisk TimeRider (~steve@92.40.247.208.threembb.co.uk) |
| 21:49.52 | kodomo | wcselby: it is the asterisk initiating a FIN sequence, and the INVITE msg. receives a RST reaction |
| 21:50.20 | kodomo | so it really appears to be asterisk closing the connection |
| 21:50.31 | jkroon | leifmadsen, thanks, i'll spend some time looking at the iax2 code and trying to understand what exactly t38 does ito the fax protocol itself, it seems udptl is merely the transport layer when SIP is involved... |
| 21:50.40 | kodomo | the snom immediately drops the 'registered' LED upon TLS teardown |
| 21:50.57 | wcselby | kodomo i'm not sure. have you checked the issues tracker for tls issues? |
| 21:51.21 | kodomo | wcselby: starting from line 1541, you can see the snom's side of the story (SIP dump) |
| 21:51.42 | kodomo | wcselby: I did, as best I could... there's a hell lot of issues in the tracker ;) |
| 21:51.45 | wcselby | kodomo is this you -> https://issues.asterisk.org/jira/browse/ASTERISK-19003 |
| 21:52.06 | kodomo | wcselby: yes |
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| 21:52.16 | wcselby | kodomo lol |
| 21:52.28 | wcselby | it'll have to be looked at by someone better than me, sorry |
| 21:53.04 | kodomo | sorry, but I stumbled across the channel after writing the report, otherwise I'd have checked in here first ;) |
| 21:53.12 | wcselby | no worries |
| 21:53.17 | vetal | __OpenGL__, Why in issue.asterisk.org, said that it is configuration issue? if not? |
| 21:53.24 | __OpenGL__ | vetal as I said earlier it sounds like some issue at the other device; new compatability issue with 1.8 or network issue to me |
| 21:53.31 | kodomo | wcselby: but I thought it might be worth a shot nonetheless... can you see why this looks like a bug to me? |
| 21:53.36 | wcselby | you may ask leifmadsen if he knows who the person to handle that would be, because it seems to be pretty clearly over my head :) |
| 21:53.38 | *** join/#asterisk tekoholic (~quassel@97-118-209-77.hlrn.qwest.net) |
| 21:53.43 | wcselby | kodomo definitely |
| 21:54.01 | kodomo | leifmadsen... ok... I'll try - thx for your time :) |
| 21:54.08 | __OpenGL__ | im presuming your 1.6 to 1.8 config is similar... as ive found 1.6 configs to be fairly portable to 1.8 |
| 21:54.09 | wcselby | no worries :) |
| 21:54.15 | kodomo | leifmadsen: do you have a minute (or more? ;) ) |
| 21:54.23 | leifmadsen | eh? |
| 21:54.34 | wcselby | he's not going to fix it, he's just the bug marshall, he may know who's going to look at ti |
| 21:54.35 | wcselby | it* |
| 21:54.43 | kodomo | ah - ic |
| 21:55.02 | kodomo | leifmadsen: was referred to you by wcselby, as you can see... context...: |
| 21:55.07 | kodomo | https://issues.asterisk.org/jira/browse/ASTERISK-19003 |
| 21:55.17 | kodomo | http://pastebin.com/209JVSJT |
| 21:57.07 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 21:57.30 | vetal | __OpenGL__, please look at dump https://issues.asterisk.org/jira/browse/ASTERISK-18940, there is now issues from other side, all packets come good |
| 21:57.59 | vetal | __OpenGL__, file myDebugLog |
| 21:59.20 | wcselby | afk for a bit |
| 21:59.30 | __OpenGL__ | did the config change from 1.6 to 1.8 ? |
| 21:59.38 | __OpenGL__ | vetal, ^^ |
| 22:00.05 | kodomo | leifmadsen: do you have an idea whom I might talk to about this issue? |
| 22:00.19 | leifmadsen | kodomo: just the asterisk-users mailing list |
| 22:00.43 | kodomo | ic - thx |
| 22:00.54 | leifmadsen | looks like you provided all the info |
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| 22:02.00 | kodomo | collected what I thought might help... as I told wcselby: I couldn't figure a reason for asterisk to close the connection without waiting for the new INVITE, so it looked like a bug to me |
| 22:02.24 | vetal | __OpenGL__, after install 1.8, I use it configs, I do not use configs from 1.6 |
| 22:02.42 | __OpenGL__ | vetal so whats changes is far from just the version |
| 22:02.51 | kodomo | I wondered whether it might relate to the principle stated in https://issues.asterisk.org/jira/browse/ASTERISK-14830 (that TLS/TCP wouldn't need acks anyway) |
| 22:03.10 | kodomo | ...or at least ACK resends ;) |
| 22:03.18 | __OpenGL__ | vetal, get a config working with 1.6, keep it basic and simple... just the lines you need in extensions.conf and sip.conf to make it easy to debug |
| 22:03.25 | vetal | __OpenGL__, no changes, only version of asterisk. |
| 22:03.40 | __OpenGL__ | then make a backup copy of the config, install asterisk 1.8 and retest |
| 22:04.03 | vetal | __OpenGL__, I try this, same issue |
| 22:04.12 | __OpenGL__ | when I installed 1.8 I make uninstall the 1.4 first... |
| 22:04.33 | __OpenGL__ | which meant all config was gone |
| 22:04.37 | vetal | __OpenGL__, also I try all settings to change separatly and retest, same |
| 22:05.05 | __OpenGL__ | and your configuration for 1.6 that worked was? your configuration that didnt work in 1.8 was? |
| 22:06.05 | vetal | __OpenGL__, i try working config from 1.6 to 1.8 - not working |
| 22:08.10 | __OpenGL__ | vetal im concerned that earlier you said you took the demo config from 1.6 and modified it |
| 22:08.54 | __OpenGL__ | then you tried that same config in 1.8 and it didnt work... the basic config is really really complicated; you need to first strip the config down to make the number of things that could go wrong smaller |
| 22:08.59 | vetal | __OpenGL__, I took configs by "make samples" |
| 22:09.14 | __OpenGL__ | vetal make samples config for sip.conf is complex |
| 22:09.25 | __OpenGL__ | vetal likewise extensions.conf and extensions.ael is complex |
| 22:09.53 | vetal | __OpenGL__, there is a little strings, that are not commented by ; |
| 22:10.47 | __OpenGL__ | imo the only way to have hope in hell of troubleshooting this is make uninstall 1.8 then follow the instructions for remove all config files. install 1.6 make samples then mv sip.conf sip.conf.orig likewise for extensions.conf and extensions.conf.orig |
| 22:11.18 | __OpenGL__ | and also extensions.ael ro extensions.ael.orig |
| 22:11.28 | __OpenGL__ | so you have a clean ish slate to start from |
| 22:12.08 | vetal | __OpenGL__, I try, same issue |
| 22:12.20 | vetal | __OpenGL__, all is default |
| 22:12.55 | r0m|u | p3nguin: |
| 22:13.07 | __OpenGL__ | so paste your entire simple config for sip.conf and extensions.conf into pastebin please |
| 22:13.40 | r0m|u | p3nguin: Verbose(${CALLERID(num)} is calling ${EXTEN}) <----- no 0 after Verbose( ? |
| 22:16.56 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 22:17.27 | vetal | __OpenGL__,http://pastebin.com/NTXgtsS6 |
| 22:17.34 | __OpenGL__ | looks |
| 22:18.19 | mjordan | vetal: are your peers sending you OPTION packets? |
| 22:18.26 | __OpenGL__ | vetal, what is the configuration of your nat device like? |
| 22:19.36 | __OpenGL__ | vetal, earlier you said you had no nat, your config suggests you have nat. what is the ip of your asterisk box, what is the ip of your 00002 peer? |
| 22:21.28 | vetal | __OpenGL__, no i got NAT, but when I test without NAT i've got same issue |
| 22:22.09 | __OpenGL__ | what is the configuration of your nat device; what kind of nat device are you working with |
| 22:24.18 | vetal | __OpenGL__, now it is ADSL modem |
| 22:24.27 | __OpenGL__ | the issue you describe could so easilly be caused by the nat... |
| 22:25.23 | vetal | __OpenGL__, now I am testing, and got voice two way, but after I press hangup I've got Lagged peer ez |
| 22:25.34 | __OpenGL__ | ive run out of time to help you unfortunately but i have a few words of advice. simplify your testing environment. remove variables like nat etc as these can cause troubles that require more advanced troubleshooting |
| 22:26.27 | __OpenGL__ | remember that some adsl routers dont just pass your voip traffic through, some try to do all kinds of weird rubbish that causes hard to diagnose issues. for example thomson routers have a sip helper that broke my sip connection for asterisk till disabled |
| 22:26.39 | __OpenGL__ | netgear have sipalg sometimes which can cause issues |
| 22:27.02 | __OpenGL__ | some nat devices have ddos protection which can often detect rtp streams as ddos and shut them off |
| 22:27.26 | __OpenGL__ | ive wasted many an hour troubleshooting router related problems |
| 22:27.41 | wcselby | r0m|u the number after verbose( just means what level of verbosity do you want it to appear on |
| 22:27.43 | wcselby | by default it's 1 |
| 22:27.57 | wcselby | if you want it on 0, then you use Verbose(0, blah blah) |
| 22:28.34 | __OpenGL__ | good luck and dont give up; but DO try to simplify your setup. its honestly the first thing id do. right now you have too many variables |
| 22:28.57 | __OpenGL__ | wcselby thanks ive just realised why my debug messages never came up :-D |
| 22:29.06 | wcselby | why's that? |
| 22:29.11 | wcselby | oh lol\ |
| 22:29.13 | __OpenGL__ | Verbose(0, blabla) would be all the time? |
| 22:29.23 | wcselby | should be yeah |
| 22:29.28 | __OpenGL__ | cheers |
| 22:29.29 | wcselby | i don't know, never tried setting it to 0 |
| 22:29.37 | wcselby | :) |
| 22:30.03 | __OpenGL__ | ill play |
| 22:31.54 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-zxtukodsnvcijyqb) |
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| 23:31.02 | citywok | has anybody written a good test/monitor script to make sure an asterisk deployment is fully functional / processing calls? |
| 23:31.28 | *** join/#asterisk timeshell (~chatzilla@gw.lusi.on.ca) |
| 23:31.32 | timeshell | leifmadsen ping |
| 23:31.39 | citywok | i'm thinking like something that makes a test call every few minutes just to make sure it goes through |
| 23:33.07 | pdtpatrick1 | isn't there an snmp OID for asterisk? im not positive. But if there is, maybe you can get information from that? |
| 23:33.23 | pdtpatrick1 | i doubt if it would tell you anything about calls |
| 23:33.43 | pdtpatrick1 | but in your case, you can always have something connect to the AMI and originate a call and see what happens |
| 23:33.53 | citywok | if you set up SNMP you can see calls, but for example on friday our polycom ip6000 wasn't in a call anymore, but * thought it was. so when the nightly "core restart gracefully" was run it stopped processing calls waiting for the ghost channel to clear |
| 23:34.36 | citywok | which of course wasn't going to happen because the channel wasn't really open, so it wasn't going to be closed |
| 23:34.37 | WIMPy | Maybe you shoule use 'when convenient' instead? |
| 23:34.50 | citywok | WIMPy: that's exactly what i've done actually |
| 23:35.04 | WIMPy | Do you have session timers active? |
| 23:35.11 | timeshell | When attaching files to an issue, how to restrict access to the files to the developers only? |
| 23:35.34 | citywok | Sess-Timers : Accept; Sess-Refresh : uas; Sess-Expires : 1800 secs |
| 23:36.00 | citywok | the call had been going for at least 8 hours |
| 23:36.05 | citywok | "call" |
| 23:36.42 | WIMPy | Looks like you hit some bug then. |
| 23:36.56 | vetal | citywok, do you have rtptimeot set? |
| 23:37.17 | citywok | vetal: i didn't know that was a setting |
| 23:37.46 | timeshell | what is a confluence user? |
| 23:37.58 | citywok | vetal: thanks for the tip, i'll set that :) |
| 23:42.15 | *** join/#asterisk cyford (~allen@c-24-98-175-41.hsd1.ga.comcast.net) |
| 23:42.21 | cyford | hi is anyone an expert in a2billing in here? |
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| 23:59.41 | pdtpatrick1 | what is the pgsql module called that asterisk needs for cdr_pgsql? I've installed Postgresql-9.1 and yet it still marks cdr_pgsql with XXX in make menuselect |
| 23:59.57 | *** join/#asterisk vetal (~chatzilla@96-13-134-95.pool.ukrtel.net) |
