| 00:13.14 | *** join/#asterisk dundel (~daniel@190.98.82.213) |
| 00:17.18 | *** join/#asterisk jasonwert (~jasonwert@71.89.137.28) |
| 00:26.59 | file | yawnnnnnnn |
| 00:47.14 | lvlinux | is there a way to record echo on a call??? |
| 00:48.09 | lvlinux | I have a client that has a system that is giving a bad echo on PSTN calls---I had them record with automixmon---sounded 100% fine to me, no echo at all. |
| 00:48.27 | lvlinux | but they said there was echo on the call that they recorded. |
| 00:49.21 | lvlinux | the fxo card is a Sangoma B600 if that makes any difference. |
| 00:49.30 | bibz2 | D-Fender, hi there :) |
| 00:49.55 | lvlinux | and I have OSLEC turned on (at least it says it's active :-) |
| 00:52.07 | lvlinux | ideas anyone? |
| 00:55.25 | WIMPy | WHERE has that echo been heard? |
| 00:56.18 | lvlinux | the Asterisk side is where they here it, not the remote parties. |
| 00:57.04 | lvlinux | and they are across the country so I can't go hear it myself unfortunately. |
| 00:57.23 | WIMPy | Then I don't see how that wouldn't be recorded. |
| 00:58.18 | lvlinux | that's what i thought---but they insist that it's there and very bad. |
| 00:58.31 | MRH2 | get them to record with their smartphone and send you the file |
| 00:58.31 | lvlinux | and they affirmed that they were hearing it on the recorded call. |
| 00:59.07 | MRH2 | if it is echo on speaker |
| 01:00.07 | bibz2 | when I call a number which is routed to the asterisk, the asterisk dials out to a mobile phone and I can establish a call and even talk to the person on the other end, but after a few seconds there is a weird noise and we can't hear each other anymore.. here is the call: http://pastebin.com/jLQY5ziu (full debug, sorry about that) |
| 01:02.29 | lvlinux | MRH2: i think I will try that---hard to diagnose when I can't hear it myself |
| 01:02.39 | MRH2 | is it the same people they call |
| 01:02.43 | MRH2 | could be the other end |
| 01:02.54 | lvlinux | no they say it's all calls |
| 01:03.11 | MRH2 | do they use handsfree speakerphone |
| 01:03.13 | MRH2 | ? |
| 01:03.17 | [TK]D-Fender | bibz2: pastebin your [1202] |
| 01:03.28 | [TK]D-Fender | bibz2: masking only the secret |
| 01:03.32 | bibz2 | 1202? the extension? |
| 01:03.39 | [TK]D-Fender | sip.conf section |
| 01:03.48 | lvlinux | MRH2: I don't think so---I believe the echo is there with the handset. |
| 01:03.51 | bibz2 | there is no 1202 |
| 01:03.52 | WIMPy | bibz2: Do you have ICE enabled or something? |
| 01:03.54 | [TK]D-Fender | or whatever uses that as the authuser |
| 01:03.57 | bibz2 | I'll paste the whole |
| 01:04.13 | MRH2 | they have it the right way round? |
| 01:04.37 | [TK]D-Fender | bibz2: SIP/2.0 200 OK From: "1202" <sip:1202@192.168.3.217>;tag=e66b05bae3c71d55 CSeq: 43728 REGISTER |
| 01:04.42 | lvlinux | you mean the receiver? haha i sure hope so :) |
| 01:04.51 | [TK]D-Fender | bibz2: Your server sure seems to think there is a user with that name |
| 01:05.17 | MRH2 | could be high volume plus latency |
| 01:05.20 | bibz2 | I have a table which is called sipusers.. there are some users inside |
| 01:05.46 | MRH2 | like if you call a mobile and have the volumes right up |
| 01:05.52 | *** join/#asterisk slav3_kitten (~frankthet@unaffiliated/slav3-kitten/x-0866809) |
| 01:05.53 | lvlinux | i backed the txgain off 6db earlier, so maybe that will make a difference? |
| 01:06.18 | lvlinux | I thought it could be the volume (I've tried lots of echocanceller settings with no change) |
| 01:06.57 | bibz2 | weird.. there is no 1202 user.. |
| 01:07.18 | MRH2 | try making an extension that records them and plays it back |
| 01:07.48 | lvlinux | what do you mean? |
| 01:07.49 | *** join/#asterisk bmurt (~brendan@208-58-116-232.c3-0.upd-ubr1.trpr-upd.pa.cable.rcn.com) |
| 01:08.01 | MRH2 | even if they leave a voicemail |
| 01:08.04 | MRH2 | for themselves |
| 01:08.06 | bibz2 | ok, here is my sip.conf: http://pastebin.com/xvQrmg1P |
| 01:08.23 | MRH2 | assume this is ok |
| 01:08.39 | MRH2 | voip is ok |
| 01:08.42 | MRH2 | is just to pstn |
| 01:08.48 | lvlinux | yes I believe there is no echo within the office, just when calling out the FXO |
| 01:09.12 | bibz2 | this issue is driving me crazy. we have 50% lost calls in the last 2 weeks :( |
| 01:09.36 | bibz2 | long time I thought its because of NAT so we removed the NAT |
| 01:10.13 | bibz2 | we thought it may be the big queries that are being executed on the SQL server on every call so we upgraded the server .. didn't help |
| 01:10.25 | bibz2 | new internet provider, didn't help |
| 01:10.54 | bibz2 | I can't even find in the debug anything helpful. There is no error when the audio interrupts.. nothing |
| 01:11.04 | [TK]D-Fender | bibz2: You are masking anything that could possibly match that entry I clearly showed you. |
| 01:11.15 | [TK]D-Fender | bibz2: Go really look... because you aren't showing us. |
| 01:11.20 | bibz2 | ok |
| 01:11.37 | *** join/#asterisk bmurt (~brendan@208-58-116-232.c3-0.upd-ubr1.trpr-upd.pa.cable.rcn.com) |
| 01:11.56 | bibz2 | the user 1202 is not existent in the sip.conf. and also not in the sipusers table |
| 01:12.05 | bibz2 | what are some other places to look? |
| 01:12.41 | [TK]D-Fender | Look at the IP it's registering from |
| 01:13.00 | MRH2 | i guess recording is helpful (and proof) - maybe try without the hardware echo cancellation if possible. |
| 01:13.30 | lvlinux | the card doesn't have any hardware EC, only software. |
| 01:14.26 | lvlinux | i had it on MG at first and when they complained the first time I switched to OSLEC---they said there was no change in the echo. |
| 01:15.08 | WIMPy | What kind of phone are they using? |
| 01:19.18 | lvlinux | Yealink T46G |
| 01:19.34 | lvlinux | but I had them plug in a Polycom 430 and they said it was the same echo on it. |
| 01:21.07 | *** join/#asterisk 6JTAA5RKQ (~quassel@89.101.203.26) |
| 01:23.07 | MRH2 | does the phone do echo cancellation as well? |
| 01:23.20 | *** join/#asterisk felimwhiteley (~quassel@89.101.203.26) |
| 01:23.28 | MRH2 | cause i don;t u think u want both |
| 01:26.58 | bibz2 | but D-Fender.. I don't understand what the peer 1202 has to do with the call I initiated |
| 01:27.26 | bibz2 | I have tried to log into that IP, seems to be a Grandstream testing device we have in our office |
| 01:27.40 | bibz2 | I can't imagine how it could influence other calls |
| 01:31.44 | [TK]D-Fender | ... |
| 01:31.55 | [TK]D-Fender | that call you actually placed in there did NOT say NAT |
| 01:32.28 | [TK]D-Fender | The only thing that did was a REGISTER |
| 01:32.41 | [TK]D-Fender | You are clearly not even looking at the output you are providing. |
| 01:32.53 | *** join/#asterisk giesen (~ggiesen@2001:4900:2:650::180) |
| 01:33.34 | bibz2 | but my call-id is 6943857619bf0bcb5b15bcd8219f802b ... |
| 01:33.58 | [TK]D-Fender | [19:41]bibz2my asterisk isn't really behind a NAT but there are a lot of packets which state (Transmitting NAT) in the header. is that normal? |
| 01:33.59 | bibz2 | sorry if I explained it wrong, or if I just don't understand it |
| 01:34.01 | [TK]D-Fender | You asked this earlier |
| 01:34.25 | [TK]D-Fender | "is tht normal?" <- |
| 01:34.31 | bibz2 | I have removed the nat=force_rport,comedia line which was "hidden" on top of sip.conf |
| 01:34.40 | bibz2 | I thought that should've done it.. |
| 01:34.40 | *** join/#asterisk giesen (~ggiesen@2001:4900:2:650::180) |
| 01:34.42 | *** join/#asterisk u0m3 (~u0m3@92.80.93.41) |
| 01:35.02 | [TK]D-Fender | Doesn't matter where it is.... it matters what you TELL it about itself and other peers that may try to connect |
| 01:35.21 | bibz2 | so its still a NAT problem, even if the packet corresponding to my call states NO NAT? |
| 01:35.50 | [TK]D-Fender | No, we are talking about 2 different things |
| 01:36.11 | [TK]D-Fender | you have changed topic and still not realized they are SEPARATE points |
| 01:36.12 | bibz2 | only that call (1202) seems to be natted.. shouldn't affect my call.. or should it? |
| 01:36.25 | bibz2 | well that wasn't my intention |
| 01:36.30 | [TK]D-Fender | You asked the question |
| 01:36.36 | [TK]D-Fender | and provided the debug,. |
| 01:36.40 | bibz2 | seems like I'm confused now |
| 01:37.09 | [TK]D-Fender | at elast in as much as remembering the questions you asked. |
| 01:37.16 | [TK]D-Fender | least* |
| 01:37.31 | [TK]D-Fender | As for your audio... I see several peers with no reinvite rules. |
| 01:37.47 | [TK]D-Fender | You should just prevent them explicitly in all your peers and under [general] |
| 01:37.49 | bibz2 | ok, I try it one more time to describe the problem. without taking the first question about NAT in conclusion |
| 01:38.04 | [TK]D-Fender | [pbx3] <- not set |
| 01:38.33 | [TK]D-Fender | [sipgate-3] <- missing the "nat=no" they certainly require |
| 01:38.54 | [TK]D-Fender | You have done little pieces of your job spread across all these entires and have not been thorough or consistent |
| 01:39.01 | [TK]D-Fender | Go clean all of them up |
| 01:39.18 | bibz2 | I should either work on my discipline or sleep more |
| 01:39.24 | bibz2 | thanks |
| 01:44.04 | bibz2 | ok, set every peer to nat=no. |
| 01:45.34 | bibz2 | reinvite rules should be set? seems like I have commented it out for some reason.. |
| 01:45.45 | bibz2 | probably while experimenting around in the beginning.. |
| 01:47.08 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 01:48.53 | [TK]D-Fender | directmedia=no |
| 01:49.06 | [TK]D-Fender | \"canreinvite" is long been deprecated |
| 01:50.19 | bibz2 | oh you were right. seems like pbx3-out has this set, but pbx3 not... |
| 01:56.17 | bibz2 | damnit, now its too late to test anytihng since there aren't any agents signed in anymore. should better change my working hours |
| 02:05.43 | *** join/#asterisk stevePearPear (~stevePear@202.166.82.164) |
| 02:21.59 | *** join/#asterisk Cynagen (~cynagen@ip70-190-135-8.ph.ph.cox.net) |
| 02:22.49 | *** join/#asterisk SpeedEvil (~quassel@tor/regular/SpeedEvil) |
| 02:53.34 | *** join/#asterisk CrashHD (~na@204-195-127-247.wavecable.com) |
| 03:01.44 | *** join/#asterisk gringo (~gringo@unaffiliated/gringo) |
| 03:02.09 | *** join/#asterisk SpeedEvil (~quassel@tor/regular/SpeedEvil) |
| 03:05.46 | *** join/#asterisk arosen1 (~arosen@geni/support/arosen) |
| 03:06.20 | arosen1 | Can you have one phone number and load balance that over several asterisk boxes? I've been googling around and it seems like you need to use: OpenSIPS/OpenSER/Kamillo but all the things i'm finding are ~2009 time peroid |
| 03:37.38 | *** join/#asterisk timahvo1 (~rogue@197.237.134.227) |
| 03:41.39 | *** join/#asterisk fstd_ (~fstd@unaffiliated/fisted) |
| 03:58.50 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 04:01.07 | *** join/#asterisk stevePearPear (~stevePear@203.126.171.206) |
| 04:03.53 | *** join/#asterisk gerhard7 (~gerhard7@77-172-20-151.ip.telfort.nl) |
| 04:04.32 | *** join/#asterisk wolrah_ (~wolrah@24.239.210.140) |
| 04:07.59 | *** join/#asterisk bkruse (~Adium@24.42.207.11) |
| 04:13.06 | *** join/#asterisk darkdrgn2k3 (~darkdrgn2@69-165-131-20.dsl.teksavvy.com) |
| 04:13.21 | darkdrgn2k3 | whats the latest polycom firmware for asterisk |
| 04:41.30 | [TK]D-Fender | Thre is no such thing as "polycom firmware for asterisk" |
| 04:41.42 | [TK]D-Fender | polycom firmware is for polycom |
| 04:41.53 | [TK]D-Fender | And the latest version... depends on the model. |
| 05:00.53 | *** join/#asterisk reconwireless (uid10170@gateway/web/irccloud.com/x-gvddbmrfixacabvm) |
| 05:10.12 | *** join/#asterisk ttyS3 (~o@gateway/tor-sasl/omlib) |
| 05:10.25 | *** join/#asterisk spicyramen_ (~Adium@c-98-210-160-181.hsd1.ca.comcast.net) |
| 05:14.29 | *** join/#asterisk stevePearPear (~stevePear@203.126.171.206) |
| 05:21.18 | darkdrgn2k3 | [TK]D-Fender: Latest firmware has a note that its for LYNC only |
| 05:21.50 | darkdrgn2k3 | "Polycom UC Software 4.1.1 is for use with Microsoft® LyncTM Server and using a single registered line only." |
| 05:27.34 | [TK]D-Fender | Polycom Unified Communications (UC) Software 5.1.1 Rev B is a general release for all open SIP platforms including Microsoft Lync Server 2010 and Microsoft Lync Server 2013 |
| 05:27.41 | [TK]D-Fender | Tha implies open as well. |
| 05:28.57 | [TK]D-Fender | And clearly much newer than your ver you quoted |
| 05:34.47 | darkdrgn2k3 | http://downloads.polycom.com/voice/voip/sip_sw_releases_matrix.html |
| 05:34.53 | darkdrgn2k3 | matrix doesnt go that high !?!? |
| 05:35.14 | *** join/#asterisk gringo (~gringo@unaffiliated/gringo) |
| 05:35.32 | [TK]D-Fender | You aren't looking very hard or being very specific |
| 05:36.24 | darkdrgn2k3 | i have a IP 335 |
| 05:36.33 | darkdrgn2k3 | the support site doesnt specify 5.x either |
| 05:36.50 | [TK]D-Fender | Context matters |
| 05:37.01 | [TK]D-Fender | because ther was a link of that page for OTHER Polycom models |
| 05:37.18 | [TK]D-Fender | Which DOES go higher and support standa along with Lync |
| 05:37.50 | [TK]D-Fender | You asked about firmware and waited this long before telling us what model you were looking for. |
| 05:38.13 | darkdrgn2k3 | yes you are right |
| 05:41.13 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 05:46.36 | [TK]D-Fender | 3.3.53.3.5 as listed there... which is what I run on mine... |
| 05:46.40 | [TK]D-Fender | 3.3.5* |
| 05:46.48 | [TK]D-Fender | heads to bed |
| 05:54.04 | *** join/#asterisk jiku (~jiku@1.22.175.229) |
| 05:57.36 | *** join/#asterisk stevePearPear (~stevePear@202.166.82.164) |
| 06:21.10 | *** join/#asterisk Cyna (~cynagen@ip70-190-135-8.ph.ph.cox.net) |
| 06:21.49 | *** join/#asterisk greynix (~greynix@unaffiliated/greynix) |
| 06:26.06 | *** join/#asterisk spicyramen_ (~Adium@c-98-210-160-181.hsd1.ca.comcast.net) |
| 06:26.33 | spicyramen_ | anyone using HD video endpoints registered to asterisk 720p+? |
| 06:44.49 | *** join/#asterisk dsr (6a33f31d@gateway/web/freenode/ip.106.51.243.29) |
| 06:45.13 | dsr | hi |
| 06:45.25 | dsr | what is the recommended memory requirement for asterisk 12.4? |
| 06:47.04 | *** join/#asterisk mirela666 (~mirko.bra@iecommailer.itaf.eu) |
| 06:57.36 | *** join/#asterisk mirela666 (~mirko.bra@iecommailer.itaf.eu) |
| 07:02.45 | *** join/#asterisk babak (uid19622@gateway/web/irccloud.com/x-lbjecjbguucmkeep) |
| 07:03.25 | *** join/#asterisk jhlavacek (~jirka@87.89.218.63) |
| 07:16.36 | *** join/#asterisk WHiZZi (~whizzi@D4B2620B.static.ziggozakelijk.nl) |
| 07:17.21 | *** join/#asterisk geoid (~geoid@host90-152-1-170.ipv4.regusnet.com) |
| 07:23.00 | *** join/#asterisk gringo (~gringo@unaffiliated/gringo) |
| 07:33.42 | wdoekes | dsr: those questions are impossible to answer. it depends on the kind of load you expect |
| 07:36.31 | wdoekes | and even then it's far from trivial. our office pbx (does practially nothing), uses (together with OS/daemons) about 100MB on a 1GB system |
| 07:43.32 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:ac8b:13fd:9a8b:f605) |
| 07:47.08 | *** join/#asterisk gringo (~gringo@unaffiliated/gringo) |
| 08:07.07 | *** join/#asterisk areski (~areski@80.174.128.86.dyn.user.ono.com) |
| 08:07.38 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 08:10.56 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 08:18.48 | *** join/#asterisk ttyS3 (~o@gateway/tor-sasl/omlib) |
| 08:26.36 | *** join/#asterisk wanna (~vincent@91.178.203.216) |
| 08:39.47 | *** join/#asterisk mnathani (~mnathani@butterfly.winvive.com) |
| 08:41.11 | *** join/#asterisk evil_gordita (robert@ip70-188-56-12.rn.hr.cox.net) |
| 08:56.42 | *** join/#asterisk calum_ (~calum_@94.197.121.204.threembb.co.uk) |
| 08:58.25 | *** join/#asterisk W0rmDr1nk (~wormdrink@unaffiliated/wormdrink) |
| 09:11.33 | *** join/#asterisk af_ (~af@93-43-38-121.ip90.fastwebnet.it) |
| 09:21.39 | *** join/#asterisk workingcats_ (~workingca@212.122.48.77) |
| 09:42.28 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 09:50.52 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 10:01.54 | *** join/#asterisk MaliutaLap (~nobusines@unaffiliated/maliuta) |
| 10:02.34 | *** join/#asterisk areski (~areski@177.Red-83-45-113.dynamicIP.rima-tde.net) |
| 10:03.14 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 10:10.58 | *** join/#asterisk D-Boy (~D-Boy@unaffiliated/cain) |
| 10:17.27 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 10:34.54 | *** join/#asterisk infobot (ibot@rikers.org) |
| 10:34.54 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: Asterisk 11.11.0 (2014/07/10), 1.8.29.0 (2014/07/10); Standard: Asterisk 12.4.0 (2014/07/10); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- Read the Code of Conduct bit.ly/1hH6P22 |
| 10:57.53 | *** join/#asterisk [[thufir]] (~thufir@S0106c8fb2677e8ca.vs.shawcable.net) |
| 11:02.40 | *** join/#asterisk creativx (~creadurex@226.62-97-205.bkkb.no) |
| 11:09.32 | *** join/#asterisk stevePearPear (~stevePear@202.166.82.164) |
| 11:09.51 | *** join/#asterisk Gugge (gugge@kriminel.dk) |
| 11:20.13 | *** join/#asterisk WHiZZi (~whizzi@D4B2620B.static.ziggozakelijk.nl) |
| 11:21.22 | WHiZZi | Is it normal that Asterisk 11.10 doesn't actually reload sip when the sip.conf-file is untouched and only the include files are changed ? |
| 11:23.52 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 11:24.29 | *** part/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 11:29.31 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 11:39.24 | *** join/#asterisk ttyS3 (~o@gateway/tor-sasl/omlib) |
| 11:42.58 | *** join/#asterisk SpeedEvil (~quassel@tor/regular/SpeedEvil) |
| 11:43.06 | *** join/#asterisk mirela666 (~mirko.bra@iecommailer.itaf.eu) |
| 11:46.12 | *** join/#asterisk mirela666 (~mirko.bra@iecommailer.itaf.eu) |
| 11:50.20 | *** join/#asterisk mirela666 (~mirko.bra@iecommailer.itaf.eu) |
| 12:06.00 | wdoekes | WHiZZi: https://issues.asterisk.org/jira/browse/ASTERISK-23683 fixed in 11.11 |
| 12:06.09 | WHiZZi | ok, tnx :) |
| 12:18.25 | *** join/#asterisk stevePearPear (~stevePear@cm244.epsilon47.maxonline.com.sg) |
| 12:25.14 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 12:33.52 | *** join/#asterisk paolo_ (~smuxi@ip-62-143-36-139.hsi01.unitymediagroup.de) |
| 12:36.20 | paolo_ | [TK]-D-Fender: Hi, so you remember our last conversation. It was about directrtpsetup. I gave you a link to pastebin (debug and sip.conf). Did you have the time to take a look at it ? |
| 12:39.49 | *** join/#asterisk SpeedEvil (~quassel@tor/regular/SpeedEvil) |
| 12:42.52 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 12:44.43 | marceloamorim | guys, I had an issue with my asterisk, I just tested right now about this, I make a call and the side B answered but the audio wait like 2 or 3 secs to pass through |
| 12:44.57 | marceloamorim | the call loke is at the same second, but it isn`t |
| 12:45.20 | marceloamorim | "[2014-08-15 09:41:37] -- DAHDI/2-1 answered SIP/gerente_fantasminha-0000025c" -> " [2014-08-15 09:41:37] > 0x2416f40 -- Probation passed - setting RTP source address to 192.168.2.101:10002 " |
| 12:46.29 | *** join/#asterisk cw1972 (~cw1972@213.123.58.248) |
| 12:47.28 | *** join/#asterisk Draecos (~Draecos@106-69-0-55.dyn.iinet.net.au) |
| 12:57.17 | marceloamorim | anyone knows how to fix? |
| 13:01.11 | *** join/#asterisk bibz2 (bibz2@194-96-98-72.adsl.highway.telekom.at) |
| 13:01.15 | *** join/#asterisk MarkSX (~MarkSX@C-59-101-49-109.hay.connect.net.au) |
| 13:03.08 | *** join/#asterisk riess82 (~riessma@mail.p-riess.at) |
| 13:03.22 | *** join/#asterisk shmzadmin (uid28588@gateway/web/irccloud.com/x-uxprxnkwsbiejzoi) |
| 13:06.40 | omenSP | hey c: who |
| 13:06.48 | omenSP | is up this morning*] |
| 13:10.22 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 13:10.41 | bibz2 | any freelancers in here? |
| 13:10.42 | *** join/#asterisk babak (uid19622@gateway/web/irccloud.com/x-xyvuckyklqugixey) |
| 13:10.48 | omenSP | um |
| 13:10.56 | omenSP | I'm not |
| 13:11.23 | [TK]D-Fender | lances are expensive these days... |
| 13:11.51 | [TK]D-Fender | And the time it takes to break a horse? |
| 13:12.02 | marceloamorim | [TK]D-Fender: can I ask you something? |
| 13:12.02 | [TK]D-Fender | You can't be serious... |
| 13:12.10 | [TK]D-Fender | SURELY YOU JOUST?! |
| 13:12.16 | [TK]D-Fender | :) |
| 13:12.18 | omenSP | ohey d-fender! |
| 13:12.24 | omenSP | you helped me out like a month ago. |
| 13:12.48 | marceloamorim | "[2014-08-15 09:41:37] -- DAHDI/2-1 answered SIP/gerente_fantasminha-0000025c" -> " [2014-08-15 09:41:37] > 0x2416f40 -- Probation passed - setting RTP source address to 192.168.2.101:10002 " |
| 13:13.16 | [TK]D-Fender | marceloamorim: Stop thinking that single lines like that hold the story.... |
| 13:13.33 | [TK]D-Fender | marceloamorim: SIP is involved and you aren't showing a full call with matching debug |
| 13:13.53 | *** join/#asterisk TSM2 (~the_softw@fw-lon2.wenn.com) |
| 13:14.45 | marceloamorim | this log on 09:41:37 but this isn`t true, after I get those DAHDI/2-1 answered the audio isn`t pass through yet |
| 13:16.09 | marceloamorim | it takes like 2 or 3 seconds to have tx and rx audio |
| 13:16.26 | *** join/#asterisk ThatDamnRanga (~wiretap@unaffiliated/wiretap) |
| 13:17.08 | *** join/#asterisk zerohalo (~zerohalo@2601:6:f80:224:3418:dedd:d0ec:4cfe) |
| 13:18.03 | *** join/#asterisk theron (~theron@66.220.145.150) |
| 13:20.06 | omenSP | Is there an example that I can base my outbound calling routes and trunks off of? I bought the 8-port FXO card and plugged all the trunks into it, then set it up as 8 trunks in the system. Upon restarting Asterisk, outside calls work for like 15 minutes and then just give a constant dial tone. |
| 13:20.16 | *** join/#asterisk bmurt (~brendan@8.39.115.8) |
| 13:21.14 | *** join/#asterisk gerhard7 (~gerhard7@77-172-20-151.ip.telfort.nl) |
| 13:22.12 | *** join/#asterisk cw1972 (~cw1972@213.123.58.248) |
| 13:23.20 | *** join/#asterisk file (~file@asterisk/developer-and-muffin-lover/file) |
| 13:23.20 | *** mode/#asterisk [+o file] by ChanServ |
| 13:24.24 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 13:25.34 | omenSP | Is there an example that I can base my outbound calling routes and trunks off of? I bought the 8-port FXO card and plugged all the trunks into it, then set it up as 8 trunks in the system. Upon restarting Asterisk, outside calls work for like 15 minutes and then just give a constant dial tone. |
| 13:26.59 | *** join/#asterisk Frojoe (Frojoe@2a01:7e00::f03c:91ff:fe70:bc74) |
| 13:29.03 | omenSP | What are DAHDI Channels? |
| 13:30.58 | WIMPy | I think you should start with the |
| 13:31.03 | WIMPy | ~book |
| 13:31.03 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 13:31.30 | omenSP | I'm using FreePBX though. |
| 13:32.22 | WIMPy | GUIs aren't supported here. Ask about that in #freepbx . |
| 13:32.28 | omenSP | Thank you |
| 13:32.30 | *** part/#asterisk omenSP (32490489@gateway/web/freenode/ip.50.73.4.137) |
| 13:32.59 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-28-58.cablep.bezeqint.net) |
| 13:37.20 | *** join/#asterisk navaismo (0c0adbe1@gateway/web/freenode/ip.12.10.219.225) |
| 13:38.37 | marceloamorim | [TK]D-Fender: http://pastebin.com/rWrstmSg |
| 13:38.53 | marceloamorim | I restart the asterisk to get the full.log |
| 13:46.19 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 13:48.56 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 13:49.15 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 13:49.36 | BlackDex | Hello there |
| 13:49.44 | BlackDex | we have an Digum TE133 PCI-E card |
| 13:50.19 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 13:50.22 | BlackDex | and when we configure the card with dahdi we always get 24 channels, so it seems that it uses T mode.. But i need E mode |
| 13:50.42 | BlackDex | i can't seem to get it to the E mode, don't know where to look |
| 13:51.24 | *** join/#asterisk yano (~yano@freenode/staff/yano) |
| 13:52.09 | hexanol | I've found something a little strange on 2 different asterisk while looking for a "no sound" problem |
| 13:52.16 | hexanol | one is a 11.9.0, the other a 11.11.0 |
| 13:52.35 | hexanol | the usage count of the "res_rtp_asterisk" module is high, 811 on one, 6 on the other |
| 13:52.47 | [TK]D-Fender | BlackDex: Got set the channels in your configs |
| 13:52.51 | hexanol | and there's lots of UDP socket (bind on a port in the RTP range) that are open |
| 13:53.01 | hexanol | but there's no active channels |
| 13:53.20 | hexanol | like, on the one with a 811 use count, there's around ~1600 open UDP socket in the RTP range |
| 13:53.33 | hexanol | and on the one with a 6 use count, there's 12 open UDP socket |
| 13:53.44 | hexanol | someone else has seen this kind of things on similar asterisk version ? |
| 13:55.00 | hexanol | I've not found yet on how to reproduce the problem; it's happening, but I don't know how |
| 13:58.15 | *** join/#asterisk lbazan (~LoKoMurdo@fedora/LoKoMurdoK) |
| 13:59.38 | *** join/#asterisk reconwireless (uid10170@gateway/web/irccloud.com/x-yunnfimgllpmzmer) |
| 13:59.44 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-chghihqpeqttgfue) |
| 13:59.45 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 14:28.00 | marceloamorim | I`ll test later about the problem I had with audio, I set canreinvite=yes on the sip.conf |
| 14:28.56 | BlackDex | [TK]D-Fender: I think i have found something.. Could be the some settings i need to change modprobe |
| 14:56.06 | *** join/#asterisk MarkSX (~MarkSX@C-59-101-49-109.hay.connect.net.au) |
| 15:05.40 | *** join/#asterisk lnb (~lnb@CPE4c5e0c417c51-CM602ad06bec2f.cpe.net.cable.rogers.com) |
| 15:09.05 | *** join/#asterisk lbazan (~LoKoMurdo@fedora/LoKoMurdoK) |
| 15:11.28 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 15:11.28 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 15:12.22 | Katty | my state is embarassing. |
| 15:13.28 | [TK]D-Fender | Its a homonym.... |
| 15:15.33 | Katty | [TK]D-Fender: you're a homonym! |
| 15:16.08 | [TK]D-Fender | UR MOM ...? ;) |
| 15:16.21 | Katty | your face! |
| 15:17.08 | *** join/#asterisk jasonwert (~jasonwert@75-134-81-98.static.aldl.mi.charter.com) |
| 15:17.51 | puzzled | hi |
| 15:17.59 | Katty | hey puzzled (= |
| 15:18.08 | puzzled | hi Katty |
| 15:18.12 | Katty | how're you dear? |
| 15:18.22 | puzzled | looking forward to the weekend :) |
| 15:18.27 | Katty | meeee too! |
| 15:18.30 | Katty | big plans? |
| 15:18.47 | puzzled | nope just relaxing |
| 15:19.01 | Katty | sometimes that's the best kind of weekend! |
| 15:23.44 | puzzled | Anyone have any experience with didlogic.com for DIDs? |
| 15:25.40 | *** join/#asterisk sgriepentrog (~sgriepent@107-220-72-223.lightspeed.brhmal.sbcglobal.net) |
| 15:28.09 | Nugget | puzzled: I tried to sign up for service with them last week for some australian DIDs but got thwarted because their online signup process was too manual |
| 15:29.01 | Nugget | they wanted me to scan the credit card and upload images of it and whatnot and that was enough of a deterrent for me to seek another provider |
| 15:29.41 | puzzled | Nugget: thanks, that sounds like 1995 wants their process back :) I'll pass then |
| 15:29.49 | Nugget | oh, and my ID and a copy of a utility bill |
| 15:30.00 | Nugget | I decided they didn't really want business customers |
| 15:32.10 | puzzled | that sounds insane |
| 15:32.33 | file | Nugget, well you ARE a troublemaker... |
| 15:32.57 | puzzled | oh and they are located in Hong Kong and I doubt my consumer rights I enjoy in the EU applies there |
| 15:40.19 | *** join/#asterisk anonymouz666 (~anonymouz@189-25-93-37.user.veloxzone.com.br) |
| 15:40.24 | anonymouz666 | I am back on! |
| 15:41.16 | anonymouz666 | sends a complete /var/log/asterisk/full to [TK]D-Fender and also a sip debug. |
| 15:41.23 | *** join/#asterisk af_ (~af@static-82-85-142-163.clienti.tiscali.it) |
| 15:43.05 | anonymouz666 | the new SIP channel driver is something that I'd love to test. |
| 15:43.55 | *** join/#asterisk timahvo1 (~rogue@197.237.134.227) |
| 15:59.29 | *** join/#asterisk theron (~theron@66.220.145.150) |
| 16:06.05 | bibz2 | any freelancers in here? |
| 16:06.52 | [TK]D-Fender | Told you earlier.. lances are expensive... don't expect them to be free... |
| 16:07.18 | [TK]D-Fender | Perhaps you should start with what you are actually trying to accomplish before people start lining up over things they can't help you with. |
| 16:09.00 | bibz2 | I still have calls which don't go through.. and I don't get it what the problem could be |
| 16:09.20 | bibz2 | the thing about is, that I'm trying to mantain/repair a pbx which is already 2 years old and quite big |
| 16:09.37 | bibz2 | I resolved all the NAT Problems I guess so |
| 16:10.46 | *** join/#asterisk fstd_ (~fstd@unaffiliated/fisted) |
| 16:11.48 | [TK]D-Fender | Show your failed calls. |
| 16:11.53 | [TK]D-Fender | "guessing" is bad... |
| 16:12.21 | *** join/#asterisk calum_ (~calum_@92.40.249.81.threembb.co.uk) |
| 16:16.06 | *** join/#asterisk danjenkins_ (~dan@nat/digium/x-hmnbzkqglgtxipcm) |
| 16:25.11 | Katty | [TK]D-Fender: here's my pastebin of a failed call. What did I do wrong???? http://pastebin.com/AjGXF7yV |
| 16:25.23 | *** join/#asterisk dundel (~daniel@190.98.82.213) |
| 16:25.46 | [TK]D-Fender | Well you finally broke double-digits with that one.... |
| 16:26.00 | [TK]D-Fender | You keep trying to get a laugh out of these, but no pun in ten did ;) |
| 16:26.15 | Katty | rolls eyes |
| 16:29.49 | *** join/#asterisk theron (~theron@66.220.145.150) |
| 16:31.27 | *** join/#asterisk dkwiebe (~dkwiebe@aglandcorp.com) |
| 16:32.55 | *** join/#asterisk phonebuff (~chatzilla@c-71-196-61-237.hsd1.fl.comcast.net) |
| 16:39.08 | zamba | i'm seeing lots of these messages in my system logs: Aug 15 17:26:23 tahiti asterisk[30498]: NOTICE[30522]: chan_sip.c:22546 in handle_request_invite: Sending fake auth rejection for device 100<sip:100@ip>;tag=6d247154 |
| 16:39.17 | zamba | what do they mean? can they be ignored/removed? |
| 16:41.45 | [TK]D-Fender | it means they are getting chanllenged,. |
| 16:41.52 | [TK]D-Fender | Doesn't mean they don' get accepted |
| 16:41.55 | [TK]D-Fender | Go look for that |
| 16:42.47 | *** join/#asterisk Iamnacho (~Iamnacho@ip72-213-56-241.om.om.cox.net) |
| 16:42.58 | zamba | is there a way to get rid of these messages? |
| 16:43.35 | [TK]D-Fender | Disable all "notice" in your logging |
| 16:47.33 | stevePearPear | Hi, whenever i reset Asterisk, it will automatically attempt to register all my peer with the siptrunk provider automatically |
| 16:47.45 | stevePearPear | can I only register when a call is attempted? |
| 16:48.09 | [TK]D-Fender | no |
| 16:49.13 | stevePearPear | [TK]D-Fender are you replying to me for the ânoâ ? |
| 16:50.38 | [TK]D-Fender | I am |
| 16:51.55 | stevePearPear | I am allowing my users to update the credentials for the siptrunk, so when too many of my users has invalid credential, the siptrunk provider block me as it thought that I am trying to brute force |
| 16:51.59 | stevePearPear | is there a way out of this? |
| 17:14.32 | anonymouz666 | surface 3 is such a good product |
| 17:15.10 | *** join/#asterisk calum_ (~calum_@cpc67428-harg5-2-0-cust142.7-1.cable.virginm.net) |
| 17:17.37 | *** join/#asterisk danjenkins__ (~dan@cpc11-folk2-2-0-cust171.1-2.cable.virginm.net) |
| 17:22.10 | *** join/#asterisk SpeedEvil (~quassel@tor/regular/SpeedEvil) |
| 17:22.46 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 17:22.47 | bibz2 | my asterisk fills the cdr table with double entries |
| 17:22.52 | bibz2 | why could that be |
| 17:23.07 | bibz2 | for every call there are 2 entries in the cdr table |
| 17:24.23 | [TK]D-Fender | exactly identical? |
| 17:25.12 | pabelanger | anybody using sipml5? Specifically, are you getting native (webrtc client side) ringback on dial? |
| 17:25.24 | pabelanger | I know asterisk and generate it, but want to see if the client side works |
| 17:25.30 | bibz2 | yes exactly identical |
| 17:25.42 | [TK]D-Fender | bibz2: probably a double-entry in CDR configs |
| 17:26.09 | bibz2 | even the uniqueid is identical |
| 17:26.09 | bibz2 | ok |
| 17:34.41 | navaismo | pabelanger: you mean hear the ring tone when is dialing? |
| 17:35.57 | *** join/#asterisk ttyS3 (~o@gateway/tor-sasl/omlib) |
| 17:38.05 | navaismo | I use https://github.com/navaismo/Elastix-Agent_Console_WebRTC/blob/master/modules/themes/default/js/ml5.js#L331 and https://github.com/navaismo/Elastix-Agent_Console_WebRTC/blob/master/modules/themes/default/js/ml5.js#L152 |
| 17:38.15 | navaismo | based on the doubango's example |
| 17:39.48 | bibz2 | no double entry in the cdr configs. checked cdr_odbc also.. |
| 17:39.51 | bibz2 | what the heck |
| 17:44.27 | [TK]D-Fender | check your backend status |
| 17:45.05 | mjordan | bibz2: which version of Asterisk are you using? |
| 17:47.22 | pabelanger | navaismo, yes, from sipml5 dial a number. Do you hear local ringback? |
| 17:48.15 | bibz2 | mjordan: 11.6-cert2 |
| 17:48.35 | navaismo | pabelanger: Yes, but i need to call the function and the audiofile must exist |
| 17:48.38 | bibz2 | the problem occured yesterday, but I don't know why. I have made some alternations but didn't change anything in the CDR config |
| 17:48.42 | mjordan | k. You probably do have a configuration issue somewhere then. What all CDR backends do you have loaded? |
| 17:48.44 | WIMPy | Are IAX timestamps somehow related to realtime? |
| 17:49.21 | anonymouz666 | pabelanger: are you using ARI heavily? |
| 17:49.41 | bibz2 | mjordan: http://pastebin.com/tS9akCKg |
| 17:51.23 | mjordan | you have both odbc and adaptive odbc loaded. |
| 17:51.29 | mjordan | Just sayin. |
| 17:51.34 | mjordan | You may want to double check that. |
| 17:52.43 | doobeh | I know credit card machines have issues with the compression algo's-- if I have 8 FXO ports and 16 FXS ports on my channel bank-- will I still run into compression problems, since it's not going up to a sip trunk? |
| 17:52.55 | bibz2 | how can I unload the adaptive one? or better: which one should I unload |
| 17:53.15 | pabelanger | navaismo, perfect, that is the issue. We are not invoking that function |
| 17:53.17 | pabelanger | changing it now |
| 17:53.30 | pabelanger | anonymouz666, Well, we are using it. Mostly bridges an playback |
| 17:54.07 | WIMPy | doobeh: They don't only have issues with reducing codecs they also have issues with jitter. But as long as there's no IP involved, everything should wrok fine. |
| 17:56.13 | pabelanger | navaismo, thanks again. Owe you a beer |
| 17:56.37 | bibz2 | funny thing: the cdr_adaptive_odbc is empty |
| 17:57.35 | doobeh | WIMPy: thanks, I was confused when the compression started, so if I was just FXS#1 --> FXO#4 it doesn't bother, but if I FXS#1 --> WAN it'll kick in? |
| 17:58.12 | navaismo | no problem |
| 17:59.03 | WIMPy | doobeh: Exactely. |
| 18:01.29 | anonymouz666 | navaismo: I'll test some of your software specifically spy agents |
| 18:02.00 | navaismo | good, so far like nobody test that LOL |
| 18:02.21 | anonymouz666 | not even the author? heh... |
| 18:03.48 | navaismo | I do, but that doesnt count |
| 18:04.12 | navaismo | there are some videos about |
| 18:04.20 | navaismo | in my youtube account |
| 18:04.36 | anonymouz666 | yeah I am watch the plain asterisk |
| 18:04.38 | anonymouz666 | I don't use the elastix |
| 18:05.02 | doobeh | WIMPy: thanks :) |
| 18:07.16 | bibz2 | I don't get it why I get doubled CDR entries |
| 18:08.51 | anonymouz666 | navaismo: I understood the native video language!!! |
| 18:09.27 | *** join/#asterisk gusto (~gusto@2001:470:1f0b:a42:224:1dff:fecd:234c) |
| 18:10.07 | bibz2 | I'm also getting the warning, that the application delimiter is now the comma.. and the question if I did forget to convert my dialplan |
| 18:11.24 | navaismo | anonymouz666: super! |
| 18:15.40 | *** join/#asterisk mirela666 (~mirko.bra@95.180.116.173) |
| 18:21.48 | *** join/#asterisk usnavi (~DevWork2@static-100-32-93-142.lsanca.fios.verizon.net) |
| 18:22.44 | usnavi | Hi. Trying to send email messages from *. I set serveremail = vm@mydomain.com but postfix is showing the FROM line as asteriskuser@machinename.machinedomain.com |
| 18:22.56 | usnavi | Postfix doesn't like this and spits it out ofcourse. |
| 18:28.49 | *** join/#asterisk riess82 (~riessma@62-46-211-53.adsl.highway.telekom.at) |
| 18:29.37 | [TK]D-Fender | always used to read that as "Severe Mail" and though the tone was a bit harsh... |
| 18:38.10 | bibz2 | Currect auth, but based on stale nonce received from XXXX |
| 18:38.15 | bibz2 | what does this mean? |
| 18:38.35 | bibz2 | getting a 401 unauthorized answer |
| 18:42.33 | usnavi | So how does asterisk know to send "from" a certain address? |
| 18:43.55 | usnavi | does general's "serveremail" apply to [default] entries ? |
| 18:48.27 | *** join/#asterisk theron (~theron@66.220.145.150) |
| 18:49.25 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 18:57.26 | usnavi | So serveremail = vm@domain.com, got that set in [general] of /etc/asterisk/voicemail.conf .... what else do I need to do here? |
| 18:57.46 | *** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it) |
| 18:59.03 | usnavi | Aug 15 11:53:33 SNCA00-PH00 postfix/pickup[29173]: 196021422CA: uid=500 from=<asteriskpbx> |
| 18:59.03 | usnavi | <PROTECTED> |
| 18:59.21 | usnavi | Does the pickup mean I can't set the from address? |
| 19:00.11 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 19:15.05 | Katty | a dial tone the line may be quite continental, but sip channels are a girl's best friend! |
| 19:19.25 | file | Katty, here we are in the center of the first world, it's laid out before us, who are we to break down? |
| 19:19.33 | *** join/#asterisk ttyS3 (~o@gateway/tor-sasl/omlib) |
| 19:19.54 | *** part/#asterisk hexanol (~bibi@modemcable094.94-70-69.static.videotron.ca) |
| 19:20.32 | Katty | ^_^ |
| 19:27.21 | *** join/#asterisk jasonwert (~jasonwert@75-134-81-98.static.aldl.mi.charter.com) |
| 19:47.54 | *** join/#asterisk Eric-K (eric@188.226.212.8) |
| 19:52.38 | *** join/#asterisk reconwireless (uid10170@gateway/web/irccloud.com/x-qrdnzzclftdgvktp) |
| 19:54.24 | *** join/#asterisk Eric-K (eric@188.226.212.8) |
| 19:54.27 | *** join/#asterisk Hydrosine (hydrosine@188.226.212.8) |
| 19:57.01 | *** join/#asterisk mushroomed (~mushroome@backend.pe) |
| 19:58.13 | *** join/#asterisk Neptu (~Neptu@spa66.core-developers.com) |
| 20:06.02 | *** join/#asterisk tzafrir (~tzafrir@bzq-84-109-18-138.red.bezeqint.net) |
| 20:12.30 | *** join/#asterisk SpeedEvil (~quassel@tor/regular/SpeedEvil) |
| 20:16.07 | *** join/#asterisk bkruse (~Adium@64.89.97.113) |
| 20:20.43 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 20:25.06 | *** join/#asterisk dundel (~daniel@190.98.82.213) |
| 20:44.16 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 20:49.15 | *** join/#asterisk areski (~areski@80.174.128.86.dyn.user.ono.com) |
| 20:53.42 | *** join/#asterisk BlackDex (~BlackDex@ori.vyus.nl) |
| 21:14.47 | *** join/#asterisk Yon (~Yon@c80-217-139-243.bredband.comhem.se) |
| 21:18.25 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 21:28.34 | *** part/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 21:44.16 | *** join/#asterisk jasonwert (~jasonwert@75-134-81-98.static.aldl.mi.charter.com) |
| 21:45.48 | *** join/#asterisk usnavi2 (~DevWork2@static-100-32-93-142.lsanca.fios.verizon.net) |
| 21:46.21 | usnavi2 | I'm looking for a refresher but fairly complete tutorial/documentation on the asterisk dialplan syntax and idioms |
| 21:46.45 | usnavi2 | https://wiki.asterisk.org/wiki/display/AST/The+Asterisk+Dialplan would be workable? |
| 21:49.31 | *** join/#asterisk ipengineer (~zconkle@static-71-252-134-63.dllstx.fios.verizon.net) |
| 21:50.59 | *** join/#asterisk jasonwert (~jasonwert@75-134-81-98.static.aldl.mi.charter.com) |
| 21:51.51 | ipengineer | Is there an AMI action that will return active queue calls? QueueStatus does not show calls in progress for the given queue |
| 21:55.04 | *** join/#asterisk arosen (~arosen@geni/support/arosen) |
| 22:09.44 | *** join/#asterisk jasonwert (~jasonwert@75-134-81-98.static.aldl.mi.charter.com) |
| 22:23.14 | *** join/#asterisk FreezingCold (~FreezingC@135.0.41.14) |
| 22:23.20 | *** join/#asterisk jasonwert (~jasonwert@75-134-81-98.static.aldl.mi.charter.com) |
| 22:25.40 | *** join/#asterisk chaisangeen (~chaisange@84.255.134.58) |
| 22:35.37 | *** join/#asterisk jameswf (uid27319@gateway/web/irccloud.com/x-yyslyfbkwrcyqlez) |
| 22:37.27 | [TK]D-Fender | usnavi2: > |
| 22:37.31 | [TK]D-Fender | ~book |
| 22:37.31 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 22:37.35 | [TK]D-Fender | ~asteriskwiki |
| 22:37.35 | infobot | asteriskwiki is probably http://wiki.asterisk.org |
| 22:37.39 | [TK]D-Fender | ^^^ |
| 22:37.49 | usnavi2 | cool. |
| 23:01.47 | *** join/#asterisk reconwireless (uid10170@gateway/web/irccloud.com/x-vwpdtpsxfmnccwuh) |
| 23:06.29 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 23:15.41 | *** join/#asterisk Hyper_Eye (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 23:24.13 | *** join/#asterisk Hyper_Eye (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 23:37.59 | *** join/#asterisk tzafrir (~tzafrir@bzq-84-109-18-138.red.bezeqint.net) |