IRC log for #asterisk on 20150604

00:10.24*** join/#asterisk mjordan (~mjordan@75.76.55.191)
00:10.24*** mode/#asterisk [+o mjordan] by ChanServ
00:12.55*** part/#asterisk mjordan (~mjordan@75.76.55.191)
00:25.02*** join/#asterisk superscrat (~asanders@99-194-177-34.dyn.centurytel.net)
01:09.19*** join/#asterisk babak (uid19622@gateway/web/irccloud.com/x-rfpphobgdvctweeh)
01:11.19*** join/#asterisk jasonwert (~jasonwert@71.89.137.28)
01:18.21*** join/#asterisk Cyford33_i (junkmail@c-73-207-183-115.hsd1.ga.comcast.net)
01:23.13*** join/#asterisk K1rk (~Kirk@equinox.epecweb.com)
01:24.22*** join/#asterisk chandoo (~chandoo@ool-4a59659f.dyn.optonline.net)
01:25.54*** join/#asterisk azerus (~badass@unaffiliated/badass)
01:35.01*** join/#asterisk ipengineer (~zconkle@108-206-120-218.lightspeed.mckntx.sbcglobal.net)
01:42.53*** join/#asterisk mjordan (~mjordan@75.76.55.191)
01:42.53*** mode/#asterisk [+o mjordan] by ChanServ
01:44.23*** join/#asterisk bmurt (~brendan@64-121-3-32.c3-0.upd-ubr2.trpr-upd.pa.cable.rcn.com)
02:19.35*** join/#asterisk xiddus (~xiddus@ns328768.ip-37-187-115.eu)
02:34.54*** join/#asterisk infobot (ibot@rikers.org)
02:34.54*** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.3.2 (2015/04/08), 11.17.1 (2015/04/08), 1.8.32.3 (2015/04/08); Standard: 12.8.2 (2015/04/08); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu
02:38.55*** join/#asterisk superscrat (~asanders@99-194-177-34.dyn.centurytel.net)
02:46.53*** join/#asterisk bkruse (~Adium@24.42.207.11)
02:58.09*** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it)
03:36.08leftleghell all
03:36.13leftleghello all
03:41.27ChannelZhi
03:43.33*** join/#asterisk vader- (~Adium@pool-173-49-160-70.phlapa.fios.verizon.net)
03:55.08*** join/#asterisk infina (~infina@unaffiliated/infina)
03:57.26*** join/#asterisk azerus (~badass@unaffiliated/badass)
04:06.08*** join/#asterisk infina (~infina@unaffiliated/infina)
04:14.10leftleganyone use freepbx here?
04:15.49leftlegi'm looking for a robodialer for it.
04:16.18leftlegfor a school to notify parents when school is not in session.
04:18.19*** join/#asterisk joako (~joako@opensuse/member/joak0)
04:40.22*** join/#asterisk MaliutaLap (nikolai@unaffiliated/maliuta)
04:47.39LiuYanleftleg: for asterisk, i used to create a cron job, so it repeatly scan a database, and fetch information from it, and make a .call file, feed it to asterisk, asterisk will dial the number.
04:58.33*** join/#asterisk azerus (~badass@unaffiliated/badass)
04:59.45*** join/#asterisk joako (~joako@opensuse/member/joak0)
05:02.46danielbwwhy would asterisk 13 be inserting silence and talking spurts on a dahdi to sip call
05:07.38*** join/#asterisk gerhard7 (~gerhard7@77-172-82-111.ip.telfort.nl)
05:09.10*** join/#asterisk infina (~infina@unaffiliated/infina)
05:10.36*** join/#asterisk doome_ (~doome@82.150.48.146)
05:23.10*** join/#asterisk war9407 (war@static-72-73-18-14.clppva.fios.verizon.net)
05:23.32*** join/#asterisk funnymanva (~funnymanv@pool-70-106-251-155.clppva.fios.verizon.net)
05:59.22*** join/#asterisk azerus (~badass@unaffiliated/badass)
06:04.55*** join/#asterisk bulkorok (~Benjamin_@217.110.197.225)
06:59.55*** join/#asterisk azerus (~badass@unaffiliated/badass)
07:03.21*** join/#asterisk tzafrir (~tzafrir@local.xorcom.com)
07:03.49*** join/#asterisk Panther_Modern (~Panther_M@unaffiliated/panther-modern/x-6168176)
07:05.39*** join/#asterisk ChannelZ (~bobm@burner.com)
07:09.04*** join/#asterisk [d__d] (~d__d]@ec2-54-85-45-223.compute-1.amazonaws.com)
07:10.07*** join/#asterisk pchero_work (~pchero@109.70.54.56)
07:19.20*** join/#asterisk NoobSaibot (~NoobSaibo@cpe-65-31-251-48.new.res.rr.com)
07:20.33*** join/#asterisk areski (~areski@80.174.128.82.dyn.user.ono.com)
07:20.58NoobSaibotif 'show sip peers' is displaying a phone with an extension that is active with Status: OK, but the phone itself is reporting itself as being registered, where's the first place to look?
07:21.07NoobSaibot*Not being registered
07:21.33NoobSaibotgod it's late, i need to proofread better. sorry about that jumble of garbage.
07:22.31*** join/#asterisk sekil (~sekil@78.24.104.73)
07:45.02cookiezHello all, in comedian mail. Is there anyway to disable some menu-choices? I'm thinking mostly about disabling all other folder choices except new and old messages. Thank you in advance!
07:46.56*** join/#asterisk madax (~madax@c183-176.i02-6.onvol.net)
07:56.51NoobSaibotFigured out the problem - erroneous internal DNS entry pointing to  the wrong server. sip.ourdomain.com aimed at an esxi server's IP.
08:19.22*** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw)
08:20.30*** join/#asterisk doome_ (~doome@apn-185-10-125-76.vodafone.hu)
08:38.23*** join/#asterisk Guest9903 (~martin@185.32.9.250)
08:38.51Guest9903Hi, is it possible to store sip traffic from asterisk to a file instead of using sip set debug on which will splatter the CLI?
08:43.01*** part/#asterisk LiuYan (~hola@unaffiliated/liuyan)
08:52.24ChainsawGuest9903: You could do a wireshark capture and handle the traffic that way?
09:00.30*** join/#asterisk zizo_ (525a6a5d@gateway/web/freenode/ip.82.90.106.93)
09:04.06zizo_Hi all, sorry for my non-technical language. I have a fully working asterisk, which, among usual thigs, provide a callback service. If a certain number calls the server the call hangs and asterisk calls-back the number asking to digit a number to call. Is there a way for the first calling number to send dtmf tones to asterisk without answering?
09:06.38zizo_Or better, there is a way to read tones without an answer? Thanks
09:07.26skrustyi dont think so
09:07.29Kunsii don't think that's possible. maybe you want extension numbers instead?
09:10.10Guest9903Chainsaw: not a very nice solution indeed.
09:10.33ChainsawGuest9903: It stores the SIP traffic like you asked. It does not splatter the CLI.
09:10.50ChainsawGuest9903: You will have to qualify "nice", as it ticks all the boxes.
09:10.50Guest9903Can it be run as a service?
09:11.37ChainsawGuest9903: Wireshark? Sure. It can write to the file continuously, and your filters can be set up such that only the traffic you are interested in makes it to disk.
09:14.57Guest9903I think I'll go with tcpdump for this one though. All together I wished to avoid any external programs.
09:15.21*** part/#asterisk zizo_ (525a6a5d@gateway/web/freenode/ip.82.90.106.93)
09:17.14*** join/#asterisk Draecos (~Draecos@120.16.180.16)
09:24.11*** join/#asterisk D-Boy (~D-Boy@unaffiliated/cain)
09:31.21wdoekesGuest9903: you could drop the verbose messages from the cli altogether and only write to file (logger.conf), but that may be more silent than you wish for
09:43.57*** join/#asterisk Dovid (~dovid@ool-4356e96f.dyn.optonline.net)
09:56.25*** join/#asterisk CeBe (~CeBe@port-92-200-64-70.dynamic.qsc.de)
10:20.34*** join/#asterisk _0x5eb_ (~seb@seb-hpws2.w1.tele.crt1.net)
10:22.40*** join/#asterisk tzafrir (~tzafrir@local.xorcom.com)
10:26.51*** join/#asterisk areski (~areski@32.Red-83-47-150.dynamicIP.rima-tde.net)
10:38.51*** join/#asterisk matrix1233 (~matrix123@197.0.167.74)
10:45.48*** join/#asterisk Thorn (~Thorn@unaffiliated/thorn)
10:53.19*** join/#asterisk jkroon (~jkroon@dustpuppy.is.co.za)
10:55.10jkroonhi all, with ReceiveFAX, it used to be that FAXMODE was set.  I like to log what fax mode was used for transfer (it helps to track incorrectly configured SIP gateways that doesn't have t.38 enabled), however, it seems this variable hasn't been set for a while and I can't find the appropriate replacement in FAXOPT() ... has this been lost or is there some other mechanism?
10:56.43*** join/#asterisk doome_ (~doome@apn-185-10-125-76.vodafone.hu)
11:14.53*** join/#asterisk cusco (~tralala@2001:41d0:1:6caf::cafe)
11:15.03*** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw)
11:25.39*** join/#asterisk theHub (~chatzilla@ool-182e21ee.dyn.optonline.net)
11:44.14*** join/#asterisk wonderworld (~ww@ip-176-199-166-61.hsi06.unitymediagroup.de)
11:46.18*** join/#asterisk chandoo (~chandoo@ool-4a59659f.dyn.optonline.net)
11:57.27*** join/#asterisk robl^ (~robl@pdpc/supporter/active/robl)
11:58.59*** join/#asterisk mjordan (~mjordan@75.76.55.191)
11:58.59*** mode/#asterisk [+o mjordan] by ChanServ
11:59.36*** join/#asterisk fstd (~fstd@unaffiliated/fisted)
12:05.24*** join/#asterisk coppice (~chatzilla@123203240102.ctinets.com)
12:13.01*** join/#asterisk Hydrosine (~Hydrogone@185.47.1.1)
12:16.40*** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net)
12:27.40*** join/#asterisk vader- (~Adium@204.183.88.4)
12:47.48*** join/#asterisk bmurt (~brendan@8.39.115.8)
12:52.28*** join/#asterisk theHub (~chatzilla@199.246.139.2)
12:54.03*** join/#asterisk Draecos (~Draecos@58-7-95-88.dyn.iinet.net.au)
13:01.31vader-uhhg, i have to figure out why this office which has about (5) Polycom 650s and (4) Polycom 350s, are rebooting randomly now. They have been in place for 2-3 years. No updates. Running 4.0.2 code on them with FreePBX. Any ideas?
13:02.07*** join/#asterisk brad_mssw (~brad@66.129.88.50)
13:07.03*** join/#asterisk bkruse (~Adium@24.42.207.11)
13:14.49*** join/#asterisk mjordan (mjordan@nat/digium/x-mclygcetabdjsqrn)
13:14.49*** mode/#asterisk [+o mjordan] by ChanServ
13:18.10*** join/#asterisk jasonwert (~jasonwert@75-134-81-98.static.aldl.mi.charter.com)
13:28.28DivideBy0vader-: bad power?
13:39.14*** join/#asterisk gerhard7 (~gerhard7@77-172-82-111.ip.telfort.nl)
13:39.57*** join/#asterisk newtonr (RustyNewto@nat/digium/x-qobixjaajwdtwvvd)
13:39.58*** mode/#asterisk [+o newtonr] by ChanServ
13:48.34vader-they are PoE
13:53.55DivideBy0still sticking with power, is it a old POE switch?
14:00.07*** join/#asterisk sgriepentrog (sgriepentr@nat/digium/x-uxpnekmuwvbmxrqk)
14:03.07*** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it)
14:03.18fileneeds more power!
14:03.45Kunsiunlimited power!
14:06.58*** join/#asterisk kharwell (kharwell@nat/digium/x-ezrbfwgmvfztmqrm)
14:10.48vader-Netgear GSM7224P
14:11.36*** join/#asterisk doome_ (~doome@82.150.48.146)
14:18.10*** join/#asterisk azerus (~badass@unaffiliated/badass)
14:19.53*** join/#asterisk Dovid (~dovid@50.153.117.130)
14:20.43*** join/#asterisk italorossi (~Adium@187.60.66.11)
14:21.32*** part/#asterisk bulkorok (~Benjamin_@217.110.197.225)
14:21.47*** join/#asterisk ipengineer (~zconkle@static-71-252-134-63.dllstx.fios.verizon.net)
14:25.04*** join/#asterisk putnopvut (putnopvut@asterisk/master-of-queues/mmichelson)
14:25.04*** mode/#asterisk [+o putnopvut] by ChanServ
14:40.55*** join/#asterisk azerus (~badass@unaffiliated/badass)
14:45.26vader-it looks like it might be 1 phone causing the issue which is causing the switch to knock out a few PoE ports
14:45.45vader-I displayed the "culprit" phone's port and it seems to have leveled the rest of the ports off
15:03.20*** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it)
15:03.44*** join/#asterisk Dovid (~dovid@static-173-63-105-210.nwrknj.fios.verizon.net)
15:10.17*** join/#asterisk theHub (~chatzilla@199.246.139.2)
15:13.46*** join/#asterisk reveal (reveal@2401:7000:0:7::e376:fc6)
15:24.13*** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it)
15:25.04*** join/#asterisk bkruse (~Adium@64.89.97.113)
15:32.34*** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it)
15:34.48*** join/#asterisk rmudgett (rmudgett@nat/digium/x-tbxlvdsuihlthybo)
15:40.56*** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it)
15:48.01*** join/#asterisk xochilpili (~xochilpil@unaffiliated/xochilpili)
15:49.30*** join/#asterisk Sudravirodhin (~Sudraviro@185.94.31.100)
15:49.53xochilpilihi all, im having issues tryingg to provisioning an h323 avaya phone with asterisk. I have loaded the module, and the phone seems to seek to asterisk, but asterisk, is not displaying anything
15:50.04xochilpilidoes anyone made this?
16:03.27[TK]D-FenderWhat does "seem to seek" mean?
16:08.04xochilpilihey [TK]D-Fender :D
16:08.58xochilpilii meant, when i turn on the phone, search http to asterisk http server, and read the config files
16:09.07*** join/#asterisk robl^ (~robl@pdpc/supporter/active/robl)
16:09.29xochilpilibut in phone's display, just said : discover 10.0.1.55 (asterisk server)
16:10.22xochilpilithe real question is if i can connect an h323 phone to asterisk
16:11.17xochilpiliif it's possible to make it, researching on internet, everyone make a trunk between asterisk an avaya, im not trying to use avaya pbx.
16:31.01[TK]D-Fenderxochilpili: * can talk H323
16:31.10[TK]D-FenderBut that has NOTHING to do with it looking for HTTP for configs
16:31.15[TK]D-FenderDo not mix up topics like that
16:32.53xochilpiliwhat you mean? the phone is calling to HTTP to read his settings. (im not mixing this up with *), the question is about, how does ooh323 is communicated with the phone
16:38.05[TK]D-FenderH.323 is H.323
16:38.18[TK]D-FenderThe fact the phone wants configs is SEWCONDARY
16:38.28[TK]D-FenderPut it's configs wherever it is loking to get them from
16:38.36[TK]D-FenderThat has NOTHING to do with Asterisk
16:44.58xochilpili[TK]D-Fender, thanks again!
16:44.59xochilpili:D
16:45.04xochilpilibrb
17:53.30*** join/#asterisk robink_ (~quassel@unaffilated/robink)
17:54.20*** join/#asterisk theHub (~chatzilla@199.246.139.2)
17:55.46*** join/#asterisk TazzNZ (~TazzNZ@office-nat1.ytechops.co.nz)
17:59.33*** join/#asterisk kannan (~chatzilla@1.22.47.206)
18:06.27*** join/#asterisk chazzam (~chazz@donutokyo.info)
18:26.10*** join/#asterisk robink_ (~quassel@unaffilated/robink)
18:28.10*** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw)
18:33.47*** join/#asterisk [d__d] (~d__d]@ec2-54-85-45-223.compute-1.amazonaws.com)
18:37.21*** join/#asterisk ipengineer (~zconkle@static-71-252-134-63.dllstx.fios.verizon.net)
18:38.01*** join/#asterisk nanoha-sama (~nanoha-sa@van-app-svr.ad.v10networks.ca)
18:38.31*** join/#asterisk TriJetScud (~TriJetScu@van-app-svr.ad.v10networks.ca)
18:39.14*** join/#asterisk ch4plin (~uribes@187.149.55.6)
18:39.32ch4plinhi everyone
18:40.10ch4plinI have a question, is it possible an agent logged in to a queue to dial manually?
18:40.38[TK]D-Fender"agents" don't dial
18:40.41[TK]D-FenderDevices dial
18:41.06[TK]D-FenderAlso, what defines "manually" in this case?
18:42.56ch4plinI'm creating a web app with elastix, so I'm using the Call center module, my app works with ECCP protocol, but they dont have the dial command implemented yet, that's why I looking for alternative, to let the agents put a number and send it to dial
18:43.12ch4plinhope you can get what a means
18:43.33[TK]D-FenderWhat protocol?
18:44.16ch4plinit's a own elastix protocol, ECCP is named, works with XML languague
18:45.06[TK]D-FenderWell whatever that actually means and does is purely an Elastix convention, and is not supported here, along with their entire GUI
18:45.56ch4plinI know, just want to need if there was an option by using asterisk core to make a call with an agent
18:46.28ch4plinso I guess i goint to develop a service to dial using ami
18:46.39ch4plinthanks for your comments
18:47.13*** join/#asterisk areski (~areski@80.174.128.121.dyn.user.ono.com)
18:48.14[TK]D-Fender"Agent" isn't really a thing with Asterisk
18:48.22[TK]D-FenderThe terms are vague
18:48.46[TK]D-FenderAMI  (Originate) dials TO a device instead of you dialing FROM a device
18:49.11[TK]D-FenderSo in the end that's jsut a device ending up on a call .  Difference being which end well ... originates it
18:49.34[TK]D-FenderAnd if the device is capable of recieving a call.. it sure is capable of placing one itself
18:51.11ch4plinI was thinking to originate to a channel like SIP/{number}@trunk, then "transfer" to exten where a Queue, something like
18:51.36[TK]D-FenderHow does that relate to your concept of "agent"?
18:51.58[TK]D-FenderYou're calling "something" and when they answer are dumpign them INTO a queue... not as a member, but as a caller.
18:52.01WIMPyYou wan to call someone to put him in to a queue?
18:52.17WIMPyI don;t think I understand anthing.
18:54.47*** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood)
18:55.10ch4plinsomething like that, it just an idea, I'll try and if I succeed I'll let you know
18:55.44*** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it)
18:57.43*** join/#asterisk tzafrir (~tzafrir@bzq-179-40-172.cust.bezeqint.net)
19:00.27*** join/#asterisk hexanol (~bibi@modemcable094.94-70-69.static.videotron.ca)
19:00.59hexanolI'm doing some tests on asterisk 13 to eventually update to it from asterisk 11
19:01.13hexanoland, using chan_sip and when direct media is activated
19:01.35hexanolI've found that features code (for example, my feature code to hangup a call is *0) needs an additionnal # key
19:01.42hexanolto be accepted
19:01.44*** join/#asterisk babak (uid19622@gateway/web/irccloud.com/x-odwhejsxpdjlgufu)
19:02.08hexanoli.e. if I'm in a call with direct media enabled, I need to press *0# to end the call, while if direct media is disabled, I only need to press *0
19:02.27hexanolam I crazy or is this something that happened to other people ?
19:02.35[TK]D-FenderYou're crazy....
19:02.53hexanolalright, so could this be related to my configuration ?
19:02.58[TK]D-Fender* can't do direct media AT ALL if features are enabled
19:03.14hexanolhum that's not what I'm seeing here
19:03.20hexanolI have "rtp set debug on" activated
19:04.29hexanolDial("SIP/je5qtq-0000000a", "SIP/4wiilx,10,hHtT") in new stack
19:04.29hexanol...
19:04.29hexanol> Remotely bridged 'SIP/4wiilx-0000000b' and 'SIP/je5qtq-0000000a' - media will flow directly between them
19:06.27[TK]D-FenderHrm...
19:14.45mjordanuhm
19:14.58mjordanI'm not sure how you would be getting DTMF at all, unless it was via SIP INFO
19:15.11mjordanmuch less why it is remotely bridging with hHtT parameters added.
19:15.29hexanolI'm using SIP INFO, yes
19:16.12mjordanwell, then it is theoretically possible to do what you are doing
19:16.48hexanolit does work on asterisk 11
19:17.22mjordanI didn't say it didn't
19:17.35mjordanSIP INFO is just not what I'd typically recommend for DTMF, but okay
19:18.34hexanolwhat are the reason ?
19:18.56*** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw)
19:19.31mjordanit has no beginning or end, and duration has to be completely emulated
19:19.36mjordanthis can cause some rather strange effects
19:19.56mjordanit also doesn't interrupt the media on the sending side, but can interrupt the media on the receiving side when passed through, causing more weird effects
19:20.18*** join/#asterisk MaliutaLap (nikolai@unaffiliated/maliuta)
19:22.02*** join/#asterisk sparetire_ (~sparetire@unaffiliated/sparetire)
19:23.16hexanoldoing some more tests, and in fact I just need to press any DTMF after pressing *0 for the feature code to be considered, not # in particular
19:24.05hexanoldo you think it's worth opening a bug on the asterisk's bug tracker (if it's not already opened)?
19:25.56wanda_hiiii people
19:26.21wanda_I organised 2 BarCamp about voip open source :)
19:26.43wanda_participants want an other one with mummmmmble : http://barcamp.org/w/page/97007235/BarCampMumble-VoIP-OpenSource
19:27.38wanda_so, please. let us know your news or favorite tips about VoIP and we could discuss together: wednesday 17th june
19:28.26wanda_16H France, 10H north america coast , for the meeting, with mumble :) - Thank you ! (I'm here #xivo) :]
19:47.48DovidDoes anyone know wht would cause this? "res_musiconhold.c: getcwd() failed: No such file or directory". I cheked and the files are on the system and readable. we had an issue earlier with nfs but it was resolved. what's interesting is asterisk gives the same errors for files that are stored locally and files that are on the NFS server
19:49.03*** join/#asterisk darkbasic_ (~quassel@niko.linuxsystems.it)
19:50.59WIMPyhexanol: It's probably worth looking closely in what's actually happening. I got a similar issue with native bridging.
19:59.23*** join/#asterisk exuberocity (~exuberoci@cpe-174-101-43-157.columbus.res.rr.com)
20:00.50*** join/#asterisk polysics (~polysics@93-38-164-94.ip71.fastwebnet.it)
20:00.53polysicshello!
20:00.59leftleghey
20:01.09polysicsare there some tricky differences between Asterisk 11 and 13 media wise?
20:01.41polysicsbecause I am in a situation where an 11 box can dial a peer on the PBX fine and play music (or have media), while 13 with the exact same config can't
20:03.34Chainsawpolysics: If you are using the pjsip stack with 13, then yes, quite significant differences.
20:04.18polysicsI did not install this box myself, so I don't know. What is hte best way to check quickly?
20:05.22*** join/#asterisk pchero (~pchero@0x555140b5.adsl.cybercity.dk)
20:06.51polysicsfor what it's worth I do not think that is the case, though, since this was an upgrade of an existing build script
20:07.19*** join/#asterisk caseyd (~Casey.Dav@olivaw.mdteam.com)
20:08.46newtonrpolysics, 'module show like chan_sip.so' and 'module show like chan_pjsip.so'
20:09.03newtonralso, you can check for a configured sip.conf or pjsip.conf
20:09.12polysicsthey seem to both be there
20:09.28newtonrboth are loaded?
20:09.35polysicsyes
20:10.06newtonrthen you can check which config file has actual configuration vs a sample configuration/comments
20:10.10polysicsif I change something in sip.conf it does affect calls
20:10.15caseydI'm having some trouble, with I guess getting sip to work with my checkpoint firewall. It works if I plug the phone straight into the internet, or a more simple firewall. I'm not seeing anything dropped in the firewall logs. I've tried all sorts of different rules, I've even allowed all content and its still not working. I also tried setting the static nat to match the asterisk server. Does anyone have any ideas?
20:10.31polysicsnewtonr: pjsip.conf is definitely a sample
20:11.24newtonrpolysics, then it sounds like you are 'using' chan_sip. However you also have pjsip loaded and you probably don't want that if you are not using it.
20:11.26newtonrpolysics, https://wiki.asterisk.org/wiki/display/AST/Migrating+from+chan_sip+to+res_pjsip#Migratingfromchan_siptores_pjsip-Disablingres_pjsipandchan_pjsip
20:11.33*** part/#asterisk Kunsi (kunsmannf@unaffiliated/kunsi)
20:11.42newtonrThat talks about how to disable pjsip while you are still using chan_sip
20:11.57polysicsmight that be causing media issues? calling a peer on the same box does have media, whatever that means
20:12.07newtonrIf you don't understand how to configure them both and use them at the same time then you shouldn't have them both loaded and running
20:12.22newtonrand for 99.99% of people you will only want one or the other
20:12.40leftlegcaseyd it might be the app control on the firewall
20:13.08newtonrpolysics, possibly. I'd make sure the pjsip modules don't load to avoid any issues.
20:13.15leftlegi don't have that brand, but i've run into that problem where is firewall identifies voip traffic and woudl block if so enabled.
20:13.43polysicsok, the modules are gone
20:14.15polysicsmedia is still not there but w/e :)
20:14.27polysicsthe same exact sip.conf works with 11 so it must be some new option
20:15.04polysicsiptables has no rules... no idea on where to look now
20:15.26caseydleftleg, I wish.. we don't have that. We only have the basic firewall
20:16.06newtonrpolysics, how is your media issue presented to the user? Does a call between two phones not have any audio?
20:16.27polysicsoriginate to the same PBX peer from the CLI has music in one case, no music in the other
20:16.32polysicssame for a Dial()ed call
20:16.41polysicsin fact ,the call is not even acknowledged
20:16.47newtonrwhat is the difference between the two cases?
20:16.49polysicsthe call goes down after 5-6 seconds
20:16.59[TK]D-FenderWe don't know
20:17.05polysicsbasically, the Asterisk version
20:17.06[TK]D-FenderYou haven't shown us anything yet
20:17.31polysicsthey are in the same network, and using the same peer to call the PBX
20:17.36polysicswhat could I show you?
20:17.49newtonrpolysics, https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
20:18.12newtonryou can leave debug on 0
20:18.32newtonrthe important parts will be verbose and the "sip set debug on"
20:21.26polysicsgist is ok?
20:22.22newtonrsure
20:23.04polysicshere's what I got out of that
20:23.05polysicshttps://gist.github.com/polysics/5635b593b5909c2caecd
20:25.03newtonrpolysics, what were the parameters to originate?
20:25.35polysicsoriginate SIP/talkbox-peer/3997 application musiconhold
20:28.47newtonrthe 200OK and ACK are repeating. Perhaps the ACK is not reaching the far end?
20:30.07newtonryeah the peer even hangs up with "X-Asterisk-HangupCause: No user responding"
20:31.35newtonrpolysics, well, you have something to dig into there.
20:31.53polysicsat what level might the issue be? networking?
20:32.46newtonrcould be that or could be NAT related config options in sip.conf (on either end) that are misconfigured
20:33.10polysicsare there any differences between 11 and 13 on that side of config?
20:33.57newtonrProbably. You could search through the upgrade and changes files included with Asterisk
20:34.11polysicsyeah, I am reading through those
20:34.29newtonrhttps://wiki.asterisk.org/wiki/display/AST/Updating+or+Upgrading+Asterisk
20:35.09*** join/#asterisk jrose_atDigium (jrose_atDi@nat/digium/x-lubquiyudbpsctmi)
20:45.31*** join/#asterisk newtonr (RustyNewto@nat/digium/x-uzpojpmrsuddlbbw)
20:45.31*** mode/#asterisk [+o newtonr] by ChanServ
20:59.48*** join/#asterisk [TK]D-Fender (~joe@64.235.216.2)
21:20.28*** join/#asterisk wmchris (~Krim@ip-176-198-200-19.hsi05.unitymediagroup.de)
21:21.27wmchrishi, need some help with my asterisk installation. i can connect to my sip account, but when i add the register => command in the sip.conf, i can't connect anymore. dunno why.
21:22.51*** join/#asterisk jhlavacek (~jirka@84.19.95.180)
21:22.58wmchrisusing Asterisk 1.8.10.1 on a openwrt router.
21:23.18wmchrisreally need help :( trying around this since 2 days...
21:23.29*** join/#asterisk alexw (~textual@unaffiliated/alexw)
21:23.53wmchrisi don't even know how i can debug it, because there are no error messages in console or log file...
21:24.50wmchrisnoone online right now?
21:25.46*** join/#asterisk Panther_Modern (~Panther_M@unaffiliated/panther-modern/x-6168176)
21:32.24*** part/#asterisk jhlavacek (~jirka@84.19.95.180)
21:38.26newtonrwmchris, yo
21:38.39newtonrwmchris, you can turn on "sip set debug on" on the console to see the SIP traffic
21:38.57newtonrthen you can look to see if a REGISTER is going out
21:39.35wmchrisalready did... nothing.
21:39.56wmchristhere is really nothing in the console
21:40.31newtonrwhat do you see with "module show like chan_sip.so" ?
21:40.34wmchrisit's like the whole sip subsystem crashes when i add a register command
21:40.51wmchrismodule loaded.
21:40.51wmchrischan_sip.so                    Session Initiation Protocol (SIP)        0
21:41.03*** part/#asterisk mjordan (mjordan@nat/digium/x-mclygcetabdjsqrn)
21:41.07*** join/#asterisk alexw (~textual@unaffiliated/alexw)
21:41.16newtonrso chan_sip is loaded
21:41.25newtonryou could reload it and watch for console messages
21:41.36newtonryou need to make sure of some things first
21:41.42newtonr"logger show channels"
21:41.57newtonrsee what log message types are enabled for the console
21:42.12wmchrisafter module reload chan_sip the console stops working
21:42.13newtonrand remember that both verbose and debug have indepdent log levels
21:42.59wmchrisif i open a new one and redo the command i'll get OpenWrt*CLI> module reload chan_sip
21:42.59wmchrisThe previous reload command didn't finish yet
21:43.11*** join/#asterisk D-Boy (~D-Boy@unaffiliated/cain)
21:43.36wmchrisconsole has notice, error and warnings active. seems enough for me
21:43.57newtonrin a case like this you want verbose and debug, turn them both up to 5
21:44.03newtonr*in addition
21:44.13wmchrisdone
21:44.30newtonrthen try the reload again
21:44.34newtonrfrom your original state
21:44.38*** join/#asterisk bkruse (~Adium@64.89.97.100)
21:44.46wmchriswait a sec, core set debug 5 doesn't enable verbose and debug
21:44.54wmchrisso what is the command to add verbose and debug on console?
21:45.15newtonrsafest to do it from the config file logger.conf
21:45.38newtonrhttps://wiki.asterisk.org/wiki/display/AST/Logging+Configuration
21:45.41wmchrisdone
21:46.05newtonrbut you can do it from the console too
21:46.06newtonrhttps://wiki.asterisk.org/wiki/display/AST/Basic+Logging+Commands
21:46.08wmchrisand again.... after module reload chan_sip => asterisk crashes
21:46.18wmchrisor at least the command line interface stops working.
21:46.23wmchriswithout an error message
21:46.42wmchrisif i comment the register=> command out, everything works fine.
21:47.06newtonrcan you pastebin the register line?
21:47.44wmchrisit doesn't matter, every register line results in this behavior.
21:47.57wmchriseven a register => user@nothing
21:48.28newtonrah, very odd
21:50.30newtonrI'd say file a bug and post logs to it, but not only is that version really old within the 1.8 branch, but the 1.8 branch is only supported for security issues now.
21:50.54newtonrI'd recommend comparing at least the latest version of 1.8
21:52.27wmchrisi'm using an old router right now with openwrt.
21:52.51wmchrisso it's really complicated to compile things for
21:53.52newtonrGood luck! Not sure what else to try if absolutely any register line causes Asterisk to hang when you load chan_sip.so
21:54.04newtonrIt certainly shouldn't do that
21:55.35wmchrismaybe there is another solution for my problem.
21:56.24wmchrisi just need to proxy the sip from my home line to a VPN line. first i thought using siproxd - didn't work. :( asterisk doesn't work either...
21:56.31wmchrismaybe there is another solution?
21:58.29wmchrisah dammit, i'm installing a debian machine.
21:58.48wmchrisi tried using a router... i failed... ^^
21:59.03wmchristhank you very much for your help :)
21:59.08wmchris(no irony!)
21:59.27wmchrisat least now i'm sure it's not a fault of my config file. ^^
22:00.58*** join/#asterisk vader- (~Adium@2600:1002:b123:c26e:bdac:13ac:78f1:d41b)
22:01.53newtonrwmchris, can you pastebin your entire config file?
22:02.19newtonrwmchris, there is always a chance that the location of the register line, or a combination of configuration elements in the file have caused some strange issue
22:03.20*** join/#asterisk theHub (~chatzilla@ool-182e21ee.dyn.optonline.net)
22:04.03wmchrisi'm using this setup (http://www.ip-phone-forum.de/showthread.php?t=250556&s=9a0411e6e35b6e22f560a302dc87170b&p=1856801&viewfull=1#post1856801)
22:04.16wmchrisofc extensions.conf and sip.conf are inversed here ;)#
22:04.45*** join/#asterisk bkruse (~Adium@172.56.20.205)
22:04.56wmchrisbut i also tried the automatic configuration of the module luci-btx
22:06.02wmchrisand this config works fine on the debian machine i'm running right now :)
22:06.41wmchrisso problem not solved, but it's definitly a bug in asterisk 1.8 on ar71xx chipset
22:08.26newtonrdid you try disabling srvlookup?
22:10.39*** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw)
22:13.00wmchrisnope
22:13.04wmchriswhy should i disable this?
22:14.50newtonrsee https://issues.asterisk.org/jira/browse/ASTERISK-21378 and duplicates
22:16.07wmchrishm i'll try
22:16.35wmchrisbut i can do a lookup manually and i don't get a timeout even after 8hours....
22:16.41wmchrisso i doubt this is the bug :/
22:16.57wmchris(and tbh.. i just made my first call with the debian machine ;-))
22:17.13newtonri doubt it will affect it in this case either, but it is worth a shot
22:26.01*** join/#asterisk AlHafoudh (~AlHafoudh@echo.freevision.sk)
22:33.24*** join/#asterisk znf (~ibm86@unu.card-share.eu)
23:04.58*** join/#asterisk _danielbw (~danielw@ip70-181-229-202.sd.sd.cox.net)
23:06.28*** part/#asterisk ch4plin (~uribes@187.149.55.6)
23:08.20_danielbwhello
23:09.05*** join/#asterisk steelbrain (~steelbrai@119.159.178.233)
23:18.00*** join/#asterisk steelbrain (~steelbrai@119.159.178.233)
23:25.23*** join/#asterisk brc007 (~brc007@wsip-70-166-113-101.ph.ph.cox.net)
23:27.24*** join/#asterisk theron (~theron@173-20-126-202.client.mchsi.com)
23:37.35*** join/#asterisk bkruse (~Adium@24.42.207.11)
23:45.40*** join/#asterisk babak (uid19622@gateway/web/irccloud.com/x-ntmlwejptptguhrq)
23:52.11*** join/#asterisk chandoo (~chandoo@ool-4a59659f.dyn.optonline.net)
23:56.26*** part/#asterisk kharwell (kharwell@nat/digium/x-ezrbfwgmvfztmqrm)
23:59.36*** join/#asterisk fstd (~fstd@unaffiliated/fisted)

Generated by irclog2html.pl Modified by Tim Riker to work with infobot.