| 00:08.32 | *** join/#asterisk gusto (~gusto@2a02:810d:8600:8d4:21b:63ff:fe31:8426) |
| 00:10.38 | *** join/#asterisk X-Rob (sid14615@gateway/web/irccloud.com/x-vbwjftgkbamhetsh) |
| 00:15.26 | *** join/#asterisk snadge (~snadge@unaffiliated/snadge) |
| 00:19.00 | *** join/#asterisk serafie (~erin@24.96.64.240) |
| 00:52.07 | *** join/#asterisk supremekai (~a@78.29.156.110) |
| 00:52.11 | supremekai | Hello guys |
| 00:52.43 | supremekai | I want to make on extensions.conf this |
| 00:52.47 | supremekai | [demo] |
| 00:52.57 | supremekai | exten => s,1,Answer(500) |
| 00:53.15 | supremekai | same => Playback(tt-monkeys) |
| 00:53.37 | supremekai | And I want to redirect it to my 101 extension |
| 00:53.54 | supremekai | Using this:exten => 101,1,Goto(demo,s,1) |
| 00:54.01 | supremekai | where do I put this part? |
| 00:54.39 | *** join/#asterisk Akuma (~Akuma@137.175.233.59) |
| 00:57.42 | *** join/#asterisk Akuma (~Akuma@137.175.233.59) |
| 00:57.59 | *** join/#asterisk Vutral (~ss@mirbsd/special/Vutral) |
| 00:58.16 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.184) |
| 01:17.35 | *** join/#asterisk YannikS (1f12c1dd@gateway/web/freenode/ip.31.18.193.221) |
| 01:19.16 | YannikS | Hi, I am receiving an 405 Method not Allowed error when Asterisk contacts my siptrunk provider with a OPTIONS request: http://pastebin.com/sAJcjCit - Apparantly OPTIONS is not an allowed method for this provider. Can anyone tell me how to make asterisk register to the server without using the OPTIONS method? |
| 01:19.54 | pabelanger | YannikS: are you using qualify=yes? |
| 01:20.03 | YannikS | yeah |
| 01:20.06 | pabelanger | that is why |
| 01:20.16 | pabelanger | OPTIONS message are usually harmless |
| 01:20.51 | YannikS | okay, so setting it to qualify=no should fix the issue? |
| 01:21.10 | pabelanger | it should |
| 01:21.18 | YannikS | okay, will try |
| 01:21.20 | pabelanger | are you behind a NAT? |
| 01:21.30 | YannikS | yes |
| 01:21.42 | pabelanger | you might run into audio issues |
| 01:21.56 | pabelanger | OPTIONS also help keep the firewall / NAT happy |
| 01:22.07 | YannikS | I see |
| 01:22.15 | YannikS | well, having it register to the sip server is the first step :-) |
| 01:22.26 | YannikS | oh and btw, it works from a softphone inside the nat |
| 01:22.35 | YannikS | so i hope asterisk won't make any problems either |
| 01:23.23 | YannikS | now that is interesting |
| 01:23.52 | YannikS | Set qualify=no, no more messages appearing in the sip debug log. However, still unregistered :-( |
| 01:24.49 | pabelanger | OPTIONS wouldn't stop you from registering |
| 01:25.28 | YannikS | hmm |
| 01:26.05 | YannikS | *CLI> sip show registry Host dnsmgr Username Refresh State Reg.Time proxy.kabelphone.de:5060 N 405325xxxx@r 120 Unregistered |
| 01:26.10 | YannikS | :/ |
| 01:26.12 | pabelanger | have you setup: register => foo:bar@sip.example.org in sip.conf |
| 01:26.24 | *** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
| 01:26.24 | YannikS | yes |
| 01:26.42 | YannikS | this is my config: http://pastebin.com/b03URip9 |
| 01:26.53 | YannikS | (except for the foo bar thing) |
| 01:27.09 | pabelanger | well, do you have the foo bar thing? |
| 01:27.20 | pabelanger | because that setup will not register to the proxy |
| 01:27.33 | YannikS | register => {username}@reg172.kabelphone.de:{password}:{username}@proxy.kabelphone.de/{username} |
| 01:27.40 | YannikS | thats the register string i am using |
| 01:28.01 | pabelanger | where in your sip.conf, above this entry? |
| 01:28.19 | YannikS | well actually it got added by the freepbx gui |
| 01:28.36 | YannikS | inside an additional sip_registrations.conf file |
| 01:28.48 | pabelanger | okay, you'll need to confirm with #freepbx |
| 01:29.04 | pabelanger | but enable *CLI> sip set debug on |
| 01:29.12 | pabelanger | and look for the registration attempt |
| 01:29.19 | pabelanger | you should get back something |
| 01:29.23 | pabelanger | and a reason code |
| 01:29.31 | pabelanger | 401, 403, etc |
| 01:29.48 | YannikS | that's the thing thats bugging me |
| 01:29.55 | YannikS | I am not getting any messages at all |
| 01:30.06 | pabelanger | then it sounds like a configuration file issue |
| 01:30.21 | YannikS | well, before I got the OPTIONS messages |
| 01:30.30 | YannikS | is it possible to raise the loglevel? |
| 01:30.33 | pabelanger | I suspect freepbx doesn't like your registration string syntax |
| 01:30.52 | pabelanger | *CLI> core set verbose 5 |
| 01:31.00 | pabelanger | *CLI> core set debug 5 |
| 01:31.04 | YannikS | I am also getting messages for my extensions when not using set ip <ip> |
| 01:31.18 | YannikS | what is the default loglevel? |
| 01:31.21 | YannikS | just so I know for afterwards. |
| 01:31.35 | pabelanger | depends on how you launch asterisk |
| 01:31.41 | pabelanger | agian, #freepbx question |
| 01:31.59 | pabelanger | you can use cli.conf to setup commands to run everything asterisk starts |
| 01:32.42 | YannikS | okay |
| 01:37.58 | YannikS | Verbosity/Debuglevel 5 Log after sip reload: http://pastebin.com/7NrQf9c0 |
| 01:38.05 | YannikS | maybe this helps? |
| 01:43.56 | YannikS | Do you have any idea? |
| 01:52.28 | [TK]D-Fender | YannikS: [2014-02-08 02:34:00] DEBUG[2914]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting '0.0.0.0' into... |
| 01:52.41 | [TK]D-Fender | YannikS: I'm thinking you might have a DNS issue and the name isn't resolving |
| 01:52.48 | [TK]D-Fender | go ping it from your server CLI |
| 01:52.57 | [TK]D-Fender | and prove what you machine finds it at |
| 01:53.46 | YannikS | I can ping the proxy |
| 01:54.46 | [TK]D-Fender | Show us your actual register statement and that ping from CLI |
| 01:54.56 | [TK]D-Fender | masking ONLY the password |
| 01:56.31 | YannikS | register => 4053256570@reg172.kabelphone.de:<password>:4053256570@proxy.kabelphone.de/4053256570 |
| 01:56.45 | YannikS | I can resolve the dns request for proxy.kabelphone.de |
| 01:56.53 | YannikS | it won't answer icmp requests though |
| 01:57.58 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.164) |
| 01:58.07 | [TK]D-Fender | show it |
| 01:59.33 | [TK]D-Fender | it's als uncommon to need a separate authuser. normally it's just user:pass@host/contact |
| 02:00.04 | *** join/#asterisk YannikS_ (1f12c1dd@gateway/web/freenode/ip.31.18.193.221) |
| 02:00.20 | YannikS_ | sorry, my browser crashed... |
| 02:00.36 | YannikS_ | Did I miss anything? |
| 02:00.43 | [TK]D-Fender | [20:57][TK]D-Fendershow it |
| 02:00.45 | [TK]D-Fender | [20:59][TK]D-Fenderit's als uncommon to need a separate authuser. normally it's just user:pass@host/contact |
| 02:01.02 | YannikS_ | oh okay |
| 02:01.09 | YannikS_ | I will try user:pass@host/contact |
| 02:02.08 | YannikS_ | What should I use as host? |
| 02:02.10 | YannikS_ | the proxy? |
| 02:03.25 | [TK]D-Fender | leave those, kill the authuser. |
| 02:03.36 | [TK]D-Fender | then retest without CORE, and with SIP debugs |
| 02:04.23 | YannikS_ | can you elaborate how to change the register string? |
| 02:05.25 | *** join/#asterisk snadge (~snadge@unaffiliated/snadge) |
| 02:07.00 | YannikS_ | i don't quite understand how i am meant to change it |
| 02:09.11 | [TK]D-Fender | 4053256570@reg172.kabelphone.de:<password>:4053256570@proxy.kabelphone.de/4053256570 |
| 02:09.18 | [TK]D-Fender | 4053256570@reg172.kabelphone.de:<password>@proxy.kabelphone.de/4053256570 |
| 02:09.20 | [TK]D-Fender | ^ |
| 02:09.46 | YannikS_ | ah okay |
| 02:15.04 | YannikS_ | this looks good |
| 02:15.08 | YannikS_ | :-) |
| 02:15.08 | YannikS_ | seems to work |
| 02:15.09 | YannikS_ | thanks! |
| 02:15.24 | [TK]D-Fender | you're welcome |
| 02:16.00 | YannikS_ | sip show peers shows "unmonitored" for the siptrunk - why? |
| 02:16.17 | [TK]D-Fender | because you didn't tell it to "qualify" |
| 02:16.28 | YannikS_ | ah i see |
| 02:16.47 | [TK]D-Fender | * uses SIP OPTION to check for a response to consider them there or not |
| 02:17.05 | [TK]D-Fender | +S |
| 02:23.33 | *** join/#asterisk zerick (~eocrospom@190.187.21.53) |
| 02:29.25 | YannikS_ | Thanks again for the help, I am going to bed now :-) Cya later |
| 02:32.00 | *** join/#asterisk darkbasic (~quassel@niko.linuxsystems.it) |
| 02:58.35 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.196) |
| 03:05.44 | *** join/#asterisk MaliutaLap (~Finux@eth637.qld.adsl.internode.on.net) |
| 03:13.33 | *** join/#asterisk zelele (~zelele@177.133.255.158) |
| 03:13.37 | *** join/#asterisk chare (322f51ac@gateway/web/freenode/ip.50.47.81.172) |
| 03:13.46 | chare | any update on google voice? |
| 03:19.07 | [TK]D-Fender | no |
| 03:19.15 | [TK]D-Fender | Go subscribe to the mailing list |
| 03:22.14 | *** join/#asterisk X-Rob (sid14615@gateway/web/irccloud.com/x-fcckpvldzgcfjjpf) |
| 03:28.53 | *** part/#asterisk zelele (~zelele@177.133.255.158) |
| 03:31.03 | *** join/#asterisk Maliuta_ (~Finux@eth637.qld.adsl.internode.on.net) |
| 03:38.24 | *** join/#asterisk jasonwert (~w3rt@71.89.137.28) |
| 03:58.32 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.178) |
| 04:10.57 | *** join/#asterisk MaliutaLap (~nobusines@eth637.qld.adsl.internode.on.net) |
| 04:43.13 | *** join/#asterisk dijib (~dijib@d157-77-106.home4.cgocable.net) |
| 04:58.11 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.201) |
| 05:57.58 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.201) |
| 06:58.30 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.219) |
| 07:04.06 | *** join/#asterisk gerhard7 (~gerhard7@77-172-35-234.ip.telfort.nl) |
| 07:14.32 | *** join/#asterisk simNIX (~simNIX@156-60.bbned.dsl.internl.net) |
| 07:39.10 | *** join/#asterisk liquidamber (1817a92b@gateway/web/freenode/ip.24.23.169.43) |
| 07:45.19 | *** join/#asterisk jsjc (~Adium@217.Red-79-147-160.dynamicIP.rima-tde.net) |
| 07:58.01 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.194) |
| 08:03.33 | liquidamber | why shouldn't i do this? exten => numbers,n,swift("Your name is, ${CALLERID(name)}. Your number is, ${CALLERID(num)}") |
| 08:04.07 | liquidamber | is someone going to set their caller ID to some shellcode and root my box |
| 08:07.24 | *** join/#asterisk evil_gordita (~evilgordi@ip70-188-56-12.rn.hr.cox.net) |
| 08:58.32 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.252) |
| 09:09.10 | *** join/#asterisk Vutral (ss@mirbsd/special/Vutral) |
| 09:15.08 | *** join/#asterisk Vutral (ss@mirbsd/special/Vutral) |
| 09:43.28 | liquidamber | this is BS... asterisk won't play MP3s ? so confuse |
| 09:44.38 | *** join/#asterisk YannikS (1f12c1dd@gateway/web/freenode/ip.31.18.193.221) |
| 09:48.04 | *** join/#asterisk rfi (~rfi@cpe-76-95-3-143.socal.res.rr.com) |
| 09:51.52 | rfi | Is anyone awake at this hour? |
| 09:52.07 | YannikS | I am ;-) |
| 09:52.13 | rfi | Yay :-) |
| 09:52.29 | *** join/#asterisk MaliutaLap (~Finux@poltava.lusan.id.au) |
| 09:52.42 | rfi | I have a really random question I wanted to bounce off someone before I submitted an asterisk issue and sounded like a total n00b: |
| 09:53.08 | YannikS | Ask away, not sure if I can be of help though ;-) |
| 09:54.57 | rfi | I have a problem with PhonePower sending URI parameters in Contact field of the 200 OK response to a REGISTER. After looking at the source, and testing with another carrier that doesn't add URI parameters; it seems Asterisk will only respect a expires header and update the expiry timer if the Contact URI in the 200 OK exactly matches the one Asterisk sent in the REGISTER request. |
| 09:56.01 | rfi | A similar issue was discussed in https://issues.asterisk.org/jira/browse/ASTERISK-14870 and closed because the Contact field did not exactly match the one being sent. |
| 09:57.49 | *** join/#asterisk lok-jpr-india (lok-jpr-in@202.191.157.74) |
| 09:58.04 | rfi | I've been reading through the RFC's and I'm trying to figure out who is wrong; is Asterisk not following the RFC, is the RFC vague with respect to Contact binding (RFC3261 Section 10.3 Paragraph 8), or is the PhonePower's switch not following the RFC and the Contact header must exactly match. |
| 09:58.26 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.211) |
| 09:59.42 | rfi | I can edit the source and solve this issue for me... but I would like to know who is right and if it is something that should be added to the Asterisk branch or if I should chase the carrier's switch manufacturer for a resolution. |
| 10:00.14 | lok-jpr-india | Dear all, we are getting 486-BUSY here, though the softphones are not busy. we are using asterisk 11.7.0, SIP UA is QJSIMPLE, no NAT...any clue? thnx |
| 10:01.06 | rfi | Is this a new system, or did it just start happening? |
| 10:01.54 | rfi | (lok-jpr-india) |
| 10:04.34 | YannikS | rfi: I think I don't have the expertise to answer on that, sorry. Might be best if you opened a ticket or waited for someone else to respond in here |
| 10:08.20 | *** join/#asterisk MaliutaLap (~Finux@poltava.lusan.id.au) |
| 10:10.16 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.211) |
| 10:13.24 | lok-jpr-india | rfi: its a new setup |
| 10:14.54 | ChannelZ | we can only make random guesses without seeing a SIP debug of the failed attempt |
| 10:15.44 | ChannelZ | (and console verbose set to at least 3) |
| 10:15.48 | lok-jpr-india | it keeps saying everyone is busy congested, but can call IVR extension |
| 10:16.53 | ChannelZ | One possible is your devices aren't registered or asterisk otherwise doesn't know their IP |
| 10:16.55 | *** join/#asterisk vlad_starkov (~vlad_star@91.206.59.131) |
| 10:18.02 | lok-jpr-india | they are registed: sip show peers says registered |
| 10:18.38 | ChannelZ | well again I'm not going to sit here and guess 100 things. Show us verbose console output with sip debug |
| 10:22.45 | ChannelZ | ~pb |
| 10:22.45 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 10:25.26 | lok-jpr-india | plz have a look http://pastebin.ca/2635761 |
| 10:27.40 | ChannelZ | <--- SIP read from UDP:192.168.10.40:51889 ---> |
| 10:27.45 | ChannelZ | SIP/2.0 406 Not Acceptable |
| 10:28.03 | ChannelZ | Guess you need to figure out why the device that is SIP/306 is rejecting the INVITE |
| 10:29.52 | ChannelZ | maybe it doesn't like the massive list of codecs, I dunno. |
| 10:30.07 | lok-jpr-india | thats what we are unable to figure out, may be something related to TLS or srtp modules |
| 10:32.12 | *** join/#asterisk MaliutaLap (~Finux@poltava.lusan.id.au) |
| 10:37.12 | lok-jpr-india | any clue plz ? |
| 10:37.53 | rfi | 306 is a QJSIMPLE client? Can you pb the QJSIMPLE Debug Logs from 303 & 306? |
| 10:40.18 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 10:40.48 | rfi | Does QJSIMPLE support direct IP call? Can you call from 303 to 306 directly without using asterisk? |
| 10:43.41 | *** join/#asterisk Maliuta_ (~Finux@poltava.lusan.id.au) |
| 10:46.00 | lok-jpr-india | http://pastebin.ca/2635763 logs of qjsimple |
| 10:48.25 | lok-jpr-india | we do not know how to test direct sip call 303 to 306 without asterisk. what we have to set as registrar in qjsimple ? |
| 10:56.20 | rfi | I'm not sure I don't know QJSIMPLE. Is SRTP set to optional or mandatory on the QJSIMPLE clients? |
| 10:56.39 | lok-jpr-india | mandatory |
| 10:57.24 | rfi | try changing it to optional on both extensions you are calling between and see if the call proceeds |
| 10:58.18 | rfi | From your pb, this is probably a SRTP issue: 16:13:44.560 pjsua_call.c Error initializing media channel: Unknown error 220227 [status=220227] |
| 11:00.09 | rfi | If calls are successful with srtp optional then you might read through this: https://code.google.com/p/csipsimple/issues/detail?id=470 |
| 11:08.47 | *** join/#asterisk Milarepa (~Milarepa@host-74-211-92-125.beyondbb.com) |
| 11:14.00 | lok-jpr-india | we are unable to call from 303 to 306 with or without asterisk, tried optional, mandatory and disabled options of srtp :( |
| 11:26.22 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 11:37.28 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.127.208) |
| 11:44.49 | *** join/#asterisk fling (~fling@fsf/member/fling) |
| 11:47.52 | *** join/#asterisk CeBe (~CeBe@port-92-206-2-195.dynamic.qsc.de) |
| 11:52.09 | *** join/#asterisk danjenkins (~dan@cpc65687-folk2-2-0-cust207.1-2.cable.virginmedia.com) |
| 11:55.01 | *** join/#asterisk simNIX (~simNIX@156-60.bbned.dsl.internl.net) |
| 12:01.21 | *** join/#asterisk MaliutaLap (~Finux@poltava.lusan.id.au) |
| 12:02.53 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 12:08.22 | *** join/#asterisk CeBe (~CeBe@port-92-206-2-195.dynamic.qsc.de) |
| 12:16.33 | YannikS | I am trying to use a Yate softphone with asterisk. However, most times the registration does not work. ("Login failure, Reason: timeout" in yate). This is the asterisk-log of an unsuccessful registration attempt: http://pastebin.com/YFaCm4wa. This is the extension config: http://pastebin.com/DwJ1NL4s. Any ideas how to fix this? |
| 12:16.55 | *** join/#asterisk simNIX (~simNIX@156-60.bbned.dsl.internl.net) |
| 12:44.12 | *** join/#asterisk CeBe (~CeBe@port-92-206-2-195.dynamic.qsc.de) |
| 12:47.26 | *** join/#asterisk cyborg-one (~cyborg-on@212-178-14-91.broadband.tenet.odessa.ua) |
| 12:53.15 | *** join/#asterisk puzzled (~patrick@randombits.xs4all.nl) |
| 12:54.58 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.124.196) |
| 12:56.44 | *** join/#asterisk vlad_starkov (~vlad_star@212.34.52.66) |
| 13:30.41 | *** join/#asterisk mnathani (~mnathani@umbozero.winvive.com) |
| 13:43.07 | *** join/#asterisk uRDeSiRe (~yes@unaffiliated/urdesire) |
| 13:43.44 | uRDeSiRe | hello there! I am just wondering has anyone got outgoing skype working without paying the crazy price per channel directly from skype? |
| 13:45.24 | *** join/#asterisk DougsTech (~DougsTech@c-98-230-105-210.hsd1.al.comcast.net) |
| 13:52.37 | *** join/#asterisk fling (~fling@fsf/member/fling) |
| 13:57.41 | *** join/#asterisk YannikS (1f12c1dd@gateway/web/freenode/ip.31.18.193.221) |
| 13:58.23 | *** join/#asterisk gerhard7 (~gerhard7@77-172-35-234.ip.telfort.nl) |
| 13:58.42 | YannikS | Hi, when trying to register sip clients to my asterisk server, the registration always fails with a timeout error. Log here: http://pastebin.com/sqjAqa2T Can anyone help me to solve this? |
| 14:06.40 | Faustov | can you actually make dahdi compile with a 3.10 kernel? |
| 14:14.27 | Gugge | Faustov: i would think so, i have it working with 3.8 and 3.11 |
| 14:18.17 | Faustov | Gugge: sorry, I mean the 2.6 version, 2.8 works for me but wanpipe doesn't compile against it |
| 14:18.35 | Faustov | did you actually get 2.6.x? |
| 14:26.58 | Gugge | im pretty sure i uses the 2.9 version |
| 14:29.32 | Faustov | blast... |
| 14:49.07 | *** join/#asterisk Draecos (~Draecos@124-149-95-163.dyn.iinet.net.au) |
| 14:51.41 | YannikS | [14:58] <YannikS> Hi, when trying to register sip clients to my asterisk server, the registration always fails with a timeout error. Log here: http://pastebin.com/sqjAqa2T Can anyone help me to solve this? <- any ideas, anyone? |
| 14:53.53 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.27) |
| 15:01.57 | *** join/#asterisk wonderworld (~ww@ip-62-143-158-113.unitymediagroup.de) |
| 15:07.26 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 15:16.51 | YannikS | Hi D-Fender, got a few minutes for me? :-) |
| 15:17.27 | YannikS | [TK]D-Fender |
| 15:17.45 | [TK]D-Fender | just ask |
| 15:17.57 | YannikS | <YannikS> Hi, when trying to register sip clients to my asterisk server, the registration always fails with a timeout error. Log here: http://pastebin.com/sqjAqa2T Can anyone help me to solve this? |
| 15:18.24 | YannikS | it fails about 95% of the time |
| 15:18.35 | YannikS | sometimes it works, which seems really odd to me |
| 15:22.20 | YannikS | I have already tried turning nat on and off, binding the asterisk server to a specific ip, turning off firewalls on server and client |
| 15:22.27 | YannikS | To no avail, sadly :-( |
| 15:22.41 | Faustov | there's one retransmission, maybe you got too much packet loss? |
| 15:23.16 | YannikS | I don't think there should be much packet loss |
| 15:23.37 | YannikS | the asterisk server is wired to the network and i've got full reception on the clients |
| 15:24.01 | YannikS | Here is my sip.conf extension config: http://pastebin.com/SBMgJNkD |
| 15:26.08 | YannikS | any ideas? |
| 15:26.48 | *** join/#asterisk puzzled (~patrick@randombits.xs4all.nl) |
| 15:28.01 | YannikS | I tried CSIPSimple, PhonerLite, X-Lite and Yate as clients... always the same symptoms |
| 15:29.04 | *** join/#asterisk u0m3 (~u0m3@109.96.173.42) |
| 15:31.27 | [TK]D-Fender | That is the client running on? |
| 15:31.40 | YannikS | You mean the log? |
| 15:31.46 | YannikS | the log is from asterisk |
| 15:32.03 | [TK]D-Fender | no, the SOFTPHONE |
| 15:32.08 | [TK]D-Fender | what hardware, networked how? |
| 15:33.24 | YannikS | CSIPSimple on Android 4.2.2, X-Lite/Yate/PhonerLite on Windows 7 x64, gigabit network |
| 15:33.54 | YannikS | the asterisk server is connected by ethernet, the clients using 802.11n |
| 15:35.01 | WIMPy | 11n is not gigabit |
| 15:35.10 | YannikS | th ethernet network is fully gigabit |
| 15:35.52 | YannikS | connection speed of the wireless network is about 35mbit/s |
| 15:36.06 | YannikS | should be enough for a 64kbps connection.. |
| 15:40.15 | YannikS | [TK]D-Fender / Faustov, do you have any idea how I can fix this? |
| 15:44.58 | *** join/#asterisk happy-dude (~Adium@darwin-mbp2012-sxc.wireless.rit.edu) |
| 15:46.00 | uRDeSiRe | hello there! I am just wondering has anyone got outgoing skype working without paying the crazy price per channel directly from skype? |
| 15:47.41 | wonderworld | hi, I am trying to build asterisk12 with pjsip on debian. i followed instructions @ https://wiki.asterisk.org/wiki/display/AST/Installing+pjproject . everything seemed to work well, but in asteriks menuselect pjproject shows XXX because of umet dependencies. (Depends on: pjproject(E), res_sorcery_config(M)) what else could i do? |
| 15:48.16 | *** part/#asterisk happy-dude (~Adium@darwin-mbp2012-sxc.wireless.rit.edu) |
| 15:54.58 | *** join/#asterisk fling (~fling@fsf/member/fling) |
| 15:56.18 | wonderworld | never mind. forgot to install pkg-config. works now |
| 16:03.57 | YannikS | When trying to register sip clients to my asterisk server, the registration fails about 95% of the time with a timeout error. Asterisk log here: http://pastebin.com/sqjAqa2T, sip.conf: http://pastebin.com/SBMgJNkD, I have already tried turning nat on and off, binding the asterisk server to a specific ip, turning off firewalls on server and client, tried many different clients, but always the same symptoms |
| 16:04.06 | YannikS | any ideas how to fix this? |
| 16:19.08 | wonderworld | ping asterisk from one of the client machines and check packet loss |
| 16:19.17 | wonderworld | had a broken switch a while ago |
| 16:19.27 | wonderworld | same problem back then |
| 16:27.03 | *** join/#asterisk MaliutaLap (~Finux@eth637.qld.adsl.internode.on.net) |
| 16:29.12 | *** join/#asterisk MaliuatLap (~nobusines@eth637.qld.adsl.internode.on.net) |
| 16:32.30 | *** join/#asterisk serafie (~erin@24.96.64.240) |
| 16:34.51 | *** join/#asterisk fling (~fling@fsf/member/fling) |
| 16:37.19 | YannikS | wonderworld: how can i check the packet loss? |
| 16:37.30 | YannikS | wonderworld: pings do work both ways |
| 16:52.03 | wonderworld | ping shows the packet loss in it's sumary |
| 16:57.46 | *** join/#asterisk infernix (nix@unaffiliated/infernix) |
| 16:57.52 | [TK]D-Fender | YannikS: create a second peer to the phone witha FIXED IP and enable qualify and see if it responds |
| 17:33.28 | YannikS | [TK]D-Fender: "[2014-02-08 18:32:40] ERROR[3518]: chan_sip.c:16882 register_verify: Peer '12' is trying to register, but not configured as host=dynamic [2014-02-08 18:32:40] NOTICE[3518]: chan_sip.c:27952 handle_request_register: Registration from '<sip:12@192.168.178.26>' failed for '192.168.178.69:38195' - Peer is not supposed to register" - any advice? set host=192.168.178.69 |
| 17:33.40 | [TK]D-Fender | I said a second peer..... |
| 17:33.43 | [TK]D-Fender | not the SAME one |
| 17:34.07 | YannikS | the other one was extension 11 |
| 17:34.33 | [TK]D-Fender | Actually, yes, it might end up matching because of that... |
| 17:34.39 | [TK]D-Fender | look at the qualify packets |
| 17:34.42 | [TK]D-Fender | ignore registration |
| 17:35.07 | YannikS | sip show peers shows the correc thost but says unreacheable |
| 17:35.46 | [TK]D-Fender | ok, not at all what I said |
| 17:35.55 | [TK]D-Fender | Look at the PACKETS going out and look for an answer |
| 17:36.33 | YannikS | like with tcpdump or is there a asterisk command for that? |
| 17:38.33 | [TK]D-Fender | sip set debug on |
| 17:40.34 | *** join/#asterisk k1ng (~k1ng@unaffiliated/k1ng) |
| 17:42.46 | YannikS | ah okay now i understand what you mean |
| 17:42.56 | YannikS | i will try this later, got to go now |
| 17:43.01 | YannikS | thanks alot for your assistance |
| 17:45.01 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 17:45.02 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 17:55.23 | *** join/#asterisk gusto (~gusto@2a02:810d:8600:8d4:21b:63ff:fe31:8426) |
| 17:55.41 | gusto | hi folks |
| 17:56.06 | gusto | apparently there are ppl out there who are not using asterisk, but some cisco commercial product instead |
| 17:56.11 | gusto | are they crazy? |
| 17:56.14 | gusto | how can anyone NOT use asterisk? |
| 17:57.03 | WIMPy | Using Asterisk is less crazy? |
| 17:57.21 | gusto | yes |
| 18:14.36 | Kobaz | the interesting thing about those ciscos |
| 18:14.47 | Kobaz | is that they don't tend to randomly crash and deadlock as much as asterisk does :/ |
| 18:14.56 | Kobaz | but asterisk has gotten waaaaaay better |
| 18:15.05 | Kobaz | the new versions are pretty awesome, so yeah it's kind of crazy to not use it |
| 18:15.25 | WIMPy | And SCCP is much more suited to telephony than that SIP shit. |
| 18:16.06 | Kobaz | sccp is pretty cool in that it's all pretty much controlled at the server |
| 18:16.17 | Kobaz | even dialtone is sent off the server |
| 18:16.49 | WIMPy | Wich makes sense. |
| 18:16.58 | Kobaz | yeah true |
| 18:17.05 | gusto | w8 w8 |
| 18:17.11 | Kobaz | getting local dialtone on a polycom and then not being able to dial out, is kind of stupif |
| 18:17.13 | Kobaz | d |
| 18:17.16 | gusto | when does asterisk crash? did not happen to me yet |
| 18:17.23 | Kobaz | gusto: haha |
| 18:17.35 | WIMPy | gusto: Have you used it? |
| 18:17.54 | WIMPy | Ok, it hasn't crashed that often on me, but it has locked up many times. |
| 18:17.55 | Kobaz | gusto: i have a health check run every 60 seconds to make sure asterisk is alive and well |
| 18:18.02 | gusto | the only time asterisk did deadlock itself was when it was running in a loop resolving SRV record ... so i am disabeling that one from then on and resolving manually if needed |
| 18:18.07 | Kobaz | to check for deadlocks mostly |
| 18:19.41 | Kobaz | asterisk locks up on startup for me sometimes, which i have to watch |
| 18:19.50 | gusto | ??? |
| 18:19.54 | Kobaz | it's not actually asterisk's fault though, there's some lockup in the pg driver |
| 18:20.01 | gusto | what do you do with it? what version are you running? a testing one? |
| 18:20.03 | WIMPy | pg? |
| 18:20.07 | Kobaz | postgres |
| 18:20.13 | WIMPy | oh, yes |
| 18:20.18 | Kobaz | gusto: i do LOTS of custom stuff |
| 18:20.37 | gusto | so maybe it's your stuffs fault |
| 18:20.53 | Kobaz | gusto: any time i build a new feature or application on top of asterisk i will always hit, a) deadlocks, b) crashes, and c) crazy-ass edge cases |
| 18:20.56 | WIMPy | 11.7 locked up every time I tried a transfer. |
| 18:21.06 | gusto | and i can also confirm that the compatibility drivers for postgres are poor, also for mailservers and so on |
| 18:21.17 | gusto | mysql/mariadb works better |
| 18:21.25 | Kobaz | 1.8.0 crashed every time i made a call from one of my aastra phones |
| 18:21.40 | Kobaz | i fixed a null pointer problem in the sip header parsing and then that fixed it |
| 18:22.19 | Kobaz | i fixed two crash bugs and one ref leak bug in 1.8 before i could use it in production |
| 18:22.34 | gusto | i am using SPA112 and PAP2T |
| 18:22.52 | gusto | i have 2 of either of them |
| 18:23.13 | Kobaz | no |
| 18:23.20 | Kobaz | er, wrong chan |
| 18:23.42 | gusto | the only thing i am upset about is that these devices and also network printers do not support ipv5 |
| 18:23.45 | gusto | the only thing i am upset about is that these devices and also network printers do not support ipv6 |
| 18:23.54 | Kobaz | welcome to the world |
| 18:23.55 | gusto | so i have to use ipv4 in my lan for them to work |
| 18:24.22 | Kobaz | it'll be another 10-15 years before every consumer device supports native v6 |
| 18:24.36 | gusto | that i am not going to accept |
| 18:25.12 | Kobaz | you can always reverse nat or proxy |
| 18:25.20 | gusto | yes |
| 18:25.20 | Kobaz | have a v6 to v4 gateway |
| 18:25.25 | gusto | however |
| 18:25.32 | Kobaz | give all your v4 devices a routable v6 address |
| 18:25.37 | Kobaz | with proper firewalling of course |
| 18:25.44 | gusto | but 10 years? do you know how old i will be by then? |
| 18:25.50 | Kobaz | no, i don't |
| 18:25.54 | Kobaz | 10 years older than you are now? |
| 18:25.57 | gusto | yes |
| 18:26.00 | gusto | and that is a lot |
| 18:26.06 | gusto | i am already 25 |
| 18:26.36 | Kobaz | haha |
| 18:26.38 | Kobaz | youngin |
| 18:26.54 | Kobaz | don't complain |
| 18:27.14 | gusto | and you know what time it is? |
| 18:27.32 | gusto | 2014 and we still not have 100% ipv6 coverage! in 2014! |
| 18:27.49 | gusto | i have ipv6 only on my cable, but not over dsl |
| 18:28.46 | Kobaz | i dont have v6 at all |
| 18:28.48 | gusto | in 2014! and we havent been on the moon since the last time to check if everything is still there |
| 18:28.51 | Kobaz | don't particularly need it |
| 18:28.55 | Kobaz | but it would be cool |
| 18:29.01 | gusto | ipv6 is cool |
| 18:29.10 | Kobaz | i want a dead:beef:cafe:babe ip address |
| 18:29.18 | gusto | i am using HEnet where i do not have native IPv6 |
| 18:29.34 | WIMPy | Yes. It's cool to have all those issues bac that have been fixed on v4 many years ago :-) |
| 18:29.54 | gusto | i do not have any issues |
| 18:30.01 | gusto | quite the contrary, it works too well |
| 18:30.13 | WIMPy | Get some hackers to visit :-) |
| 18:30.18 | gusto | sometimes i have to intentionally break some stuff to stop it from working so well |
| 18:30.38 | WIMPy | wonders how something could possibly work too well. |
| 18:30.54 | WIMPy | You should see a doctor. |
| 18:31.22 | gusto | when you need to chflag your /etc/resolv.conf to nochg so that it does not overwrite your nameserver setting with an IPv6 one |
| 18:32.03 | gusto | or whatever that no-chage-flag setting is called |
| 18:32.11 | WIMPy | i |
| 18:32.41 | gusto | i can not see a doctor, because i am not paying my insurance right now |
| 18:33.00 | gusto | and ... i have no intent to, because doctors are also only worsening the situation |
| 18:35.16 | gusto | so |
| 18:36.15 | Kobaz | chattr +i /etc/resolv.conf |
| 18:36.31 | Kobaz | or you can edit your dhclient options and make it not monkey your nameservers |
| 18:36.45 | gusto | yes, i will ... sonn |
| 18:36.47 | gusto | soon |
| 18:37.01 | gusto | after i come back from the toilet |
| 18:37.05 | wonderworld | why use any newer product? there is ms netmeeting. rock solid. |
| 18:37.29 | WIMPy | And H.323 is better than SIP as well. |
| 18:38.06 | wonderworld | plus netmeeting has gui. no ugly console stuff. |
| 18:38.20 | WIMPy | that was my first standard voip experiment back then. From ohphone to netmeeting. |
| 18:38.30 | WIMPy | I have never seen it. |
| 18:38.44 | WIMPy | has never used Windows himself. |
| 18:38.48 | wonderworld | hehe |
| 18:39.06 | wonderworld | i switched desktops completly in 2002 or so |
| 18:41.18 | Kobaz | i got my ubnt edgerouter poe today |
| 18:41.22 | Kobaz | very excited |
| 18:46.22 | *** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood) |
| 18:59.54 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.3) |
| 19:03.34 | *** join/#asterisk gusto (~gusto@2a02:810d:8600:8d4:21b:63ff:fe31:8426) |
| 19:04.39 | WIMPy | I can't remember when I started to use X-windows regularly, but it was quite late. I still use RISC OS alongside, however. |
| 19:05.10 | WIMPy | Like for IRC ;) |
| 19:05.39 | gusto | risc os |
| 19:05.56 | gusto | that thing that runs on raspberrypi? |
| 19:06.05 | WIMPy | yes |
| 19:06.20 | WIMPy | But I'm still using it on a Risc PC. |
| 19:06.38 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.3) |
| 19:07.53 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.3) |
| 19:08.03 | gusto | what risc pc? |
| 19:14.34 | *** join/#asterisk SunTsu (miyamoto@unaffiliated/suntsu) |
| 19:14.36 | WIMPy | SA |
| 19:35.59 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.3) |
| 19:37.57 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.3) |
| 20:13.41 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-28-58.cablep.bezeqint.net) |
| 20:29.53 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.3) |
| 20:53.18 | *** join/#asterisk danjenkins (~dan@cpc65687-folk2-2-0-cust207.1-2.cable.virginmedia.com) |
| 21:07.08 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.201) |
| 21:08.27 | *** join/#asterisk brendan` (~textual@107-0-240-226-ip-static.hfc.comcastbusiness.net) |
| 21:26.03 | *** join/#asterisk calum_ (~calum_@cpc4-harg5-2-0-cust371.7-1.cable.virginm.net) |
| 21:30.38 | *** join/#asterisk Zogot (~Leon@26.sub224.ddfr.nl) |
| 21:30.41 | Zogot | Ahoyhoy |
| 21:31.18 | pabelanger | Hello, yes dog here |
| 21:32.33 | *** join/#asterisk ghost75 (~quassel@dslb-088-064-223-159.pools.arcor-ip.net) |
| 21:33.07 | ghost75 | its possible to brute force hack sip password in asterisk message log :< |
| 21:59.59 | *** join/#asterisk u0m3_ (~u0m3@92.80.106.27) |
| 22:05.49 | *** join/#asterisk serafie (~erin@24.96.64.240) |
| 22:10.09 | dwayne | I have 2 identical servers in softlayer with all the same packages and configuration files installed on them. One works great, the other Asterisk (1.8.25.0) crashes in the WELCOME_MESSAGE macro. Anyone ever see asterisk crash in WELCOME_MESSAGE? |
| 22:10.53 | WIMPy | What "WELCOME_MESSAGE"? |
| 22:12.16 | dwayne | ast_verbose("Asterisk %s, Copyright (C) 1999 - 2013 Digium, Inc. and others.\n" \ |
| 22:12.25 | dwayne | main/asterisk.c:3380 |
| 22:12.35 | pabelanger | ~backtrace |
| 22:12.35 | infobot | backtrace is a debugging tool that is useful when trying to track down asterisk crashes (use -g with asterisk to generate a core, then read doc/backtrace.txt). See also https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace |
| 22:12.42 | pabelanger | that is your best be to figure out why |
| 22:13.04 | WIMPy | Doesn't sound too easy to crash at that point. |
| 22:14.05 | dwayne | http://pastebin.ca/2636012 |
| 22:14.58 | WIMPy | Well, illegal istruction means either your build system or your (virtual) hardware are borked. |
| 22:15.21 | dwayne | yeah, I think its the VM |
| 22:15.31 | dwayne | or I guess the host |
| 22:15.53 | WIMPy | It can be the guest getting it wrong as well. |
| 22:28.30 | bsdice | 19:37 < WIMPy> And H.323 is better than SIP as well. <-- in what universe ?? |
| 22:29.10 | ghost75 | not in haxx0rs universe |
| 22:31.54 | *** join/#asterisk joako (~joako@opensuse/member/joak0) |
| 22:33.43 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.238) |
| 22:33.53 | *** join/#asterisk wonderworld (~ww@ip-62-143-158-113.unitymediagroup.de) |
| 23:10.40 | *** join/#asterisk Milarepa (~Milarepa@host-74-211-92-125.beyondbb.com) |
| 23:17.41 | *** join/#asterisk gusto (~gusto@2a02:810d:8600:8d4:21b:63ff:fe31:8426) |
| 23:19.11 | *** part/#asterisk Zogot (~Leon@26.sub224.ddfr.nl) |
| 23:19.25 | *** join/#asterisk slav3_kitten (~beepboop@unaffiliated/slav3-kitten/x-0866809) |
| 23:20.34 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.238) |