| 00:02.53 | *** join/#asterisk gusto (~gusto@2001:a60:11ff:1200::42:4) |
| 00:06.23 | *** join/#asterisk jsjc (~Adium@54.Red-83-35-54.dynamicIP.rima-tde.net) |
| 00:17.24 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 00:17.43 | SeRi | p3nguin: You around? |
| 00:17.51 | p3nguin | Si. |
| 00:18.13 | SeRi | I am having about 20% loss via udp mtr to voip.ms |
| 00:18.40 | SeRi | s/voip.ms/houston.voip.ms/ |
| 00:18.53 | SeRi | would changing to dallas help? |
| 00:18.56 | SeRi | I am thinking not |
| 00:19.11 | p3nguin | Run mtr against dallas and see what you get. |
| 00:19.11 | SeRi | I think my route would change just a tad not drastically to help |
| 00:19.22 | p3nguin | No sense in switching without testing first. |
| 00:19.36 | *** join/#asterisk b2 (~ion@pdpc/supporter/active/beckb) |
| 00:19.47 | SeRi | as I thought even wrost |
| 00:19.50 | SeRi | is comast shit man |
| 00:21.21 | *** join/#asterisk blee (~blee@72.188.117.219) |
| 00:21.41 | SeRi | damn going to dijib is even worst |
| 00:23.34 | SeRi | p3nguin: I am in your conf just testing |
| 00:24.15 | p3nguin | No problem. That's what it's for. |
| 00:24.28 | p3nguin | If I didn't want people in it, it would not be available. |
| 00:24.38 | SeRi | cool :) |
| 00:33.18 | dfgas-cr48 | <PROTECTED> |
| 00:33.48 | *** join/#asterisk wonderworld (~w@dsdf-4db55db8.pool.mediaWays.net) |
| 00:33.55 | *** join/#asterisk slav3_kitten (~kitten@unaffiliated/slav3-kitten/x-0866809) |
| 00:34.43 | ChrisInSydney | morning all |
| 00:38.06 | *** join/#asterisk vlad_starkov (~vlad_star@83.149.8.119) |
| 00:40.00 | SeRi | any body have an open echo test? |
| 00:40.24 | p3nguin | I can set up one or two for you. |
| 00:41.20 | SeRi | That would be awesome. |
| 00:41.21 | SeRi | Thanks |
| 00:41.34 | wonderworld | SeRi: http://voiceforall.com/soundsettings.php |
| 00:42.21 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 00:42.58 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 00:43.28 | p3nguin | <PROTECTED> |
| 00:43.45 | SeRi | nice |
| 00:43.50 | p3nguin | Last reload: 1 year, 5 weeks, 3 days, 5 hours, 32 minutes, 29 seconds |
| 00:44.03 | p3nguin | Asterisk has been up for a while. |
| 00:44.16 | SeRi | lol |
| 00:48.20 | p3nguin | Did you get that one? |
| 00:48.42 | SeRi | yeap |
| 00:48.44 | SeRi | trying now |
| 00:48.46 | p3nguin | Do you want one also on Comcast? |
| 00:49.05 | p3nguin | I can set up one on Comcast in Miami. |
| 00:59.14 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 00:59.21 | dijib | how do i compile chan_motif ? |
| 00:59.26 | dijib | its got XXX in menuselect |
| 01:00.24 | jpsharp | It is missing prerequisites to build. |
| 01:00.39 | WIMPy | It should say at the bottom what it's missing if you place the cursor on it. |
| 01:01.46 | dijib | res_xmpp where do i get this? |
| 01:01.52 | dijib | the others have been satisfied |
| 01:10.28 | WIMPy | res_ as in resource module. |
| 01:11.36 | dijib | XXX res_xmpp but i have dependency iksemel and openssl |
| 01:12.14 | dijib | done make clean && make distclean && ./configure && make menuselect |
| 01:13.53 | p3nguin | There's yer problem. |
| 01:14.56 | dijib | what? |
| 01:15.11 | dijib | i didnt specify 64bit ? |
| 01:17.30 | dijib | so p3nguin when i chan orig to tony's conference i get in, but when i use the same chan orig command with conf@p3.IP it dies |
| 01:19.57 | WIMPy | I hate ext4. |
| 01:20.00 | p3nguin | Last reload: 8 weeks, 3 days, 2 hours, 58 minutes, 39 seconds |
| 01:20.03 | p3nguin | 18352 calls processed |
| 01:20.19 | p3nguin | That one is quite a bit more active, even though it is on Crapcast. |
| 01:21.21 | dijib | so howcome their XXX'ed out? |
| 01:21.28 | dijib | ive got the dependencies |
| 01:23.45 | WIMPy | Did you install as packages and didn't install the -dev packages? |
| 01:24.21 | jpsharp | And once you installed the dependiencies did you rerun ./confgiure? |
| 01:25.52 | *** join/#asterisk jsjc (~Adium@54.Red-83-35-54.dynamicIP.rima-tde.net) |
| 01:26.14 | dijib | yes WIMPy |
| 01:26.21 | dijib | i did not install dev |
| 01:26.22 | dijib | lol |
| 01:27.07 | dijib | yes jpsharp |
| 01:27.16 | WIMPy | One of the usual examples how things that are meant to make things easier do the exact opposite. |
| 01:29.10 | dijib | re ./configure ing |
| 01:30.45 | dijib | whats openssl-dev -utils whats it called? |
| 01:30.55 | dijib | nvmd i know search |
| 01:31.21 | dijib | -devel |
| 01:31.42 | dijib | ok that still had them XXX'd |
| 01:32.25 | *** join/#asterisk junmin (~junmin@189.180.172.220) |
| 01:32.43 | SeRi | I fucking hate comast.... pleas Baby Jesus.... Hear me out for once and have another ISP come out to my area... Please. Please. Please. AMEN. |
| 01:32.56 | dijib | lol |
| 01:33.57 | dijib | anybody have any ideas? |
| 01:34.50 | slav3_kitten | SeRi, trade you comcast for my wisp |
| 01:35.05 | slav3_kitten | pay for public IP, get -1 nat layer |
| 01:35.09 | dijib | nvmd i found a missing dependency |
| 01:35.10 | slav3_kitten | double nat = win |
| 01:36.03 | *** join/#asterisk LiuYan (~LiuYan@211.154.128.171) |
| 01:38.57 | dijib | SWEET |
| 01:38.59 | dijib | compilin |
| 01:39.00 | dijib | g |
| 01:42.11 | dfgas-cr48 | dijib, hows the google voice going? |
| 01:42.48 | dijib | going well actually |
| 01:43.01 | dijib | it should all be configured just recompiling from source now |
| 01:43.21 | dijib | read that link i sent tells you everything you need to configure. |
| 01:44.06 | dijib | doing a make install as we speak |
| 01:44.09 | junmin | hello, with one voip provider, sometimes get NO AUDIO problem, how can i debug it? only have problem with one provider. any suggest? |
| 01:44.55 | dijib | modules loaded |
| 01:46.37 | dijib | interesting... who has a gtalk account i can test? |
| 01:46.49 | p3nguin | Get your own. |
| 01:47.41 | slav3_kitten | p3nguin, he can't last i checked they only gave them to US people an he's not in that county :D |
| 01:48.46 | p3nguin | I thought they gave them to US and Canada. |
| 01:49.48 | dijib | slav3_kitten: ive got one somewhere us have to find it |
| 01:50.00 | dijib | there once was a way |
| 01:50.10 | dijib | there still is that way |
| 01:50.28 | slav3_kitten | p3nguin, you may be correct. i thought it was US only |
| 01:50.55 | dijib | when i rant the extension for this it showed up in my gmail chat. i answered it. but the extension here kept ringing. |
| 01:51.26 | dijib | maybe try with wait is there still a google messenger? |
| 01:51.55 | dijib | nvmd |
| 01:51.59 | dijib | http://dl.google.com/googletalk/googletalk-setup.exe |
| 01:58.44 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 01:59.19 | *** join/#asterisk serafie (~erin@76.73.167.231) |
| 02:01.31 | SeRi | p3nguin: http://speedtest.phonoscope.com/myspeed/db/report?id=192875 |
| 02:02.31 | dijib | <PROTECTED> |
| 02:02.39 | dijib | ERROR |
| 02:03.23 | WIMPy | Too many IPs? |
| 02:06.58 | dfgas-cr48 | SeRi, did you try it yet? |
| 02:07.11 | SeRi | dfgas-cr48: no need. I found my issue. |
| 02:07.14 | SeRi | Thanks though! |
| 02:07.57 | dijib | "I like te-lephony and I cannot lie. You other vendors can't deny; When a call comes in with MOS so you can't hear and some echo in your ear you get angry!" - Sir Mix-a-Malcolm |
| 02:08.00 | dfgas-cr48 | np |
| 02:08.28 | dfgas-cr48 | dijib, what? |
| 02:08.38 | SeRi | http://pastebin.com/fb3fs2y4 |
| 02:08.54 | dijib | look at Seri's last post. |
| 02:09.02 | SeRi | dijib: stop doing bath salt |
| 02:09.20 | dijib | or that one lar'l |
| 02:10.07 | dijib | 1/20th |
| 02:11.00 | dfgas-cr48 | how could i do that? |
| 02:11.16 | dijib | i didnt write it. |
| 02:11.28 | SeRi | dfgas-cr48: do what? |
| 02:11.43 | dijib | http://blogs.digium.com/2012/10/31/asterisk-11-now-available/ |
| 02:12.46 | dijib | how could you do what? |
| 02:13.04 | dijib | hey dfgas-cr48 do you want google talk? |
| 02:13.19 | dijib | if you can handle the config i can handle the software |
| 02:14.42 | *** join/#asterisk Carlos_PHX_ (~Carlos@ip68-2-227-192.ph.ph.cox.net) |
| 02:17.03 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 02:17.04 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 02:33.07 | SeRi | p3nguin: you around? |
| 03:00.25 | p3nguin | It would be a lot easier if you'd just say whatever it is you want to tell me instead of first asking if I'm here, then making me answer that, then I have to wait for you to come back and type whatever it was that you wanted to tell me in the first place. |
| 03:00.45 | *** join/#asterisk vinhdizzo (~vinh@173-11-15-18-oregon.hfc.comcastbusiness.net) |
| 03:01.10 | dijib | lol |
| 03:01.21 | dijib | he's gone for the night i think |
| 03:04.09 | p3nguin | See what I mean? |
| 03:04.25 | p3nguin | If he would have just said whatever it was he wanted me to know, I'd already know it. |
| 03:04.42 | p3nguin | But now I had to respond to him that I was "here" before he'd tell me. |
| 03:04.52 | dijib | yeah you could have left an insightful reply to the issue although might be off topic from #asterisk |
| 03:05.18 | p3nguin | Asking if I am around isn't really an issue that needs to be responded to. |
| 03:05.52 | dijib | ontopic this: WARNING[4634]: res_xmpp.c:3096 xmpp_pak_presence: Received presence information about 'user@gmail.com' despite not having them in roster on client 'google' |
| 03:06.20 | dijib | he said a bunch of comcast customers are having the same issue right now |
| 03:06.26 | dijib | im thinking its skynet again |
| 03:06.30 | dijib | kidding |
| 03:06.47 | drmessano | What issue? |
| 03:07.13 | dijib | now i wish i had mixmon running |
| 03:07.19 | p3nguin | He's experiencing terrible jitter and packet loss, resulting in extremely poor audio on VoIP calls. |
| 03:07.25 | drmessano | AHHH |
| 03:07.27 | dijib | said some hub b or soething somewhere was acting up |
| 03:07.29 | drmessano | Yep, me too! |
| 03:07.38 | p3nguin | Comcast for you as well? |
| 03:07.42 | drmessano | Yep |
| 03:07.54 | drmessano | I had 60% packet loss between two of their routers |
| 03:08.29 | p3nguin | I'm sure it doesn't make him feel better knowing that he isn't alone, but it's good to know it isn't just his service. |
| 03:09.17 | dijib | http://speedtest.phonoscope.com/myspeed/db/report?id=192881 whats this Max Delay and forceIdle |
| 03:13.59 | dijib | should i take this to #networking? |
| 03:16.08 | dijib | SeRi: look up to 21:00 |
| 03:30.28 | dijib | what would this option do? same => n,Dial(SIP/300,20,D(:1)) |
| 03:31.26 | p3nguin | What does "core show application Dial" say it does? |
| 03:39.45 | dijib | ok ive found a known bug in 11.0.1 |
| 03:39.51 | dijib | regarding ICE |
| 03:40.26 | dijib | so how does one go about adding the revised lines to chan_motif.c |
| 03:40.28 | dijib | ? |
| 03:41.22 | p3nguin | If someone provided a patch, you'll have to apply the patch to the source file. |
| 03:42.15 | p3nguin | Do you have a patch for it? |
| 03:42.42 | drmessano | Its patched for RC1 |
| 03:42.49 | drmessano | 11.0.2-RC1 |
| 03:43.05 | p3nguin | You can either upgrade to that or you can simply patch what you have. |
| 03:43.56 | drmessano | Sorry, 11.1.0-RC1 |
| 03:44.12 | *** join/#asterisk vlad_starkov (~vlad_star@83.149.8.101) |
| 03:44.30 | dijib | so vi the chan_motif file? |
| 03:44.34 | p3nguin | no |
| 03:44.39 | dijib | ok |
| 03:44.40 | p3nguin | Get the patch file or upgrade. |
| 03:44.49 | dijib | crappy |
| 03:45.05 | dijib | or wait a week for retail |
| 03:49.56 | p3nguin | You could wait, or you could fix it sooner. |
| 03:51.40 | dijib | the way to go? http://svn.asterisk.org/svn/asterisk/branches/11/ |
| 03:52.15 | p3nguin | Is the patch not available? |
| 03:52.32 | dijib | well what else have they fixed that i dont know about? |
| 03:52.34 | dijib | ;) |
| 03:52.42 | p3nguin | Or broke! |
| 03:52.47 | dijib | lol |
| 03:53.26 | dijib | what do i do with the patch file? |
| 03:53.31 | p3nguin | apply it |
| 03:53.37 | p3nguin | Do you have it? |
| 03:53.56 | dijib | this? http://svnview.digium.com/svn/asterisk/branches/11/channels/chan_motif.c?view=patch&r1=375924&r2=375925&pathrev=375925 |
| 03:54.05 | dijib | from this http://svnview.digium.com/svn/asterisk/branches/11/channels/chan_motif.c?view=diff&r1=375924&r2=375925&pathrev=375925 |
| 03:55.10 | p3nguin | I don't know if that is for your version or not, but you can try to apply it if you want. |
| 03:55.34 | p3nguin | Download that file to something like chan_motif.c.patch |
| 03:56.15 | dijib | ok and? |
| 03:56.48 | dijib | ./chan_motif.c.patch in the 11/channels/ dir? |
| 03:57.02 | p3nguin | It isn't an executable file, so of course not. |
| 03:57.10 | dijib | ok then hows it work? |
| 03:57.25 | dijib | ive never patch a thing in my life |
| 03:58.22 | p3nguin | cd to the directory above channels. |
| 03:59.32 | p3nguin | I don't know the tree, so I can't say what that dir is. Maybe it's the root of the source, I don't know. |
| 04:01.20 | dijib | it is |
| 04:01.23 | p3nguin | Feel free to show me the entire path to your chan_motif.c file. |
| 04:02.07 | dijib | /usr/src/asterisk-11.0.1/channels/chan_motif.c |
| 04:02.22 | p3nguin | Okay, so you're in /usr/src/asterisk-11.0.1, then. |
| 04:02.34 | p3nguin | Is your patch file there or did you leave it somewhere else? |
| 04:02.52 | dijib | im actually havng issues downloading it |
| 04:03.51 | dijib | k its in the root. |
| 04:04.00 | dijib | of asterisk-11.0.1 |
| 04:04.24 | *** join/#asterisk moy (~moy@173.239.155.74) |
| 04:04.24 | p3nguin | patch -p0 < chan_motif.c.patch |
| 04:04.47 | p3nguin | If the patch is compatible, it should show that it was successful. |
| 04:07.15 | p3nguin | If it isn't, you should get a reject file. |
| 04:07.19 | dijib | can't find file to patch at input line 3 |
| 04:08.15 | p3nguin | Oh, the patch has longer paths in it.\ |
| 04:09.01 | *** join/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell) |
| 04:09.01 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 04:09.09 | p3nguin | try -p2 instead of -p0 |
| 04:09.10 | dijib | ok i will correct them |
| 04:09.20 | p3nguin | Just change the option in the command. |
| 04:10.06 | p3nguin | I forgot about their paths in that diff. My channel patches have a path of just channel/chan_whatever.c. |
| 04:10.16 | *** join/#asterisk dfgas-cr48 (~user@71-90-33-37.dhcp.ftbg.wi.charter.com) |
| 04:10.43 | dijib | patching file channels/chan_motif.c |
| 04:10.45 | dijib | done. |
| 04:11.10 | p3nguin | Now you have to make it again. |
| 04:11.14 | dijib | they had branch/11/ |
| 04:11.21 | dijib | channels/chan_motif.c |
| 04:11.32 | dijib | so the -p2 took two levels off? |
| 04:11.32 | p3nguin | I know. -p2 would remove the branch and the 11. |
| 04:11.35 | p3nguin | Yes. |
| 04:11.38 | dijib | sick |
| 04:12.34 | dijib | ok here is an idea. Sip password reset |
| 04:12.37 | dijib | ;) |
| 04:13.19 | dijib | so where do i recompile from? do i start over with make clean && make distclean |
| 04:13.33 | p3nguin | Don't make clean or distclean. Just make. |
| 04:13.38 | dijib | k |
| 04:15.15 | p3nguin | I always apply my patches before compiling the first time, since I build packages, so I'm not sure if it'll know you made a change to the file and recompile only that channel like if you had changed a setting in menuselect. |
| 04:15.32 | p3nguin | If it doesn't, it shouldn't be too hard to make only that one file. |
| 04:16.41 | dijib | looks like a different error |
| 04:16.42 | dijib | now |
| 04:18.22 | p3nguin | Is that progress? |
| 04:19.07 | dijib | one forward one back i would say |
| 04:19.30 | dijib | jingle_request: Unable to create Jingle channel on endpoint 'google |
| 04:20.42 | dijib | this was what i had |
| 04:20.45 | dijib | ERROR 21:02 < dijib> chan_motif.c:1971 jingle_interpret_ice_udp_transport: Incomplete ICE-UDP candidate received on session |
| 04:29.08 | dijib | https://issues.asterisk.org/jira/browse/ASTERISK-20101 |
| 04:29.21 | dijib | this is now my issue but it says Google talk capable: yes |
| 04:41.58 | *** join/#asterisk nicknam1232 (021d23fb@gateway/web/freenode/ip.2.29.35.251) |
| 04:46.01 | *** join/#asterisk mihamina (~mihamina@ip-41-190-237-66.orange.mg) |
| 04:46.16 | drmessano | pastebin your motif.conf and jabber.conf |
| 04:46.26 | drmessano | sorry xmpp.conf / motif.conf |
| 04:46.33 | dijib | with pleasure |
| 04:46.46 | p3nguin | Hopefully not too much pleasure. |
| 04:51.08 | dijib | http://pastebin.com/BNx5J3eB |
| 04:51.42 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 04:51.56 | drmessano | Ahh |
| 04:52.25 | dijib | you see something |
| 04:53.02 | drmessano | Yes.. there is no "default" context in chan_motif.. The [default] in the example is meant to be used as a template. Hang on.. let me edit |
| 04:53.33 | dijib | :D |
| 04:53.58 | p3nguin | Does it allow all those codecs, too? |
| 04:56.20 | drmessano | I made a few tweaks.. I also changed the layout a bit.. Look here and let me continue: |
| 04:56.22 | drmessano | http://pastebin.com/bNU8tLsp |
| 04:57.09 | drmessano | Set the transport and context per endpoint definition. There's no channel variables (yet), so all calls need to be dropped into a context with an s extension defined |
| 04:57.15 | drmessano | So if you added another user.. |
| 04:57.34 | drmessano | the context would need to be google2 (perhaps) |
| 04:57.59 | drmessano | I also think it's better to set the transport per channel as you could set another endpoint in motif.conf for gtalk calls |
| 04:58.52 | drmessano | Your calls werent working because nothing was being defined for that connection. All options are defined now |
| 05:02.58 | dijib | yeah now ive got a jabber hook error |
| 05:03.01 | dijib | flooding my cli |
| 05:03.15 | drmessano | What is the error |
| 05:08.11 | dijib | effin broken |
| 05:08.17 | dijib | keeps crashing my asterisk |
| 05:08.31 | drmessano | That tells me nothing |
| 05:09.30 | dijib | JABBER: socket read error |
| 05:09.37 | dijib | Parsing failure: Invalid XML |
| 05:09.42 | dijib | <PROTECTED> |
| 05:09.48 | dijib | those 3 over and over |
| 05:10.01 | dijib | <PROTECTED> |
| 05:10.06 | dijib | that before the parsing |
| 05:10.18 | dijib | this before the JABBER xmpp_client_thread: |
| 05:11.58 | drmessano | Is that with only starting asterisk or if you try to make a call? |
| 05:15.23 | dijib | damn unloading or reload chan_motif is crashing asterisk |
| 05:15.41 | dijib | thats starting asteris drmessano |
| 05:16.22 | drmessano | unloading and loading it is a bit iffy.. |
| 05:16.42 | drmessano | I noticed that as well |
| 05:17.15 | dijib | ok |
| 05:18.06 | *** join/#asterisk dfgas-cr48 (~user@71-90-33-37.dhcp.ftbg.wi.charter.com) |
| 05:18.38 | drmessano | the only thing I have different is the lack of buddy= and the debug= |
| 05:18.54 | dijib | something is wrong with xmpp |
| 05:19.17 | drmessano | Is the client showing connected? |
| 05:19.51 | drmessano | Make sure you have the openssl stuff |
| 05:20.19 | drmessano | I could get it to mostly work, but it would crash because there was no tls available |
| 05:20.50 | drmessano | res_xmpp with compile without it, just wont be happy with google |
| 05:23.03 | dijib | ok i rewrote the xmpp.conf file with what i had originaly and its connected again. no more errors |
| 05:23.21 | dijib | so this is jingle ? not gtalk i have configured? |
| 05:23.44 | dijib | i must have picked up some unicode character from pastebin |
| 05:25.08 | drmessano | Theres 3 protocols in play here.. |
| 05:26.32 | dijib | hey its calling out to google messenger now |
| 05:26.37 | dfgas-cr48 | dijib, yo |
| 05:26.39 | dijib | hey |
| 05:26.45 | dijib | sorry my server crashed |
| 05:26.49 | dfgas-cr48 | ahh, testing something |
| 05:26.50 | drmessano | Strict Jingle is "ice-udp", Google Jingle is their Jingle with a different media transport, which is what is used in the web client, and Google v1 is what Gvoice uses, along with the Windows Gtalk client, which uses an earlier jingle |
| 05:26.51 | dfgas-cr48 | ahh |
| 05:27.05 | dfgas-cr48 | you get that issue fixed? i found part of mine |
| 05:27.27 | dijib | still working on it |
| 05:27.34 | dijib | and i was able to ping you |
| 05:27.39 | dijib | ty drmessano |
| 05:27.45 | dfgas-cr48 | how bad was it? |
| 05:27.59 | drmessano | So if you want to use google voice and be able to call out to gtalk clients, you need 2 endpoints configured in motif.conf.. one for transport=google-v1 and one for transport=google |
| 05:28.33 | dfgas-cr48 | i just want google voice |
| 05:28.43 | dfgas-cr48 | idc about google talk |
| 05:29.04 | drmessano | Good for you |
| 05:29.09 | dfgas-cr48 | :D |
| 05:29.10 | drmessano | Wasnt addressing you |
| 05:29.11 | dfgas-cr48 | lol |
| 05:29.16 | dfgas-cr48 | i know that |
| 05:29.42 | dfgas-cr48 | dijib, what files do i need to edit, i am about to go back to sleep |
| 05:29.57 | dijib | edit to fix what issue? |
| 05:30.07 | drmessano | derp |
| 05:30.07 | dfgas-cr48 | ih for my google voice account |
| 05:30.17 | dfgas-cr48 | ih=oh |
| 05:32.24 | dijib | dfgas-cr48: dont worry about that today |
| 05:32.56 | dfgas-cr48 | ok |
| 05:32.58 | dfgas-cr48 | :D |
| 05:33.29 | dfgas-cr48 | part of my lag was my bluetooth headset |
| 05:33.35 | dfgas-cr48 | i had major lag in my voicemail |
| 05:37.08 | dijib | well then that was it |
| 05:41.39 | *** part/#asterisk nicknam1232 (021d23fb@gateway/web/freenode/ip.2.29.35.251) |
| 05:41.42 | dfgas-cr48 | well it was a big part |
| 05:41.46 | dfgas-cr48 | otherwise idk |
| 05:41.52 | dfgas-cr48 | how was my ping? |
| 05:42.02 | dfgas-cr48 | p3nguin, are you around? |
| 05:42.10 | dijib | no hs is asleep |
| 05:42.16 | dfgas-cr48 | k |
| 05:45.38 | dfgas-cr48 | dijib, in the dialplan how do i reach the auto attendant again? |
| 05:46.15 | *** join/#asterisk vlad_starkov (~vlad_star@83.149.8.124) |
| 05:51.20 | *** join/#asterisk Dennisvj (~dennis@unaffiliated/dennisvj) |
| 05:53.57 | *** join/#asterisk Dennisvj (~dennis@unaffiliated/dennisvj) |
| 05:59.29 | dijib | dfgas-cr48: what? |
| 05:59.50 | dijib | you goto your main did's extension |
| 06:00.16 | dfgas-cr48 | if i want something to return to the the auto attendant instead of hanging up |
| 06:01.56 | dijib | use the goto application and put them back to your main DID |
| 06:06.12 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-sisdxqhkdkqbuujy) |
| 06:09.51 | *** join/#asterisk vlad_starkov (~vlad_star@83.149.8.124) |
| 06:10.40 | dfgas-cr48 | thank you |
| 06:10.48 | dfgas-cr48 | back to bed i go |
| 06:10.53 | dfgas-cr48 | night |
| 06:12.43 | dijib | k |
| 06:32.03 | *** join/#asterisk fritz09 (~Adium@pop1-224.catv.wtnet.de) |
| 06:46.05 | *** join/#asterisk bmg505 (~leon@196.209.120.151) |
| 06:47.08 | *** join/#asterisk ThomasLocke (~ThomasLoc@pdpc/supporter/active/thomaslocke) |
| 06:53.42 | *** join/#asterisk oej (~olle@2001:16d8:cc57:1000::42:1003) |
| 06:55.16 | slav3_kitten | dijib, you around? |
| 06:55.35 | *** join/#asterisk black (~blackcat@pdpc/supporter/active/blackcat) |
| 06:56.42 | *** part/#asterisk black (~blackcat@pdpc/supporter/active/blackcat) |
| 07:09.46 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 07:29.02 | *** join/#asterisk TimeRider (~steve@188.220.34.144) |
| 07:31.59 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 07:33.36 | *** join/#asterisk Chandrakant (~chandraka@115.252.76.195) |
| 07:33.53 | *** join/#asterisk v0lZy (~Thunderbi@mail.silk-group.net) |
| 07:33.55 | Chandrakant | hello all |
| 07:34.13 | slav3_kitten | word up Chandrakant |
| 07:34.27 | *** join/#asterisk mokmeister (~mokmeiste@109.78.112.113) |
| 07:34.44 | Chandrakant | i am using asterisk 1.8.13.0 with tls |
| 07:34.57 | Chandrakant | my request forwarded from kamailio to asterisk on tls |
| 07:35.06 | Chandrakant | on both tls certificate is same |
| 07:35.16 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 07:35.18 | Chandrakant | [Nov 8 21:57:34] ERROR[16357]: tcptls.c:89 ssl_close: SSL_shutdown() failed: 5 |
| 07:35.19 | Chandrakant | [Nov 8 21:57:36] ERROR[16001]: tcptls.c:89 ssl_close: SSL_shutdown() failed: 5 |
| 07:35.19 | Chandrakant | [Nov 8 21:57:37] == Problem setting up ssl connection: error:00000000:lib(0):func(0):reason(0) |
| 07:35.27 | Chandrakant | but found above error |
| 07:35.38 | Chandrakant | i was already used below patches, but no luck |
| 07:35.43 | Chandrakant | https://issues.asterisk.org/jira/browse/ASTERISK-18345 |
| 07:35.43 | Chandrakant | https://issues.asterisk.org/jira/browse/ASTERISK-20559 |
| 07:36.09 | Chandrakant | and even posted in asterisk-users group but no replies... any idea about guys |
| 07:36.48 | WIMPy | What about using a current Asterisk version? |
| 07:37.19 | dijib | slav3_kitten: am now |
| 07:37.30 | dijib | for a couple min |
| 07:37.46 | slav3_kitten | dijib, pm |
| 07:38.20 | *** join/#asterisk eject_ck (~Evgeniy@213.159.242.65) |
| 07:38.48 | Chandrakant | its production server, so don't have rights to change |
| 07:39.07 | Chandrakant | and one more thing amr codec is working on it.. |
| 07:40.08 | WIMPy | I don't see how using patches is less of "changing" than upgrading. |
| 07:40.31 | WIMPy | The amr might be an issue, off course. Where did you find that? |
| 07:41.35 | fling | Which codec do I need if I want a fine quality? |
| 07:42.09 | WIMPy | G.722 or better |
| 07:42.49 | ChannelZ | A string with Progresso cans... not that cheap Campbells crap. |
| 07:43.10 | fling | WIMPy: what about silk? |
| 07:43.33 | WIMPy | That fits the "or better" part. |
| 07:43.57 | fling | ok |
| 07:44.00 | ChannelZ | I think there's some limitations to silk abot transcoding |
| 07:44.02 | WIMPy | Wait. Is it the one I think of? |
| 07:44.12 | ChannelZ | or maybe I'm thinking of something else.. maybe that was CELP |
| 07:44.19 | fling | tell us what you think |
| 07:44.36 | ChannelZ | I think SILK is disappearing anyway in favor of Opus but I don't know if any work has actually gone into that |
| 07:44.57 | WIMPy | I don't think so. |
| 07:46.14 | WIMPy | Too much reinventing the wheel because of patents and copyright. |
| 07:46.59 | fling | http://lists.digium.com/pipermail/asterisk-dev/2012-August/056668.html |
| 07:47.05 | ChannelZ | I though opus was open.. |
| 07:47.16 | fling | but it is |
| 07:49.08 | ChannelZ | well anyway it was CELP I was thinking of that there is no transcoder for, but SILK works (in Asterisk 10+ that is) |
| 07:49.26 | WIMPy | Looks like it could be one of the modules every user has to get on his own because of licensing. |
| 07:49.34 | ChannelZ | Ironically SFA doesn't support SILK :) (course it's dead anyway) |
| 07:51.02 | fling | ChannelZ: do I need SILK? |
| 07:51.36 | ChannelZ | not particularly |
| 07:52.21 | ChannelZ | I don't know of any devices that support it, just softphones. Something to play with. |
| 07:52.30 | fling | ChannelZ: I'm making a skype gateway so I want to have a similar sound quality |
| 07:52.48 | ChannelZ | For wideband on handsets you'll want to look into g722 |
| 07:52.56 | fling | ok, thanks |
| 07:53.25 | ChannelZ | or they might possibly support SPEEX |
| 07:53.35 | fling | and speex is better? |
| 07:53.43 | ChannelZ | better than what? |
| 07:53.50 | fling | than g722 |
| 07:54.02 | WIMPy | Many codecs do different qualities. |
| 07:54.26 | WIMPy | I haven't seen speex in hardware phones so far, either. |
| 07:54.32 | unicron | i need dtmf inband so boo |
| 07:54.51 | unicron | i hate it because it sometimes eats digits |
| 07:54.54 | WIMPy | And if I remember right it can be both worse and better than G.722. |
| 07:55.02 | fling | ok |
| 07:55.17 | ChannelZ | What is this a gateway between? |
| 07:56.16 | fling | ChannelZ: mostly softphones <-> skype |
| 07:56.39 | ChannelZ | using SIP for Skype or something? |
| 07:56.52 | fling | umm? |
| 07:57.11 | ChannelZ | How are you planning to talk to Skype? |
| 07:57.26 | fling | I have an asterisk pbx and I want to call skype users sometimes |
| 07:57.42 | fling | I have skype on server and freeswitch |
| 07:58.28 | fling | ChannelZ: so asterisk will connect to freeswitch and freeswitch will call some skype user using skype client started on server via skype api and virtual oss |
| 07:59.03 | *** join/#asterisk bulkorok (~bulkorok@85.183.36.36) |
| 07:59.07 | fling | I want it to work a simplier way but it is what I have installed now |
| 07:59.28 | ChannelZ | hmm |
| 07:59.36 | fling | also I have asterisk 1.8.15.1. Do I need to update to 10 or 11? |
| 08:03.52 | ChannelZ | probably not |
| 08:04.12 | fling | why not? is not it stable yet? |
| 08:04.33 | ChannelZ | I don't know much about that Skype API method, I'm surprised it still works, that M$ didn't take it out of the client |
| 08:04.38 | wdoekes | fling: don't do 10, it'll be EOL next month |
| 08:04.55 | wdoekes | if you can, try 11, but be prepared to report bugs |
| 08:05.12 | fling | wdoekes: I'm setting up the new server so I'll try 11 |
| 08:05.30 | fling | wdoekes: it will only work with skype farm if I'll set it up :p |
| 08:06.09 | wdoekes | I'm not sure there is a working skype though |
| 08:06.28 | fling | wdoekes: working skype? umm |
| 08:07.11 | ChannelZ | he might be talking about Skype For Asterisk, which is dead |
| 08:07.12 | fling | wdoekes: I'm doing it similar to this > http://www.personal.psu.edu/wcs131/blogs/psuvoip/2011/12/skype_for_asterisk_the_hard_way.html |
| 08:07.24 | fling | ChannelZ: oh, I know… |
| 08:07.27 | *** join/#asterisk felimwhiteley (~quassel@089-101-203026.ntlworld.ie) |
| 08:10.14 | wdoekes | horrible |
| 08:10.33 | fling | wdoekes: is there a simplier way? |
| 08:10.48 | wdoekes | not that I'm aware of =) |
| 08:11.09 | fling | I know I should use skype clients anyway but I want to get rid of freeswitch |
| 08:13.23 | fling | Which protocol do I need to use between two asterisks? iax2? |
| 08:14.32 | wdoekes | many people use sip only |
| 08:14.36 | WIMPy | You can use any protocol, but IAx was made for that purpose. |
| 08:14.58 | fling | wdoekes: why only sip? I dislike sip |
| 08:15.21 | wdoekes | yes.. but if your clients all use sip, you don't need to hassle with a second protocol |
| 08:16.10 | fling | I can use iax2 softphone |
| 08:18.07 | kaldemar | fling: you need to use a protocol that asterisk speaks. :P SIP and IAX2 are the most commonly used. |
| 08:18.20 | fling | kaldemar: ok :] |
| 08:19.33 | slav3_kitten | god i hate asterisk at this minute |
| 08:19.42 | fling | slav3_kitten: why why? |
| 08:20.00 | slav3_kitten | PROBLEMS |
| 08:20.04 | slav3_kitten | gah caps lock |
| 08:20.08 | slav3_kitten | brb getting a drink |
| 08:22.38 | Chandrakant | yup, sorry was in meeting |
| 08:22.54 | Chandrakant | any idea about tcp tls error, pasted above |
| 08:23.22 | slav3_kitten | fling, i'm having problems getting inbound from remote phones working |
| 08:25.20 | *** join/#asterisk oej (~olle@2001:16d8:cc57:1000::42:1004) |
| 08:25.52 | fling | slav3_kitten: NAT? |
| 08:26.18 | kaldemar | slav3_kitten: there's probably no reason to hate asterisk for that. on the contrary. |
| 08:26.22 | slav3_kitten | fling, don't ya know it |
| 08:26.34 | fling | know what? |
| 08:26.47 | *** join/#asterisk k610 (~Instantbi@cred.epid.ucl.ac.be) |
| 08:26.58 | slav3_kitten | NAT |
| 08:27.00 | *** join/#asterisk bombev (~bombev@PPPoE-Static-40-132.UnicsBG.Net) |
| 08:27.42 | slav3_kitten | fling, to be specific it's double NAT |
| 08:28.00 | slav3_kitten | i'm 98% sure my cisco router is setup properly to forward the ports |
| 08:28.04 | bombev | hi how to clean my asterisk log via ssh terminal |
| 08:28.08 | fling | slav3_kitten: so possible this is why you have your problems |
| 08:28.24 | bombev | i am on /var/log/asterisk |
| 08:28.28 | fling | slav3_kitten: may not you try to use routes instead of nat? |
| 08:28.50 | fling | bombev: clear the file? |
| 08:28.55 | slav3_kitten | fling, i lack control over one of the nat layers |
| 08:28.59 | bombev | and there is full full1 full2 full3 |
| 08:29.06 | bombev | should i delete this file |
| 08:29.08 | bombev | or what |
| 08:29.47 | fling | bombev: you may delete old log and create new one (if you do not need old) |
| 08:29.55 | fling | bombev: you may want to use logrotate |
| 08:30.15 | fling | slav3_kitten: most problems I had were because of nats |
| 08:30.59 | slav3_kitten | fling, same here |
| 08:31.06 | bombev | fling well, what is logrotate? |
| 08:31.37 | fling | bombev: the thing that rotates your logs, compresses old logs |
| 08:31.38 | slav3_kitten | i have outbound calling via sip flowroute working fine, inbound calling via iax2 from voip.ms is fine. what i can't get to work is anything coming in from wan |
| 08:31.41 | eject_ck | I need to record inbound calls from customers (where extension act like voicemail box). As suggested there I'm using Mixmonitor application, but it not work for me :( |
| 08:31.52 | slav3_kitten | like i want a friend to call in via a softphone. no-go |
| 08:31.55 | eject_ck | I'm getting Auto fallthrough, channel 'SIP/508-00000825' status is 'UNKNOWN' |
| 08:31.59 | fling | bombev: so your disk never get full |
| 08:32.05 | *** join/#asterisk mihamina (~mihamina@41.190.237.66) |
| 08:32.14 | eject_ck | right after call come in. |
| 08:32.36 | bombev | i got it fling |
| 08:32.42 | fling | slav3_kitten: test your packet forwarding |
| 08:32.50 | bombev | but first how to clean my log |
| 08:33.00 | eject_ck | I've tried with record() application, which is OK, but I need to convert file to after hangup and can't figure how can I do that ... |
| 08:33.01 | bombev | for example iam in ssh terminal |
| 08:33.02 | fling | slav3_kitten: run tcpdump on server and try to tonnect to some ports |
| 08:33.07 | slav3_kitten | fling you mean generate udp to my wan an look for it in tcpdump? |
| 08:33.31 | fling | slav3_kitten: yes |
| 08:34.03 | eject_ck | my dialplan I'm using with MixMonitor |
| 08:34.04 | fling | slav3_kitten: you probably need 5060,10000:20000 |
| 08:34.04 | eject_ck | exten => 508,n,Set(filename=/home/monitor/${UNIQUEID}) |
| 08:34.05 | eject_ck | exten => 508,n,Set(convert=nice -n 19 /usr/bin/lame -b 64 --silent "${filename}.wav" "${filename}.mp3" && rm -f "${filename}.wav" ) |
| 08:34.05 | eject_ck | exten => 508,n,MixMonitor(${filename}.wav, v(2) V(2) W(2), ${convert}) |
| 08:34.23 | slav3_kitten | fling, got that |
| 08:34.29 | bombev | fling : just to delete the log files: full full1 full2 full3 full4 in /var/log/asterisk ? |
| 08:34.47 | fling | bombev: I do not have these files |
| 08:36.40 | slav3_kitten | fling, you remember the linux command to generate udp traffic? |
| 08:37.10 | bombev | fling what kind of files do you have |
| 08:38.05 | eject_ck | Why it don't record call ? |
| 08:38.22 | slav3_kitten | hping! |
| 08:38.31 | fling | slav3_kitten: nmap -sU -p... |
| 08:41.22 | eject_ck | As I see MixMonitor needs to be used on Active channel, |
| 08:41.35 | fling | slav3_kitten: [asterisk] <-NAT [cisco2] <-NAT [cisco1] <- wan internets? |
| 08:41.48 | eject_ck | so how can I record inbound calls with it ? |
| 08:41.54 | fling | slav3_kitten: you do not have a control over cisco1? |
| 08:43.12 | slav3_kitten | fling, it's like this asterisk <-NAT [Cisco1] <- [ubiquiti loco 2 in bridge mode] NAT [some internal network sit with my isp] <- WAN |
| 08:43.22 | slav3_kitten | i have control of cisco1 |
| 08:43.31 | slav3_kitten | and set up forwarding like http://pastie.org/5385221 example b |
| 08:43.58 | fling | slav3_kitten: ok |
| 08:45.09 | fling | slav3_kitten: so test it and if packet forwarding works properly just set up your sip.conf |
| 08:45.38 | fling | slav3_kitten: nat, localnet, externip |
| 08:45.53 | slav3_kitten | my cisco skills are not nearly what they used to be and apparently it does not work |
| 08:45.54 | fling | slav3_kitten: are we talking about sip? :P |
| 08:46.35 | slav3_kitten | yes... sip |
| 08:46.47 | fling | slav3_kitten: you tested and DNAT is not working? |
| 08:46.56 | fling | slav3_kitten: I mean packets are not forwarded? |
| 08:47.31 | slav3_kitten | yea not being forwarded. |
| 08:47.36 | slav3_kitten | so far as i can tell |
| 08:48.25 | kaldemar | eject_ck: you have yet to show that it does not work. |
| 08:48.34 | fling | slav3_kitten: you found it! asterisk is not bad, your cisco/isp is! |
| 08:48.59 | fling | kaldemar: I want to record calls too :p |
| 08:49.29 | kaldemar | fling: all calls or start/stop recording by pressing a key on the phone? |
| 08:49.42 | fling | kaldemar: all calls |
| 08:50.01 | kaldemar | fling: use MixMonitor in your dialplan |
| 08:50.05 | fling | ok |
| 08:50.33 | slav3_kitten | fling, apparently |
| 08:50.57 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:fd28:2952:43a6:d3d3) |
| 08:51.48 | kaldemar | eject_ck: don't use whitespace around options in the MixMonitor app. |
| 08:52.21 | *** join/#asterisk hebber (~hebber@node-14pg.pool-125-25.dynamic.totbb.net) |
| 08:58.14 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 09:05.11 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 09:09.51 | *** join/#asterisk Robotman321 (~brad@50-194-126-9-static.hfc.comcastbusiness.net) |
| 09:17.18 | *** join/#asterisk vfabi (~fabi@93.116.255.140) |
| 09:22.07 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-objgxckcibdjmyil) |
| 09:23.54 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.250) |
| 09:25.01 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.250) |
| 09:29.06 | *** join/#asterisk jacekowski (jacekowski@jacekowski.org) |
| 09:29.06 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 09:35.27 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 09:39.15 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 09:39.54 | *** join/#asterisk TimeRider (~steve@81.136.216.215) |
| 09:42.03 | *** join/#asterisk EmleyMoor (phil@topdeck.tinsleyviaduct.com) |
| 09:42.57 | EmleyMoor | I need to set up a configuration server for my new Aastra phone - is there any particular recommendation as to which type? (tftp, ftp, http, https) |
| 09:43.17 | EmleyMoor | (I'd keep it on the same box as asterisk, I guess) |
| 09:46.46 | mathis_ | no idea what an Aastra phone is |
| 09:47.11 | EmleyMoor | mathis_: They're a fairly well known make of SIP desk phone |
| 09:47.23 | mathis_ | okay |
| 09:49.39 | mathis_ | well. pick what is easiest to install I guess. :) |
| 09:49.42 | mathis_ | not really asterisk related |
| 09:55.49 | *** join/#asterisk pietro (~pietro@78-134-118-111.v4.ngi.it) |
| 09:55.52 | pietro | Hello |
| 09:56.46 | pietro | Someone can explain me this ? http://pastebin.com/D6iVS5Gh |
| 09:56.51 | pietro | is this a known issue ? |
| 09:57.52 | pietro | also removing directly the key doesn't works.. |
| 09:58.24 | WIMPy | LegsVars != LegVars |
| 09:58.49 | pietro | whoo!! |
| 09:59.03 | pietro | WIMPy: I need lunettes !! ;) |
| 09:59.12 | pietro | WIMPy: thanks a lot ! |
| 09:59.46 | WIMPy | Yes. /That/ is a known issue :-) |
| 10:00.01 | pietro | ;) |
| 10:03.07 | *** join/#asterisk AliRezaTaleghani (~AliRezaTa@unaffiliated/AliRezaTaleghani) |
| 10:03.28 | AliRezaTaleghani | i have a problem with sip history... |
| 10:03.51 | AliRezaTaleghani | as mentioned in docs, i have enables history and dump of sip on in sip.conf |
| 10:04.13 | AliRezaTaleghani | but is don't be dumped in any on log channels :-/ |
| 10:04.22 | AliRezaTaleghani | I have enable debug channel too |
| 10:06.15 | kaldemar | which debug? the history is dumped in core debug. |
| 10:17.52 | AliRezaTaleghani | kaldemar: do u mean in the time servers crash? |
| 10:17.54 | AliRezaTaleghani | :-/ nooop |
| 10:18.09 | AliRezaTaleghani | I just need to have calls sip history in my log files... |
| 10:20.14 | kaldemar | AliRezaTaleghani: i didn't say anything about crashing. "core debug" is a type of debug in asterisk. enabled in CLI with "core set debug 10" for example. logged to file via logger.conf with the "debug" level. it will create a lot of logs, beware. |
| 10:21.59 | AliRezaTaleghani | kaldemar: :"> so, in the way to describe... i will not be able to have these logs (sip history) in debug log files.. unless set core debug level of 10 |
| 10:22.02 | AliRezaTaleghani | it's bad |
| 10:23.10 | kaldemar | no. now you're just making stuff up. |
| 10:23.55 | kaldemar | the CLI enable was just an example so you know what core debug is. |
| 10:24.36 | kaldemar | what gets created upon crash (if asterisk is allowed to do so) is code dump. |
| 10:24.45 | kaldemar | s/code/core/ |
| 10:24.57 | kaldemar | core dump is not the same as core debug. |
| 10:25.27 | AliRezaTaleghani | kaldemar: ok... think get what u mean... |
| 10:25.30 | AliRezaTaleghani | tnx |
| 10:39.27 | *** join/#asterisk kresp0 (~kresp0@81.61.24.25.dyn.user.ono.com) |
| 10:49.48 | *** join/#asterisk ujjain (~ujjain@unaffiliated/ujjain) |
| 11:18.32 | *** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
| 11:31.12 | *** join/#asterisk TSM (~the_softw@46-65-201-69.zone16.bethere.co.uk) |
| 11:32.21 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 11:38.07 | *** join/#asterisk santa0536 (~santa@cn-bgp-nat.portaone.com) |
| 11:47.34 | *** join/#asterisk sgimeno (~sgimeno@163.117.206.10) |
| 11:48.30 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 11:57.07 | *** join/#asterisk fritz09 (~Adium@pop1-224.catv.wtnet.de) |
| 12:00.57 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 12:13.31 | gavimobile | iptables has no rule for ports 5060-5061, how is it possible that my server is registering with my itsp |
| 12:14.48 | kaldemar | gavimobile: you have no rules that would stop it from doing so. |
| 12:15.06 | *** join/#asterisk timahvo1 (~rogue@41.212.120.182) |
| 12:18.20 | *** join/#asterisk jozza (~chatzilla@BSN-77-186-224.dsl.siol.net) |
| 12:20.27 | jozza | hi, i have problem with incoming calls from peers whose callerid is the same as one of my extensions. It is automatically rejected with 401 unauthorized. What do i need to set on the peer to skip caller id settings? |
| 12:20.50 | gavimobile | kaldemar: my default is to drop |
| 12:21.31 | kaldemar | gavimobile: doesn't sound like it is. what is your setup exactly and what happens? |
| 12:24.33 | gavimobile | kaldemar: well I have a router which forwards the necessary ports to my pbx, now I want to increase the secuirty of my server so I installed iptables on my pbx and I only allow specific ip addresses. it seems to be working cause when I change the ip address in the rule to an incorrect ip it no longer has audio but is still able to register I believe |
| 12:26.19 | *** join/#asterisk Anubi (~giovanni@89-97-222-18.ip19.fastwebnet.it) |
| 12:26.28 | *** join/#asterisk wonderworld (~w@dsdf-4db55db8.pool.mediaWays.net) |
| 12:28.24 | bulkorok | how do I can make asterisk check if a call is a fax and only then send it to ReceiveFax... otherwise it should hangup or sth... |
| 12:28.33 | kaldemar | gavimobile: your description is far from exact. |
| 12:28.55 | mirela666 | jozza: try to increase call-limit parameter for those peers |
| 12:30.20 | *** join/#asterisk nicknam1232 (021d24e5@gateway/web/freenode/ip.2.29.36.229) |
| 12:30.24 | gavimobile | kaldemar: my appoligies... |
| 12:32.02 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 12:33.09 | Anubi | hi, i'm new in asterisk |
| 12:33.22 | AliRezaTaleghani | Anubi: as all of us.... |
| 12:33.23 | Anubi | i configured it and now i can call and receive to and from extern call, but i cannot call internal number... some helps? |
| 12:33.53 | AliRezaTaleghani | u need to define some what which calls dialplan |
| 12:34.19 | AliRezaTaleghani | the simples think i can give u is this: |
| 12:34.26 | AliRezaTaleghani | vim /etc/asterisk/extensions.conf |
| 12:34.45 | AliRezaTaleghani | find the context which match to your sip caller context |
| 12:34.50 | AliRezaTaleghani | add this: |
| 12:35.09 | AliRezaTaleghani | exten => _X.,1,Answer |
| 12:35.22 | AliRezaTaleghani | exten => _X.,1,Dial(SIP/${EXTEN}) |
| 12:35.26 | AliRezaTaleghani | exten => _X.,1,Hangup |
| 12:35.41 | AliRezaTaleghani | ====== need to corrent this way: |
| 12:35.43 | mirela666 | bulkorok: by malcolmd » Wed Nov 10, 2010 3:14 pm |
| 12:35.43 | mirela666 | If the call is determined to be a fax, based upon the receipt of fax tones, then it'll fall over to a special extension in the current context called "fax." |
| 12:35.50 | AliRezaTaleghani | exten => _X.,1,Answer |
| 12:35.56 | AliRezaTaleghani | exten => _X.,2,Answer |
| 12:36.02 | AliRezaTaleghani | exten => _X.,3,Answer |
| 12:36.22 | AliRezaTaleghani | :P just correct the priority... |
| 12:36.23 | Anubi | i wrote this |
| 12:36.25 | AliRezaTaleghani | ;) |
| 12:36.28 | Anubi | exten => _X./201,1,Set(CALLERID(num)=0690282661) |
| 12:36.28 | Anubi | exten => _X./202,1,Set(CALLERID(num)=0690282662) |
| 12:36.28 | Anubi | exten => _X./203,1,Set(CALLERID(num)=0690282663) |
| 12:36.28 | Anubi | exten => _X./204,1,Set(CALLERID(num)=0690282664) |
| 12:36.28 | Anubi | exten => _X./205,1,Set(CALLERID(num)=0690282665) |
| 12:36.29 | Anubi | exten => _X./206,1,Set(CALLERID(num)=0690282666) |
| 12:36.31 | Anubi | exten => _X./207,1,Set(CALLERID(num)=0690282667) |
| 12:36.33 | Anubi | exten => _X./208,1,Set(CALLERID(num)=0690282668) |
| 12:36.35 | Anubi | exten => _X.,n,Dial(SIP/${EXTEN}@Eutelia,30,T) |
| 12:36.37 | Anubi | exten => _X.,n,Hangup |
| 12:36.39 | Anubi | exten => 2XX,1,Dial(SIP/${EXTEN},30,Tt) |
| 12:36.43 | Anubi | exten => 2XX,n,Hangup |
| 12:36.53 | WIMPy | Stop flooding the channel |
| 12:36.56 | WIMPy | ~pb |
| 12:36.56 | 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. |
| 12:37.07 | Anubi | excuseme |
| 12:37.23 | AliRezaTaleghani | Anubi: set verbose 3 and give me the result |
| 12:37.48 | bulkorok | mirela666: that's what I wanted to post now too :-) thx |
| 12:38.56 | bulkorok | we have dedicated numbers for fax, but sometimes somebody calls the faxnumber and we get a "fax could not be received" message... (for sure) |
| 12:39.17 | bulkorok | I need to stop sending the message and just hangup... |
| 12:39.45 | Anubi | AliRezaTaleghani, http://pastebin.com/U8x9j4nM |
| 12:41.25 | WIMPy | Anubi: Looks liek you haven't defined any extensions. At least not that one. pb your extensions.conf. |
| 12:41.43 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-azhslgamsgikfxns) |
| 12:41.43 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 12:42.30 | *** part/#asterisk AliRezaTaleghani (~AliRezaTa@unaffiliated/AliRezaTaleghani) |
| 12:48.48 | *** join/#asterisk felipealmeida (~user@mvx-187-16-79-187.mundivox.com) |
| 12:59.34 | *** join/#asterisk mihamina (~mihamina@41.190.237.66) |
| 13:01.03 | *** part/#asterisk kresp0 (~kresp0@81.61.24.25.dyn.user.ono.com) |
| 13:03.03 | kaldemar | Anubi: "exten => 2XX" <-- is missing an underscore in front of the pattern. it should be "exten => _2XX" |
| 13:04.17 | kaldemar | Anubi: without the underscore, the extension would match literal "2XX", not 2[0-9][0-9]. |
| 13:08.30 | *** join/#asterisk [TK]D-Fender (~TK]D-Fend@216-191-106-165.dedicated.allstream.net) |
| 13:16.28 | *** join/#asterisk italorossi (~textual@189.124.196.68) |
| 13:18.50 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 13:18.59 | *** join/#asterisk blee (~blee@72.188.117.219) |
| 13:22.17 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
| 13:22.49 | *** join/#asterisk keycruncher (~Adium@c-174-55-112-94.hsd1.pa.comcast.net) |
| 13:27.15 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 13:31.28 | Anubi | sure... kaldemar tnx now works fine, but i didn't see the caller in the display... |
| 13:31.53 | *** join/#asterisk JohnHurst (~john@109.111.202.18) |
| 13:33.18 | *** join/#asterisk elico (~Thunderbi@bzq-79-181-215-110.red.bezeqint.net) |
| 13:40.41 | *** join/#asterisk b3nt_pin (~quassel@stjhnf0148w-142134079216.dhcp-dynamic.FibreOP.nl.bellaliant.net) |
| 13:41.30 | jozza | mirela666: no, the call-limit does not help |
| 13:46.14 | *** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net) |
| 13:47.11 | Anubi | kaldemar, http://pastebin.com/e8iJtUYw why this? |
| 13:47.20 | gavimobile | should my peers all be set to type=peer? my goal is to REQUIRE valid registration to my pbx only! |
| 13:48.21 | [TK]D-Fender | gavimobile, Yes, and peers auth just fine |
| 13:48.45 | [TK]D-Fender | Anubi, [Nov 20 14:44:31] VERBOSE[5300] app_dial.c: -- Now forwarding SIP/Eutelia-0000002a to 'Local/06571711@from-internal' (thanks to SIP/202-0000002b) |
| 13:48.54 | [TK]D-Fender | Anubi, Because your phone is FORWARDED |
| 13:49.23 | Anubi | [TK]D-Fender, how can i change this? |
| 13:49.34 | gavimobile | [TK]D-Fender: will this effect anything else? my peers are all currently set to type=friend |
| 13:50.07 | [TK]D-Fender | Anubi, Walk over to your phone and turn off the forwarding. |
| 13:50.25 | [TK]D-Fender | gavimobile, Generally you can use "peer" for everything |
| 13:51.11 | gavimobile | [TK]D-Fender: ok thanks |
| 13:53.39 | gavimobile | if type=friend is a security threat, why would newer versions of asterisk use it |
| 13:53.44 | Anubi | [TK]D-Fender, tnx |
| 13:55.40 | [TK]D-Fender | gavimobile, Who said it was a threat? |
| 13:56.05 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 13:56.09 | gavimobile | [TK]D-Fender: well why would we want people to connect to out pbx without registration? |
| 13:56.53 | [TK]D-Fender | in case you run into reg timeouts. |
| 13:58.03 | gavimobile | [TK]D-Fender: I see |
| 13:58.07 | [TK]D-Fender | Also required in cases where multiple devices source from the same IP. |
| 13:58.50 | gavimobile | [TK]D-Fender: ahh, right. I remember I had an issue like that |
| 13:59.42 | *** join/#asterisk londonnet2213 (~londonnet@host86-154-213-120.range86-154.btcentralplus.com) |
| 14:00.03 | gavimobile | could you tell me about allowguest=yes? what does it mean allow anonymous callers. what makes a caller anonymous? |
| 14:03.20 | *** join/#asterisk zerohalo (~zerohalo@74.61.196.236) |
| 14:03.26 | Anubi | now... i would show internal caller id, but i see only "device"... i tried to change extension configuration, but didn't work... |
| 14:07.13 | londonnet2213 | Quick question. Are the centos asterisk repositories going to be updated to current releases of asterisk soon or have they been depreciated and updates are now only available as un compiled? |
| 14:08.09 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 14:09.11 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 14:09.43 | [TK]D-Fender | gavimobile, Not matching an entry you have. |
| 14:10.12 | [TK]D-Fender | gavimobile, I you say "Hi, I'm Bob!", and I KNOW "Bob" ..... then I guess you aren't ANONYMOUS. |
| 14:10.33 | [TK]D-Fender | Anubi, Show us the call. |
| 14:10.37 | [TK]D-Fender | ~PB |
| 14:10.37 | 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. |
| 14:10.43 | [TK]D-Fender | Anubi, ^ |
| 14:10.48 | *** join/#asterisk kriger (~norge@68-190-90-80.dhcp.mdsn.wi.charter.com) |
| 14:11.23 | [TK]D-Fender | londonnet2213, There are current 1.8/10 versions. Not sure if 11 made it yet, but should be shortly otherwise |
| 14:12.27 | jozza | i have problem with incoming calls from peers whose user in "from" header is the same as one of my extensions. It is automatically rejected with 401 unauthorized, because asterisk finds this user in local user list. How do i overcome this? |
| 14:12.42 | londonnet2213 | Good news re V11 but we are stuck on 10.7.1 where 10.11.x?? is available. Will V10 repo's get updated soon as well? |
| 14:13.02 | Anubi | [TK]D-Fender, http://pastebin.com/UYvjGwAX |
| 14:15.13 | *** join/#asterisk clack (~clarsson@152.179.75.170) |
| 14:15.20 | *** join/#asterisk nickfennell (~nickfenne@unaffiliated/nickfennell) |
| 14:15.21 | jozza | i tried with insecure=invite, which seemed (according to docs) the most reasonable solution, but ast still returns 401 |
| 14:16.23 | jozza | and i can't dictate what the incoming call's "user" will be |
| 14:16.23 | WIMPy | jozza: Call coming from where? ITSP? Guests? |
| 14:16.46 | jozza | call incoming from a sip peer |
| 14:17.00 | jozza | to which i am registered |
| 14:17.14 | WIMPy | What kind of peer? |
| 14:17.20 | jozza | sip |
| 14:17.26 | WIMPy | Then authenticate that peer via IP. |
| 14:17.34 | jozza | i did |
| 14:17.50 | jozza | i put host property in |
| 14:18.02 | jozza | host=ip address |
| 14:18.23 | WIMPy | And no user? |
| 14:19.06 | jozza | what do you mean? |
| 14:19.13 | [TK]D-Fender | Anubi, "sip set debug on" <- |
| 14:19.36 | *** join/#asterisk dwayne (~dwayne@76.29.230.45) |
| 14:19.52 | jozza | incoming calls from that peer with different user than my exiting extensions work fine |
| 14:20.14 | *** join/#asterisk vlad_starkov (~vlad_star@79.104.6.220) |
| 14:20.39 | jozza | i'm on 1.6.2.24, should i upgrade? |
| 14:20.47 | WIMPy | The calls don't seem to be hiting the peer you want them to go to. |
| 14:21.19 | WIMPy | I would upgrade that. |
| 14:22.12 | jozza | asterisk says: Found peer '506' for '506' from xxx.xxx.xxx.xxx:5060 and then sends out 401 |
| 14:22.17 | *** join/#asterisk nickfennell_ (~nickfenne@unaffiliated/nickfennell) |
| 14:22.25 | jozza | xxx being the peer's ip |
| 14:22.48 | *** join/#asterisk afink (4172ce22@gateway/web/freenode/ip.65.114.206.34) |
| 14:23.47 | *** join/#asterisk nickfennell__ (~nickfenne@unaffiliated/nickfennell) |
| 14:23.51 | kaldemar | jozza: insecure=port,invite is probably what you want. |
| 14:24.08 | jozza | i did, not working, like it is ignored |
| 14:24.32 | kaldemar | feel free to pastebin configs and sip debug. |
| 14:25.01 | Anubi | [TK]D-Fender, http://pastebin.com/vMpmgw5K is it ok? |
| 14:25.09 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 14:25.09 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 14:27.11 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:27.11 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:36.28 | *** join/#asterisk vlad_starkov (~vlad_star@83.149.9.175) |
| 14:37.59 | *** join/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net) |
| 14:39.35 | [TK]D-Fender | Anubi, From: "device" <sip:206@192.168.2.241>;tag=as3a85e015 <-- set the callerid for your peer |
| 14:39.40 | [TK]D-Fender | Anubi, You didn't put a name |
| 14:39.55 | p3nguin | <jozza> and i can't dictate what the incoming call's "user" will be <------- Then do not use type=user nor type=friend. type=peer does not know about user name. |
| 14:40.09 | Anubi | [TK]D-Fender, i tried to modify sip_additional.conf |
| 14:40.11 | jozza | i am |
| 14:40.19 | Anubi | i setted callerid with right name |
| 14:40.27 | [TK]D-Fender | Anubi, Sounds like you're trying to rebuild a FreePBX system.... |
| 14:40.27 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 14:40.28 | Anubi | and now work |
| 14:41.05 | kaldemar | p3nguin: type=peer does know about username when configured with host=dynamic. |
| 14:41.07 | Anubi | good... it's my question.... |
| 14:41.45 | Anubi | i think that after rebuild error will return |
| 14:41.51 | jozza | p3nguin: i have type=peer and host=ip address for peer and the conflicting extension is type=friend and host=dynamic |
| 14:42.00 | p3nguin | But he said he defined the IP address, so it won't know about the username. |
| 14:42.13 | [TK]D-Fender | Anubi, Are you trying to hand-modify a system that is still being configured via FreePBX? |
| 14:42.15 | p3nguin | Peers no NOTHING about extensions. |
| 14:42.31 | jozza | yes, that is the problem |
| 14:42.35 | Anubi | [TK]D-Fender, yep |
| 14:42.56 | p3nguin | It's not a problem because the problem cannot exist. Peers do not know anything about what is in extensions. |
| 14:42.56 | [TK]D-Fender | Anubi, You should not be touching the dialplan or sip configs they way you have been then. |
| 14:43.12 | Anubi | but in gui i didn't find callerid |
| 14:43.13 | [TK]D-Fender | Anubi, You are asking for serious problems and defeating the purpose of having a GUI |
| 14:43.17 | jozza | p3nguin: ast seems to find the extension before it looks for peer ip |
| 14:43.18 | [TK]D-Fender | Anubi, itsw the NAME |
| 14:43.25 | *** join/#asterisk bchia (~Adium@nat/digium/x-pqljkccyqsfzfymn) |
| 14:43.36 | p3nguin | It can't do that. Peers don't know anything about extensions. |
| 14:43.51 | kaldemar | jozza: extensions are in extensions.conf, the [506] in sip.conf is not called an extension but a device, or peer. |
| 14:44.00 | [TK]D-Fender | Display Name<- |
| 14:44.01 | p3nguin | A peer must be matched first, then the call is routed into the context assigned. |
| 14:44.16 | jozza | in freepbx, extensions are in sip.conf |
| 14:44.18 | kaldemar | jozza: you're not using correct terminology as it is in asterisk. |
| 14:44.19 | p3nguin | Once the call goes to the context, then extensions are matched and executed. |
| 14:44.37 | jozza | right |
| 14:44.39 | p3nguin | Extensions are not in sip.conf, they are in extensions.conf. |
| 14:44.43 | [TK]D-Fender | jozza, And don't assume SIP. |
| 14:44.51 | [TK]D-Fender | for "phones" |
| 14:45.03 | jozza | ok, device 506 is in sip.conf |
| 14:45.09 | Anubi | [TK]D-Fender, but didn't work i setted this right |
| 14:45.16 | jozza | and it is a local device |
| 14:45.27 | p3nguin | So the actual problem is that your physical peer is matching the peer entry of another device. |
| 14:45.44 | jozza | yes |
| 14:45.50 | *** join/#asterisk xnt14 (~xnt14@xceleo.us) |
| 14:45.50 | p3nguin | To which I say, this should be an easy fix. |
| 14:45.56 | [TK]D-Fender | Anubi, You are already executing NON FreePBX dialplan and I don't see your extension setup. You are screwing with your system manually and you are going to break things. Stop trying to break your system. |
| 14:46.10 | jozza | ok |
| 14:46.14 | *** join/#asterisk gusto (~gusto@2001:a60:11ff:1200::42:4) |
| 14:46.14 | jozza | what? |
| 14:46.42 | Anubi | [TK]D-Fender, k |
| 14:46.56 | p3nguin | If the IP address of the phone is dynamic and you have set host=dynamic in that peer entry, that part should be done. Move on to the other peer entry... |
| 14:46.58 | Anubi | i'll work only trough web gui |
| 14:47.39 | jozza | true, and the type=friend on the phone |
| 14:47.45 | p3nguin | In the other peer entry, type=peer, remotesecret=<the-password-to-send-when-making-calls-VIA-this-peer> host=<the-IP> |
| 14:47.57 | *** join/#asterisk jmetro (~nickserv@75-150-221-198-Illinois.hfc.comcastbusiness.net) |
| 14:48.26 | jozza | what will remote secret do? |
| 14:49.06 | p3nguin | remotesecret doesn't force a peer to auth calls TO YOU, but will send the password when you call to that peer and it requires you to auth to it. |
| 14:49.16 | p3nguin | Perfect for ITSPs. |
| 14:49.38 | p3nguin | They always require you to auth when you make calls, but they never send auth to you when they send calls to you. |
| 14:50.11 | jozza | yes, beacuse i' am an extension to the remote peer |
| 14:50.22 | p3nguin | That statement doesn't really make sense. |
| 14:50.35 | jozza | why |
| 14:50.51 | jozza | the remote system treats me as an extension in their system |
| 14:51.00 | p3nguin | Nonsense. |
| 14:51.13 | p3nguin | Your terms don't make sense. |
| 14:51.20 | jozza | ok, i see what bothers you |
| 14:52.59 | jozza | well, i register to this peer with registration string and have set the type=peer and host=ip address, so it all works fine |
| 14:53.49 | jozza | so when they dial my "extension" the call is routed to me on my peer |
| 14:54.15 | jozza | i am to them like a host=dynamic and type=friend with a secret |
| 14:55.36 | p3nguin | That's typical with an ITSP. |
| 14:56.09 | p3nguin | You register to them and you auth with your user/secret. Very standard. |
| 14:56.34 | jozza | but i've testet this on another ast to make call to my peer |
| 14:57.25 | jozza | and if i dont set the fromuser on that ast, the problem is the same when that ast has same user in the from header |
| 14:57.36 | p3nguin | Did you ever pastebin the sip.conf, which contains the conflicting properties? |
| 14:57.47 | jozza | no |
| 15:03.07 | jozza | i have freepbx and its a bit complicated |
| 15:03.19 | jozza | p3nguin: i have freepbx and its a bit complicated |
| 15:06.56 | p3nguin | I can imagine. That's part of the reason we don't support configuration by GUI here. |
| 15:07.37 | jozza | ok, i'll try to pastebin the most relevant info and a trace |
| 15:13.03 | jozza | p3nguin: http://pastebin.com/3AqMf53W |
| 15:13.38 | [TK]D-Fender | jozza, If you're using FreePBX, then WTF are you doing hand-editing the files? |
| 15:13.48 | jozza | i am not |
| 15:14.49 | [TK]D-Fender | jozza, Ok, A few thing looked odd for ti to have allowed through. Show us the call. |
| 15:15.16 | [TK]D-Fender | jozza, and repase the confi masking ONLY the secrets |
| 15:15.59 | jozza | that is the call |
| 15:16.13 | jozza | it ends after another ack which i didnt include |
| 15:16.24 | jozza | you want ip adresses? |
| 15:16.49 | jozza | i have distinguished tha ip adresses with x and y |
| 15:17.01 | bulkorok | does asterisk has an application or sth that shows me all (channel)-variables on cli?! |
| 15:19.05 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 15:19.05 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 15:19.36 | [TK]D-Fender | jozza, Do not mask. And is your trunk & phone at the same IP? |
| 15:19.52 | jozza | no |
| 15:19.52 | jozza | the phone is not even regitered |
| 15:19.55 | [TK]D-Fender | bulkorok, "core show channel [thechannel]" |
| 15:20.11 | *** join/#asterisk vlad_starkov (~vlad_star@83.149.9.175) |
| 15:20.19 | jozza | i have to mask the pastebin is public |
| 15:20.20 | [TK]D-Fender | jozza, "sip show peer [X]", dump both, repaste the call AND configs, masking only the secrets. |
| 15:20.27 | *** join/#asterisk tonikasch (~tonikasch@unaffiliated/tonikasch) |
| 15:20.31 | jozza | yes, i'm doing that |
| 15:20.35 | [TK]D-Fender | jozza, then make it private and PW it and PM the PW |
| 15:22.26 | jozza | cant make it private, i'm not registered |
| 15:23.42 | [TK]D-Fender | PW it |
| 15:27.32 | jozza | i signed up, how do i PW it? i cant see a setting for that |
| 15:28.01 | [TK]D-Fender | Pick another site. |
| 15:28.06 | [TK]D-Fender | There are dozens of PB sites out there. |
| 15:28.31 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 15:29.11 | jozza | got to take a break few min. |
| 15:36.20 | EmleyMoor | Anyone know where I can get help with an Aastra phone? I have a config that works, but the phone won't pull it from the server |
| 15:37.57 | slav3_kitten | zombie shuffles in |
| 15:40.32 | slav3_kitten | [TK]D-Fender, you around? |
| 15:40.44 | slav3_kitten | or p3nguin |
| 15:42.39 | *** join/#asterisk ghost75 (~trechber@dslb-188-105-026-047.pools.arcor-ip.net) |
| 15:45.16 | ghost75 | i have strange result with agi |
| 15:45.38 | ghost75 | works: exten => s,n,AGI(apachemaint.agi,www,off) |
| 15:45.58 | ghost75 | doesnt work: exten => s,n,AGI(apachemaint.agi,www,on) |
| 15:46.10 | p3nguin | Redirected from a different person: |
| 15:46.10 | p3nguin | (2100.25) <p3nguin> It would be a lot easier if you'd just say whatever it is you want to tell me instead of first asking if I'm here, then making me answer that, then I have to wait for you to come back and type whatever it was that you wanted to tell me in the first place. |
| 15:46.39 | ghost75 | bash script http://pastebin.com/Ffdudbz4 |
| 15:47.08 | ghost75 | if i start apachemaint.agi from bash with parameters it works too |
| 15:47.39 | slav3_kitten | p3nguin, i was wondering if i could get you to originate a SIP call to my ip address that i give you in PM to see if this is working or if i need to spend some hours on the phone with my ISP today. i trust you [tk]d-fender and dijib with my public ip |
| 15:48.15 | *** join/#asterisk Neptu (~Neptu@c-af90e255.113-1-64736c14.cust.bredbandsbolaget.se) |
| 15:49.55 | slav3_kitten | on an entirely unrelated note, borderlands 2 dlc is out! |
| 15:51.09 | *** join/#asterisk Vince-0 (~Vincent@41-132-156-117.dsl.mweb.co.za) |
| 15:51.10 | p3nguin | I can do that. |
| 15:51.23 | p3nguin | Need extension and address. |
| 15:51.25 | ghost75 | p3nguin: do you think this is valid visudo entry: Cmnd_Alias C5 = /bin/mv /var/www* /var/www* |
| 15:51.32 | p3nguin | nope |
| 15:51.33 | *** join/#asterisk fritz09 (~Adium@pop1-224.catv.wtnet.de) |
| 15:51.43 | ghost75 | doesnt like * ? |
| 15:52.03 | p3nguin | I wouldn't think it would work, but it would only take a minute to test it. |
| 15:52.50 | p3nguin | I'm also curious that you have directories named www<ANYTHING> in /var. |
| 15:53.03 | *** join/#asterisk [TK]D-Fender (~TK]D-Fend@216.191.106.165) |
| 15:53.14 | ghost75 | i have www, www2 and so on |
| 15:53.46 | p3nguin | And you're allowing the moving of www2 to something like wwwwwwwww? |
| 15:54.08 | ghost75 | i allow it to user asterisk |
| 15:54.13 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 15:54.14 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 15:54.56 | ghost75 | aha, no permissions :) |
| 15:55.34 | ghost75 | if i give permissions on a single file, is the user allowed to move that to a different file? |
| 15:56.59 | ghost75 | mhh looks not |
| 15:57.24 | [TK]D-Fender | jozza, Who is sending you that call? |
| 15:57.37 | jozza | RemotePeer |
| 15:57.42 | slav3_kitten | thank you very much p3nguin |
| 15:57.47 | p3nguin | No problem. |
| 15:57.49 | [TK]D-Fender | JohnHurst, WHO are they? |
| 15:57.51 | [TK]D-Fender | JOZZ^ |
| 15:59.46 | ghost75 | holy cow, i put this and still doesnt work Cmnd_Alias C5 = /bin/mv /var/www/maintenance-mode-off /var/www/maintenance-mode-on |
| 16:01.01 | ghost75 | ah i think have tomatoes on eyes |
| 16:01.58 | jozza | i am on 1.6.2.24, something stinks here |
| 16:02.39 | bulkorok | 1.6 is EOL |
| 16:02.46 | jozza | yeah |
| 16:02.59 | jozza | but wasnt once |
| 16:05.47 | ghost75 | today is now :) |
| 16:06.00 | ghost75 | visudo stinks too |
| 16:09.10 | jozza | [TK]D-Fender: does it make sense? |
| 16:09.55 | p3nguin | Better to use visudo and hate it than to use straight vi[m] and make syntax errors. |
| 16:12.01 | ghost75 | dont understand why it doesnt work |
| 16:13.58 | ghost75 | missed the sudo lol |
| 16:19.46 | *** join/#asterisk AkkerKid (~AkkerKid@50-200-18-202-static.hfc.comcastbusiness.net) |
| 16:22.21 | *** join/#asterisk a1fa (~a1fa@unaffiliated/a1fa) |
| 16:22.36 | a1fa | [TK]D-Fender : hello! |
| 16:25.27 | *** join/#asterisk JohnHurst (~john@109.111.202.18) |
| 16:26.00 | a1fa | what's a good sip provider these days? |
| 16:26.07 | Qwell | ~itsplist-us |
| 16:26.08 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 16:26.29 | a1fa | so has not changed much |
| 16:26.36 | Qwell | shrugs |
| 16:26.44 | a1fa | i'm using a combo of teliax pay as you go, and broadvoice |
| 16:27.09 | a1fa | 2 inbound dids through broadvoice, and teliax outbound :) |
| 16:27.50 | a1fa | $14/month, low volume -~ 300 minutes or so international + US |
| 16:36.41 | *** join/#asterisk BaconZombie (~baconzomb@pdpc/supporter/active/trenchspike) |
| 16:36.43 | BaconZombie | Hey |
| 16:37.19 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 16:54.29 | *** join/#asterisk vlad_starkov (~vlad_star@83.149.8.205) |
| 16:57.14 | *** join/#asterisk vlad_sta_ (~vlad_star@83.149.8.110) |
| 16:57.17 | *** join/#asterisk serafie (~erin@nat/digium/x-opvbbrldiogqrcmo) |
| 16:58.32 | *** join/#asterisk ffs (~garland@unaffiliated/ffs) |
| 17:03.12 | AkkerKid | Anyone know reasons why linux and asterisk would run really slowly for network stuff after changing LAN networks? |
| 17:04.41 | leifmadsen | no DNS lookups? |
| 17:04.45 | AkkerKid | SSH connections via putty take 15 seconds to pass for my password when logging in. Calls initiated take 5+ seconds to ring the destination... |
| 17:04.47 | leifmadsen | throws darts at a wall witha blindfold |
| 17:04.58 | leifmadsen | ya sounds like name lookup issues |
| 17:05.10 | Nivex | leifmadsen: and yet you still get pretty close to the target |
| 17:05.17 | leifmadsen | Nivex: IKR?! |
| 17:05.32 | Nivex | leifmadsen: file under "How I know I've been doing this too long" :) |
| 17:05.37 | AkkerKid | so the PBX has bad DNS servers? |
| 17:06.22 | leifmadsen | AkkerKid: your OS doesn't have valid DNS servers or they are not responding, or your gateway isn't working correctly or.... |
| 17:06.39 | leifmadsen | Nivex: I may or may not have seen this type of issue in a previous lifetime |
| 17:06.49 | leifmadsen | lunches like woah |
| 17:08.10 | Robotman321 | leifmadsen: your not allowed to have a life, let alone previous ones :D |
| 17:10.48 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 17:17.47 | *** join/#asterisk oej_ (~olle@2001:16d8:cc57:1000::42:1003) |
| 17:20.36 | *** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com) |
| 17:21.16 | *** part/#asterisk UForgotten (~u4go10@unaffiliated/uforgotten) |
| 17:22.35 | *** join/#asterisk jozza (~AndChat27@BSN-77-186-224.dsl.siol.net) |
| 17:26.22 | *** join/#asterisk jozza (~AndChat27@BSN-77-186-224.dsl.siol.net) |
| 17:26.27 | *** join/#asterisk CunningPike (~CunningPi@204.239.8.111) |
| 17:26.50 | *** join/#asterisk shido6 (~shido6@nat/yahoo/x-yxnengmxkhudhoyp) |
| 17:29.02 | *** join/#asterisk kresp0 (~kresp0@81.61.24.25.dyn.user.ono.com) |
| 17:31.00 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 17:31.11 | *** join/#asterisk parasitodelsur (~wtf@c-98-200-53-71.hsd1.tx.comcast.net) |
| 17:31.39 | parasitodelsur | waz up guys |
| 17:34.55 | *** join/#asterisk _Corey_ (~chatzilla@173.161.229.46) |
| 17:45.01 | *** join/#asterisk jsjc (~Adium@226.Red-80-33-236.staticIP.rima-tde.net) |
| 17:53.39 | *** join/#asterisk elico (~Thunderbi@bzq-79-181-215-110.red.bezeqint.net) |
| 18:00.09 | ChannelZ | The sky. HAR! |
| 18:00.56 | *** join/#asterisk navaismo (~navaismo@189.241.34.209) |
| 18:13.10 | *** join/#asterisk italorossi (~textual@189.124.196.68) |
| 18:13.46 | parasitodelsur | lol |
| 18:14.15 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 18:26.54 | *** join/#asterisk oej_ (~olle@2001:16d8:cc57:1000::42:1003) |
| 18:28.12 | *** join/#asterisk vlad_starkov (~vlad_star@83.149.8.110) |
| 18:34.20 | *** join/#asterisk martinlindhe (~martinlin@212.112.176.91) |
| 18:35.04 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 18:47.59 | pietro | I need to send a SIP INFO to both call patry when a call is connected. |
| 18:48.16 | pietro | Is there a way in asterisk dialplan to manage this ? |
| 18:48.57 | pietro | I tried using an external script via System() but this break the CSeq of next asterisk SIP messages |
| 18:49.38 | pietro | I'm searching for the right way to inject SIP messages in an active dialog.. |
| 18:49.48 | pietro | without patching chan_sip.c ;) |
| 18:52.45 | martinlindhe | hi all |
| 18:54.08 | martinlindhe | what would be the best way to Bridge two callers from a script, and then let them continue to use the script when they have finished chatting ? when i use the Bridge AMI command, one of the channels gets disconnected at the end of the call, while the other stays in the script |
| 18:58.28 | martinlindhe | (it is always the recieving end that gets disconnected), and also i use the Bridge AGI cmd, sorry |
| 18:59.00 | *** join/#asterisk [ProB]CrazyMan (~chatzilla@80.153.69.72) |
| 18:59.57 | *** join/#asterisk TheKernel[work] (~tcrowe@unaffiliated/the-kernel) |
| 19:00.35 | TheKernel[work] | I don't get why this string does not work: exten = _*21*X.,1,Set(DB(CFIM2/${CALLERID(num)})=${1}) |
| 19:00.46 | TheKernel[work] | I'm looking at the guide and that's how it explains it |
| 19:03.15 | [TK]D-Fender | TheKernel[work], ${1} <- not a valid variable name. |
| 19:03.25 | [TK]D-Fender | TheKernel[work], Nor do even see anything IN it. |
| 19:03.51 | TheKernel[work] | I just want it to have the value of "1" |
| 19:04.00 | [TK]D-Fender | the strip that other crap off |
| 19:04.31 | TheKernel[work] | so |
| 19:04.33 | TheKernel[work] | =1 |
| 19:04.37 | [TK]D-Fender | yes |
| 19:04.39 | TheKernel[work] | or =$1 |
| 19:04.40 | TheKernel[work] | ok |
| 19:05.30 | *** join/#asterisk afink (4172ce22@gateway/web/freenode/ip.65.114.206.34) |
| 19:06.23 | TheKernel[work] | [TK]D-Fender: syntax error: syntax error, unexpected '=', expecting $end; Input: |
| 19:06.24 | TheKernel[work] | <PROTECTED> |
| 19:07.40 | Qwell | and what does it look like now? |
| 19:08.17 | [TK]D-Fender | Show us the code AND the CLI output in full |
| 19:09.03 | *** join/#asterisk JerJer (~Adium@asterisk/original-h323-guy/JerJer) |
| 19:11.28 | TheKernel[work] | [TK]D-Fender: http://pastebin.com/hgEy12Ws |
| 19:11.56 | [TK]D-Fender | .... |
| 19:12.10 | [TK]D-Fender | Your set isn't even contained within the cbrackets |
| 19:12.31 | [TK]D-Fender | hold thath thought... |
| 19:14.14 | dijib | hello all! |
| 19:14.48 | [TK]D-Fender | TheKernel[work], Actually.... |
| 19:14.57 | TheKernel[work] | :/ |
| 19:15.03 | [TK]D-Fender | TheKernel[work], exten => _100,2,GoToIf($[${DB(CFIM2/100)} = 1]?CFA:NA) |
| 19:15.10 | [TK]D-Fender | TheKernel[work], That DB keypair IS blank |
| 19:15.29 | [TK]D-Fender | TheKernel[work], line #15 = FUBAR <- |
| 19:17.44 | TheKernel[work] | line 15 is supposed to set a value basically saying 1 or 0. Line 2 is supposed to check if its 1 or 0. If its 1 go to line 3, if its 0 go to line 5. |
| 19:18.16 | *** join/#asterisk timahvo1 (~rogue@41.212.120.182) |
| 19:19.12 | [TK]D-Fender | TheKernel[work], There is a tragic issue with it... |
| 19:19.19 | [TK]D-Fender | Look REALLY closely at it |
| 19:19.31 | JerJer | don't need _ without a match |
| 19:19.39 | TheKernel[work] | OH GOD |
| 19:19.42 | [TK]D-Fender | Priority ----> 1 <---- |
| 19:19.45 | TheKernel[work] | yeah |
| 19:19.46 | TheKernel[work] | I see it |
| 19:20.15 | [ProB]CrazyMan | hi, I have an problem with chan_lcr.so, I canot load it and always get " Module 'chan_lcr.so' was not compiled with the same compile-time options as this version of Asterisk." |
| 19:20.49 | Chainsaw | [ProB]CrazyMan: That would suggest that one of your packages is provided by a distributor, and the other by another distributor. You must have a matched pair. |
| 19:20.49 | [ProB]CrazyMan | I rebuilded lcr ... but still the same |
| 19:20.53 | Chainsaw | [ProB]CrazyMan: If you build LCR yourself, you should build the whole of Asterisk yourself. |
| 19:21.00 | JerJer | [ProB]CrazyMan: you need to build everything from source |
| 19:21.16 | [ProB]CrazyMan | is this new? |
| 19:21.55 | [TK]D-Fender | You didn't rebuilt it matching the rest of your * install |
| 19:22.33 | [ProB]CrazyMan | last time i also installed asterisk via yum and compiled LCR without problem |
| 19:22.41 | *** join/#asterisk bn-7bc (~bjarne-im@macbook-pro.lan-sx.noare-1.holmedal.net) |
| 19:23.38 | [TK]D-Fender | Versions do not match w/ compiled |
| 19:23.43 | dijib | [ProB]CrazyMan: you have all the dependencies? any errors on the build? |
| 19:27.24 | [ProB]CrazyMan | it seems so, didnt have an error |
| 19:28.19 | [TK]D-Fender | Just because you compiled it OK doesn't mean it won't bomb when trying to LINK to a version that isn't compatible |
| 19:29.19 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 19:30.36 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 19:30.41 | JerJer | it is the compile time options that are different… one version has a different flag set than the other version |
| 19:31.46 | JerJer | there is a way to figure out more info, i just can't remember how |
| 19:31.56 | [ProB]CrazyMan | ok so I comiled now asterisk ... |
| 19:32.09 | [ProB]CrazyMan | and lcr |
| 19:32.12 | [ProB]CrazyMan | still the same |
| 19:32.27 | JerJer | did you remove the old versions? |
| 19:32.41 | [ProB]CrazyMan | which old versions? |
| 19:32.45 | [ProB]CrazyMan | the src? |
| 19:32.49 | JerJer | installed by yum |
| 19:33.02 | [ProB]CrazyMan | no |
| 19:33.18 | *** join/#asterisk bmg505 (~leon@196.209.120.151) |
| 19:33.27 | JerJer | then the old versions of files could very well still be used |
| 19:34.37 | [TK]D-Fender | [ProB]CrazyMan, What part of "doesn't match the COMPILE TIME OPTIONS" was unclear? Doesn't matter it you rebuilt it fine. it doesn't MATCH the ones used for the REST of asterisk as you are running it. |
| 19:44.35 | ageis | how to transfer a call to a parking lot with a dialplan command? |
| 19:47.08 | [TK]D-Fender | ageis, Call "Park()" yourself or, use the auto-generated one based on our features.conf settings in the context you told it to use |
| 19:49.10 | ageis | [TK]D-Fender: what arguments does Park() take? |
| 19:49.39 | [TK]D-Fender | "core show application park" <-------- |
| 19:51.34 | martinlindhe | [2012-11-20 19:02:38] WARNING[14001] asterisk.c: The canary is no more. He has ceased to be! He's expired and gone to meet his maker! He's a stiff! Bereft of life, he rests in peace. His metabolic processes are now history! He's off the twig! He's kicked the bucket. He's shuffled off his mortal coil, run down the curtain, and joined the bleeding choir invisible!! THIS is an EX-CANARY. (Reducing priority) |
| 19:51.38 | martinlindhe | :-((( |
| 19:53.15 | *** join/#asterisk citywok (~kvirc@67-134-194-33.dia.static.qwest.net) |
| 19:59.02 | JerJer | hehe - i love it |
| 19:59.39 | Chainsaw | I didn't know Digium employed Brits :) |
| 20:00.05 | leifmadsen | I think that was written by Corydon76 ... |
| 20:00.12 | leifmadsen | who is not British :) |
| 20:00.13 | jmetro | Your Dahdi smelt of elderberries! |
| 20:01.40 | JerJer | martinlindhe: the way i understand things, if your system is setup to run with high priority and the astcanary process is unresponsive (?) Asterisk automatically lowers its priority so you have a chance to diagnose and hopefully correct the problem |
| 20:04.23 | Chainsaw | leifmadsen: An unusually well developed sense of humour then. |
| 20:05.21 | *** join/#asterisk keycruncher (~Adium@c-174-55-112-94.hsd1.pa.comcast.net) |
| 20:05.53 | JerJer | there are several funny nuggets within the code, comments and various other places :) |
| 20:06.17 | [ProB]CrazyMan | [TK]D-Fender: and how do I adjust the compile options to fit together ? |
| 20:06.39 | *** join/#asterisk n8ideas (~joshua@65.112.207.3) |
| 20:06.49 | JerJer | and in years past I remember some quite angry comments in chan_sip being made by Mark :) |
| 20:06.51 | [TK]D-Fender | [ProB]CrazyMan, Look at what was used. Use the same |
| 20:07.33 | [ProB]CrazyMan | [TK]D-Fender: and where do I see this by the yum ones ? |
| 20:07.59 | [TK]D-Fender | ARCH should be a clue... |
| 20:12.22 | dijib | anybody know what the Hierarchy Token Bucket package is in centos yum? |
| 20:12.58 | *** join/#asterisk ujjain (~ujjain@unaffiliated/ujjain) |
| 20:13.12 | *** join/#asterisk keycruncher (~Adium@c-174-55-112-94.hsd1.pa.comcast.net) |
| 20:13.37 | Qwell | ~google Hierarchy Token Bucket |
| 20:15.10 | dijib | htb and hfsc d the same thing? |
| 20:32.46 | *** join/#asterisk Hive (~Hive@173-165-205-1-jacksonville.hfc.comcastbusiness.net) |
| 20:35.08 | Hive | I'm using Fax for Asterisk for faxing(obviously). For some reason when an outbound fax fails for reason 'T1_TIMEOUT' or 'T2_TIMEOUT', asterisk keeps a channel open with Last Message: "BYE", and I get a bunch of "Autodestruct on dialog" flooding my CLI. The call goes through the Hangup Context so I'm not sure how I am supposed to deal with closing out such a channel automatically. Does anyone have any insight to shed? :D |
| 20:36.11 | Hive | From what I've gathered through google, typically the Autodestruct on dialog message comes from something taking too long to complete. However, like I said, the hangup context fires just fine and goes all the way to the end. |
| 20:42.14 | unicron | will it break anything if i have /etc/asterisk be full of symlinks? |
| 20:42.43 | Chainsaw | unicron: Mine is full of stub config files that just #include files further down. |
| 20:43.04 | jmetro | modular and object oriented programming is a + |
| 20:43.15 | unicron | oh, good thought |
| 20:49.34 | *** join/#asterisk k610 (~Instantbi@78.29.241.186) |
| 20:52.04 | mathis_ | yarks. JACK_HOOK seems to be completely b0rked |
| 20:52.38 | dijib | Hive: can you ~pb your fax extensions |
| 20:52.57 | dijib | also why not use digium free fax for asterisk |
| 20:53.21 | dijib | get out of here Seri |
| 20:56.28 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-azhslgamsgikfxns) |
| 20:58.12 | [ProB]CrazyMan | thi LCR drives me crazy .... it just take some configs ... I dont know from where it takes the config ... but its not where it should be ... |
| 20:59.05 | *** join/#asterisk vlad_starkov (~vlad_star@83.149.8.110) |
| 21:00.25 | Hive | dijib I am using Free Fax for Asterisk... but I bought more licenses so it's not free :P |
| 21:01.55 | Hive | http://pastebin.com/yWmws23E here's the context |
| 21:02.10 | Hive | and apparently it's not hitting the Hangup at the end of the context -_- |
| 21:02.38 | Hive | the CLI goes from |
| 21:03.35 | Hive | Executing [s@outboundfax:6] SendFAX("channel", "/tmp/faxes/1sdfes49.tif,df") in new stack to -- Executing [h@outboundfax:1] |
| 21:05.02 | [TK]D-Fender | Hive, Show us the complete call, not just bits and pieces |
| 21:06.48 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 21:07.29 | Hive | okay give me a minute, there are like 10 channels flooding me with warnings now lol |
| 21:15.01 | eject_ck | Hi, all. I'm using IPKALL for call forwarding. Anyone can help why ${CALLERID(num)} shows me 508 (this is account on my asterisk box)? |
| 21:16.32 | dijib | Hive: i have not worked with SendFAX only RecieveFax |
| 21:17.46 | [TK]D-Fender | eject_ck, Because that's what it is. |
| 21:18.39 | dijib | Hive: this is how im using mine. 1.0.1 currently running on swissarms (pid = 24490) |
| 21:18.42 | dijib | swissarms*CLI> |
| 21:18.44 | dfgas-cr48 | dijib, did you get it working correctly? |
| 21:18.55 | dijib | http://pastebin.com/mzK71KFR |
| 21:19.11 | dijib | no im still stuck on that bug so im waiting until the next release dfgas-cr48 |
| 21:19.30 | dfgas-cr48 | darn, heh |
| 21:19.31 | dijib | why must i always paste-fail in here |
| 21:19.44 | dijib | will only be a week dfgas-cr48 |
| 21:19.53 | dfgas-cr48 | ahh, cool |
| 21:20.09 | dfgas-cr48 | brb |
| 21:20.10 | dijib | its already at rc1 and has been for a couple weeks now |
| 21:34.31 | *** join/#asterisk Tarang (u7226@gateway/web/irccloud.com/x-iegzipgpygsbqzlg) |
| 21:34.58 | Tarang | Hi guys, I have a problem transcoding ulaw<->ilbc due to the 20ms frame size issue is there any solution to this? |
| 21:35.10 | Qwell | what 20ms frame size issue? |
| 21:35.35 | Tarang | Asterisk can only make 30ms iLBC frames, even if told to make a 20ms frame |
| 21:35.44 | Qwell | and? |
| 21:35.55 | Tarang | it results in choppyness in the sound on one end |
| 21:35.59 | Tarang | and the other is ok |
| 21:36.18 | Tarang | so is there something I can do to fix the choppyness? |
| 21:36.50 | Qwell | It won't spit out a frame until it has enough to do so. You should just be a frame behind. Which side is choppy? |
| 21:37.04 | Tarang | i found this http://forums.digium.com/viewtopic.php?p=17732 but its over 8 years old now |
| 21:37.35 | Tarang | My end will hear their voice choppy, the terminating end hears me perfectly |
| 21:37.50 | Tarang | so iLBC->ulaw is ok, but ulaw->iLBC is a bit of an issue |
| 21:38.04 | Qwell | and what version of Asterisk is this with? |
| 21:38.20 | Tarang | Asterisk 1.8.18.0 |
| 21:38.26 | WIMPy | Did you patch it to do 20ms as in that article? |
| 21:38.36 | WIMPy | Or are you using it with 30ms? |
| 21:38.48 | Tarang | i didn't want to mess with the code until it was a last resort |
| 21:39.05 | Tarang | so its unpatched and at 30ms |
| 21:39.21 | Qwell | both sides are SIP? |
| 21:39.24 | WIMPy | :-( |
| 21:39.28 | Tarang | yup both sides are SIP |
| 21:39.37 | Qwell | and you have a jitter buffer enabled? |
| 21:39.45 | Tarang | i don't believe i have that enabled |
| 21:39.50 | Tarang | how would I do that? |
| 21:40.11 | Tarang | i've heard of it but didn't think it would help? |
| 21:43.11 | Tarang | let me give it a shot and see |
| 21:44.41 | Tarang | my phone had a jb already it seems |
| 21:44.55 | *** join/#asterisk amessina (~amessina@2001:470:c1dc:7779:d6be:d9ff:fe8d:7c1e) |
| 21:45.57 | slav3_kitten | hey p3nguin can you try placing a voice call to that uri i gave you earlier? |
| 21:46.41 | dijib | slav3_kitten: soup? |
| 21:46.59 | slav3_kitten | soup? |
| 21:47.05 | dijib | wa'soup? |
| 21:47.22 | slav3_kitten | huh? |
| 21:47.33 | WIMPy | short for warm soup? |
| 21:47.49 | slav3_kitten | i think he's asking what's up |
| 21:47.50 | dijib | lol |
| 21:47.52 | slav3_kitten | like he's an asian canadian |
| 21:47.54 | dijib | slav3_kitten: http://pastebin.com/40Wyx5Ya |
| 21:47.55 | *** join/#asterisk vlad_starkov (~vlad_star@83.149.8.115) |
| 21:48.00 | dijib | wound-eye |
| 21:48.02 | slav3_kitten | fresh off the boat |
| 21:48.05 | WIMPy | 'd prefer the soup |
| 21:48.09 | *** join/#asterisk jsjc (~Adium@54.Red-83-35-54.dynamicIP.rima-tde.net) |
| 21:48.25 | dijib | sry should have PVT'd that |
| 21:48.37 | slav3_kitten | dijib, you're an asshole... |
| 21:48.50 | WIMPy | dijib: Too many IPs for ICE? |
| 21:49.03 | slav3_kitten | anyhow p3nguin was able to play me a sound file over sip today |
| 21:49.03 | dijib | i think its an issue with 11.0.1 |
| 21:49.16 | dijib | ok then you have it wrking and im broken |
| 21:49.22 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 21:49.36 | dijib | i can connect to EXTEN@dnsdyn.domain.com but not @IP |
| 21:49.37 | jaytee | anyone here use Bandwidth.com (Phonebooth)? |
| 21:50.01 | Robotman321 | and rrittgarn and jmetro do |
| 21:50.25 | dijib | Hive: whats-a-matter-for-you? |
| 21:53.53 | dijib | should i patch my 11.0.1 to 11.1.0-rc1 ? |
| 21:54.03 | dijib | or wait? |
| 21:55.00 | jaytee | The email for a new Bandwidth.com account details setting up a VoIP system to use their "trunks". In the email there is this:• Port 5060 UDP must be open for SIP. |
| 21:55.00 | jaytee | • Ports 1024 - 64000 UDP should be open for RTP Media/Audio. |
| 21:55.00 | jaytee | Isn't the RTP audio port range just a bit over the top? Do I really have to allow that many UDP ports open on the client's firewall? |
| 21:55.17 | WIMPy | dijib: Or just disable ICE. |
| 21:55.42 | dijib | WIMPy: this will not help my desire to have google voice working |
| 21:56.05 | WIMPy | dijib: You can disable it in chan_sip. |
| 21:56.55 | WIMPy | jaytee: The local ports are those configured in rtp.conf. But on a decent firewall you might not need to configure them at all. |
| 21:58.01 | dijib | WIMPy: in sip.conf? |
| 21:58.29 | WIMPy | yes. |
| 21:58.37 | dijib | [general]? |
| 21:59.07 | WIMPy | Probably. I've disabled it in rtp.conf. |
| 21:59.14 | dijib | call still failes but i dont have the extend errors |
| 21:59.26 | dijib | its still enabled in rtp.conf |
| 22:00.49 | dijib | slav3_kitten: howcome you dont have a dyndns for me? |
| 22:01.34 | jaytee | WIMPy, thanks for the input. |
| 22:01.52 | slav3_kitten | dijib, because i don't have one |
| 22:02.06 | dijib | well get one... makes it easier. |
| 22:02.23 | slav3_kitten | god i'm an idiot :| |
| 22:03.29 | slav3_kitten | just spent like 3 hours trying to debug why the fucking hell my outbound dialing worked in one context an not the other. because i mistyped the dial prefix for use with static ip outbound dial |
| 22:03.29 | *** join/#asterisk SeRi (~wtf@pdpc/supporter/professional/seri) |
| 22:03.43 | SeRi | waz up guys |
| 22:03.50 | slav3_kitten | sup SeRi |
| 22:04.00 | jaytee | slav3_kitten, know the feeling. hate it when that happens |
| 22:04.07 | WIMPy | soups's up |
| 22:04.12 | slav3_kitten | just had a derp moment myself from mistyping |
| 22:04.29 | dijib | hey SeRi |
| 22:04.33 | SeRi | WIMPy: share. |