| 00:00.23 | *** join/#asterisk fstd (~fstd@unaffiliated/fisted) |
| 00:01.43 | dan_j | Weird. A reboot seems to have sorted it. |
| 00:20.01 | *** join/#asterisk fury_ (~fury@znc.hq.codexterous.com) |
| 00:21.58 | kb3ien | Darn. Looking for the explanation of the Marked Mode in ConfBridge... |
| 00:22.59 | kb3ien | does Set(CONFBRIDGE(user,end_marked)=yes) mean when the last Admin leaves I get booted? |
| 00:28.49 | kb3ien | meetme had this: x - Leave the conference when the last marked user leaves. does this still exist in ConfBridge ? In what form ? |
| 00:30.42 | *** join/#asterisk carrar (~tim@osburn.jp) |
| 00:34.06 | *** join/#asterisk bmurt (~brendan@208-58-116-232.c3-0.upd-ubr1.trpr-upd.pa.cable.rcn.com) |
| 00:34.36 | kb3ien | It should. > https://wiki.asterisk.org/wiki/display/AST/ConfBridge+10?src=search |
| 00:35.10 | *** join/#asterisk Dovid (~Dovid@ool-2f113961.dyn.optonline.net) |
| 00:56.53 | kb3ien | Well the Options MAY work, but Dial(,,G) doesn't pass my variables to the n+1 priority so NUTS TO ME. |
| 00:58.35 | kb3ien | its a known issue, but I can't say I like it. |
| 00:58.41 | kb3ien | http://forums.asterisk.org/viewtopic.php?f=13&t=88570 |
| 01:05.22 | kb3ien | I think that there should be an inherit option to extends G and U, but that's me. |
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| 01:17.28 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 02:33.51 | kb3ien | Well the ConfBR is almost working in lieu of MeetMe, but still no video. How does one proceed debugging confBridge ? |
| 03:10.40 | *** join/#asterisk raspberrypifan (~raspberry@186.42.33.108) |
| 03:18.29 | *** join/#asterisk jameswf (uid27319@gateway/web/irccloud.com/x-zrkgiixfqlckjgzs) |
| 03:23.16 | kb3ien | dtmf_passthrough=yes isn't parsed properly.... odd... |
| 03:33.07 | kb3ien | Well MeetMe didn't do video, and now ConfBridge can't do it if use use Dial(,,G()) to get there... |
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| 04:38.47 | kb3ien | ast_openstream_full: File pbx-transfer does not exist in any format oh, really? ls can find it. Where is it looking, i wonder... |
| 05:01.06 | *** join/#asterisk raspberrypifan (~raspberry@186.42.33.108) |
| 05:01.30 | raspberrypifan | is elastix still used |
| 05:06.39 | [TK]D-Fender | I'm sure someone is still using that distro |
| 05:06.55 | raspberrypifan | but in general? |
| 05:07.21 | [TK]D-Fender | Yes, many people still use it. |
| 05:07.33 | *** join/#asterisk hos7ein (~chatzilla@94.101.189.1) |
| 05:07.50 | raspberrypifan | interesting |
| 05:10.45 | kb3ien | pbx-transfer not found,but where is it looking. I don't feel like strace the whole of asterisk... |
| 05:11.59 | kb3ien | Annouce Template:pbx-transfer:PARKED |
| 05:12.09 | kb3ien | Announce:pbx-transfer |
| 05:12.25 | kb3ien | ast_openstream report it does not exist in any format... |
| 05:13.39 | [TK]D-Fender | "core show settings" |
| 05:13.56 | [TK]D-Fender | VarLib + /sounds /(language) |
| 05:14.11 | [TK]D-Fender | Did you install sounds files with *? |
| 05:15.02 | [TK]D-Fender | Are you loading FORMATS so that * can read in those files? What about CODECs so the it can transcode if they aren't matching the one used by the channel? |
| 05:15.40 | kb3ien | yep, /var/lib/asterisk/ is VarLib directory |
| 05:17.04 | kb3ien | I have /var/lib/asterisk/sounds/fr/pbx-transfer.alaw ; the very file It cannot find. |
| 05:17.13 | *** join/#asterisk gerhard7 (~gerhard7@77-172-20-151.ip.telfort.nl) |
| 05:17.19 | [TK]D-Fender | We aren'te seeing the full call.... |
| 05:17.32 | [TK]D-Fender | PASTEBIN is your friend.... |
| 05:27.14 | kb3ien | <PROTECTED> |
| 05:30.48 | [TK]D-Fender | Show us the files as well... |
| 05:30.53 | [TK]D-Fender | and your modules.conf |
| 05:30.58 | kb3ien | :/var/lib/asterisk/sounds# find . | grep pbx-transfe | grep alaw |grep fr |
| 05:31.01 | kb3ien | ./fr/pbx-transfer.alaw |
| 05:32.57 | kb3ien | what's in modules.conf ? |
| 05:33.10 | [TK]D-Fender | go show us |
| 05:33.35 | kb3ien | I updated the one pastebin. |
| 05:35.06 | [TK]D-Fender | It gives you a new link.,.. |
| 05:35.28 | kb3ien | http://pastebin.com/mPJrpMVr |
| 05:35.30 | kb3ien | indeed so. |
| 05:37.06 | [TK]D-Fender | now lets try REALLY looking at the file.... |
| 05:37.31 | [TK]D-Fender | ls -la /var/lib/asterisk/sounds |
| 05:39.10 | kb3ien | http://pastebin.com/GeMXN9YC |
| 05:39.52 | [TK]D-Fender | those folders being owend by ROOT looks like a very bad idea |
| 05:40.03 | [TK]D-Fender | Considering Asterisk isn't running as root. |
| 05:40.14 | [TK]D-Fender | Fix your ownership |
| 05:40.30 | kb3ien | readable and executeable. should work x is like r on folders. |
| 05:40.53 | [TK]D-Fender | Fix them |
| 05:41.11 | [TK]D-Fender | And go look at the files in those folders |
| 05:43.45 | *** join/#asterisk W0rmDr1nk (~wormdrink@unaffiliated/wormdrink) |
| 05:44.56 | kb3ien | no change. |
| 05:45.32 | kb3ien | chown asterisk:asterisk pbx-transfer.* assuming no write needed. |
| 05:45.39 | kb3ien | actually it got write anyway. |
| 05:46.21 | [TK]D-Fender | Go show the fully corrected paths |
| 05:46.40 | [TK]D-Fender | And set core debug to 5 so we can prove exactly what i'ts looking for |
| 05:52.25 | kb3ien | ls -l $(locate fr/pbx-transfer.alaw) |
| 05:52.34 | kb3ien | -rw-r--r-- 1 asterisk asterisk 8308 Oct 24 2013 /var/lib/asterisk/sounds/fr/pbx-transfer.alaw |
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| 06:02.20 | *** join/#asterisk AAmit (~amit@1.39.15.44) |
| 06:03.11 | [TK]D-Fender | Go verify that you have to proper format & codec modules present and loaded |
| 06:03.33 | kb3ien | well nuff of this for one night. |
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| 08:55.37 | al_nz1 | What files should I look at to check if TCP is enabled for asterisk and for a extension? |
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| 12:27.06 | Tuju | what is the difference in line settings between 'domain' and 'fromdomain' ? |
| 12:32.24 | [TK]D-Fender | first matching INCOMING, the other is OUTGOING (goes into the "From:" header) |
| 12:33.03 | sekil | btw how can one set the domains in invite r-uri and to field w/o using host? |
| 12:33.14 | sekil | that is ...when set to host it tends to resolve |
| 12:33.23 | sekil | which sometimes is not desirable |
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| 12:51.40 | Tuju | ack |
| 12:52.11 | Tuju | i understood, that domain= is typically set in general level and fromdomain= in each line settings. |
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| 13:07.02 | Tuju | for some reason, one of my trunks sends line@1.2.3.4 instead of line@example.com. |
| 13:07.11 | Tuju | and hence registration fails. |
| 13:08.06 | kb3ien | I'm back lets see if I can debug that pbx-transfer.ulaw today! |
| 13:11.09 | [TK]D-Fender | Tuju: First, stop using the term "trunk". Second, "register =>" lines have nothing to do with any peer entry you may have. Those use [general] settings ONLY. |
| 13:12.00 | [TK]D-Fender | Tuju: There is a directive to use peer settings in 11+ instead of a sparate REGISTER" directive |
| 13:12.04 | Tuju | i understood that the latter part, example.com/plaaa <---- plaa connects to line name. |
| 13:12.11 | [TK]D-Fender | no |
| 13:12.20 | Tuju | i read it somewhere. |
| 13:12.26 | [TK]D-Fender | that sets the return CONTACT |
| 13:12.45 | [TK]D-Fender | You read wrong or they simply had no clue what they're talking about. |
| 13:12.57 | Tuju | that's the result that we don't have correct instructions nor documentation. |
| 13:13.23 | Tuju | there are zillion sites that publish their version of configuration. |
| 13:13.54 | Tuju | i also wiped out most of my sip.conf comments as those appeared to be from old version. |
| 13:14.30 | Tuju | nor there was no descriptions of those line settings regardless that people here claim that that's the only authorative source. |
| 13:15.02 | *** join/#asterisk jameswf (uid27319@gateway/web/irccloud.com/x-sfrynrxdjgrsxeto) |
| 13:15.28 | [TK]D-Fender | [09:13]Tujui also wiped out most of my sip.conf comments as those appeared to be from old version. <- the SAMPLE configs that * provides are documentation. |
| 13:15.29 | *** join/#asterisk madax (~madax@c42-156.i07-11.onvol.net) |
| 13:15.41 | Tuju | i thought that was the sample. |
| 13:16.07 | sekil | sample files are documented |
| 13:16.07 | [TK]D-Fender | [09:13]Tujuthere are zillion sites that publish their version of configuration. <- What about *'s OFFICIAL wiki? Or at least the old voip-info one (with caution)? |
| 13:16.22 | sekil | I think that's what he's referring to |
| 13:17.18 | Tuju | http://fpaste.org/135434/14113917/ centos 6.0 |
| 13:17.35 | Tuju | i don't call that documentation. |
| 13:17.47 | sekil | Tuju: host should provide you the domain part for invite r-uri and To-uri |
| 13:17.59 | sekil | Tuju: host directive |
| 13:18.00 | [TK]D-Fender | Tuju: that is not a complete sample config |
| 13:18.05 | kb3ien | seems that fileexits_core got some 'splaining to do. |
| 13:18.10 | Tuju | sekil: i try that |
| 13:19.37 | sekil | Tuju: also there's outboundproxy to use if needed |
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| 13:21.17 | [TK]D-Fender | Tuju: Sample config seems to offer a direct options for setting the domain. |
| 13:21.35 | [TK]D-Fender | realm would require a peer for sure |
| 13:21.47 | [TK]D-Fender | (with a built-in register driective) |
| 13:23.16 | madax | Hi all, i have a little question please. I have configured an atd and have recorded some greeting message. I know i'm supposed to convert them in 8Khz, 16 bit mono file, but my question is it possible to use a better quality format ? |
| 13:24.05 | sekil | [TK]D-Fender: can realm be set in peer/friend? |
| 13:24.52 | [TK]D-Fender | madax: set it to match whatever your call is going to actually use |
| 13:25.47 | [TK]D-Fender | sekil: sekil Actually that appears to be global only there |
| 13:26.08 | madax | [TK]D-Fender: you mean like alaw or ulaw ? |
| 13:26.21 | sekil | [TK]D-Fender: so only host is an option in peer/friend for domain part on invite ? |
| 13:26.24 | [TK]D-Fender | madI mean like precisely the codec your call is going to be using. |
| 13:26.44 | Tuju | wiki fails by using such vocubulary that expected audience is not familar yet. |
| 13:26.49 | [TK]D-Fender | madax: I mean like precisely the codec your call is going to be using. |
| 13:26.59 | sekil | Tuju: what is not working for you? |
| 13:27.15 | sekil | Tuju: you need the invite to be 101@blah.com rather than 101@1.2.3.4? |
| 13:27.26 | madax | ah ok, let me try that. |
| 13:27.31 | sekil | Tuju: or the REGISTER |
| 13:27.34 | Tuju | sekil: i'm talking about wiki now. |
| 13:27.41 | Tuju | but yes, i've that problem you mentioned. |
| 13:27.50 | Tuju | in OPTIONS and REGISTER |
| 13:28.32 | [TK]D-Fender | [09:26]Tujuwiki fails by using such vocubulary that expected audience is not familar yet. <- the sample config explains registration options in full detail |
| 13:28.58 | [TK]D-Fender | [09:11]Tujui understood that the latter part, example.com/plaaa <---- plaa connects to line name. <- including this |
| 13:29.02 | sekil | Tuju: register => udp://123456@ims.blah.com:secret:123456@ims.blah.com@10.0.0.2:5060/123456 |
| 13:29.28 | Tuju | sekil: i try that |
| 13:29.34 | sekil | Tuju: this is an unusual example as the auth name required is user@domain rather than user |
| 13:29.45 | sekil | Tuju: so you can change that |
| 13:30.04 | sekil | Tuju: and there's outboundproxy as 10.0.0.2 set here |
| 13:30.45 | Tuju | sekil: i had register => 1234:pwd@example.com/office-trunk |
| 13:31.04 | Tuju | and that example.com got changed to 1.2.3.4 ip-adress in From: field. |
| 13:31.45 | sekil | Tuju: register => [peer?][transport://]user[@domain][:secret[:authuser]]@host[:port][/extension][~expiry] |
| 13:34.31 | [TK]D-Fender | Use a peer entry to do your registration instead of a separate "register" directive if you want more control. |
| 13:35.35 | madax | [TK]D-Fender: it works thanks a lot |
| 13:35.49 | [TK]D-Fender | madax: You're welcome |
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| 13:40.24 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:46.23 | kb3ien | So asterisk is looking for sounds in /usr/share/asterisk/sounds/ but installed them in /var/lib/asterisk/sounds/ not sure why... |
| 13:49.55 | [TK]D-Fender | asterisk.conf <-------- |
| 13:51.19 | kb3ien | which directive sets that? |
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| 13:51.58 | [TK]D-Fender | varlib |
| 13:53.10 | kb3ien | that IS EVEN MORE STRANGE astvarlibdir => /var/lib/asterisk NOT /usr/share/asterisk |
| 13:53.48 | [TK]D-Fender | Go check your startup script..... |
| 13:54.11 | [TK]D-Fender | Perhaps it's not even pointing to the config folder you think it is... |
| 13:54.17 | kb3ien | maybe... |
| 13:54.52 | kb3ien | I did take a script I've been upgrading since 1.2 and try to pull it up to 11. It may not be that compatible anymore. |
| 13:55.37 | kb3ien | I build 1.2 on netbsd/ppc in like 2007 |
| 13:59.27 | Katty | oh man |
| 13:59.35 | Katty | i'm beat. |
| 13:59.40 | Katty | drmessano: ping |
| 13:59.44 | kb3ien | Well I symbolically linked around it for now, <<TRANSFER SEPT ZERO UN>> it works! Ca marche! |
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| 14:03.52 | *** join/#asterisk david______ (515ecce2@gateway/web/freenode/ip.81.94.204.226) |
| 14:04.01 | david______ | hi |
| 14:04.48 | david______ | I am trying to play a message using early media |
| 14:05.33 | david______ | same => n,Playback(ivr,noanswer) |
| 14:06.05 | david______ | but issue is incoming call got hanged up after a minute |
| 14:06.28 | *** join/#asterisk sgriepentrog (~sgriepent@nat/digium/x-qujqstyardadrbmk) |
| 14:06.33 | david______ | Is any way to play an audio file which is one hour long |
| 14:06.39 | [TK]D-Fender | david______: Go look at the call to see why |
| 14:08.11 | david______ | [TK]D-Fender: Not come on cli. Call simply got hangup |
| 14:08.33 | [TK]D-Fender | If you are looking at the call properly you will see why |
| 14:08.52 | david______ | actually i want to play file as ringtone |
| 14:09.10 | [TK]D-Fender | "playback" is not a ring-tone. |
| 14:09.31 | david______ | playback with noanswer parameter |
| 14:09.44 | [TK]D-Fender | Still not a ring-tone |
| 14:09.50 | *** join/#asterisk Quastor (~AndChat14@dD5778AB4.access.telenet.be) |
| 14:10.02 | david______ | or any other expert openion |
| 14:10.12 | [TK]D-Fender | This isn't an opinion. This is a fact |
| 14:10.41 | [TK]D-Fender | davAnd you aren't adding anything more to the description for the overall task you are attempting to accomplish |
| 14:10.56 | Quastor | Hi is there a way to send the caller id when using a call pickup |
| 14:11.24 | Quastor | Or blf |
| 14:11.38 | *** join/#asterisk sgriepentrog (~sgriepent@nat/digium/x-paacbegrirvhcqeu) |
| 14:12.12 | david______ | [TK]D-Fender: my plan is to play an ivr of an hour on which no one bear cost not caller nor callee |
| 14:12.36 | [TK]D-Fender | david______: Go look at the call and see why it is disconnecting then. |
| 14:12.57 | [TK]D-Fender | david______: That is still not a "ring-tone" that term in not appropriate for the description you've given. |
| 14:13.12 | *** part/#asterisk sgriepentrog (~sgriepent@nat/digium/x-paacbegrirvhcqeu) |
| 14:13.49 | david______ | want to send this ivr file as a early media. I am able to play an ivr for one minute |
| 14:14.24 | [TK]D-Fender | david______: Go look at the call. |
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| 14:15.39 | *** join/#asterisk anonymouz666 (~anonymouz@189-25-34-104.user.veloxzone.com.br) |
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| 14:18.56 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:18.56 | Tuju | i got it working. |
| 14:19.05 | Tuju | is there something i could use presence for? |
| 14:19.22 | Tuju | it sounds useful, but i'm wondering does my devices support it anyhow. |
| 14:19.34 | Tuju | do my devices... |
| 14:21.11 | [TK]D-Fender | It does what it says it does. It reports the satates of your other device definitions in *, as well as other things like queue status, custom values, etc. |
| 14:22.21 | Tuju | i was wondering could i somehow show my 'existence' next to phone like some instant message systems show |
| 14:22.40 | [TK]D-Fender | does that as well. |
| 14:22.59 | Tuju | any idea do cisco phones support it anyhow? |
| 14:23.15 | [TK]D-Fender | If you have a phone watching another peer, it'll report it as "not registered", "idle", "on call", "ringing", etc |
| 14:23.27 | [TK]D-Fender | Tuju: Depends on the model & usage |
| 14:23.49 | Tuju | i've got plenty of them :) the one next to me is c7975 with sip firmware. |
| 14:24.15 | Tuju | but that certainly sounds useful. |
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| 14:24.42 | Tuju | ideally it would have some minuteman-dead-man's-switch :) |
| 14:25.43 | [TK]D-Fender | I know 7940/60's don't support BLF with SIP. Not sure on those ones. I'd go search it up |
| 14:26.53 | *** join/#asterisk darkdrgn2k (~darkdrgn2@209.90.253.66) |
| 14:27.18 | darkdrgn2k | Hello all, any one know of a way i can prevent Asterisk from intercepting and regenerating DTFM tones (reinvite is not an option_ |
| 14:29.23 | Tuju | [TK]D-Fender: i've those too, couple 7960's - a great workhorse if you have light to watch its display. and very nice to debug thanks to its software. |
| 14:30.07 | Tuju | i need to try it in near future, like MWI etc, even i don't like messages. |
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| 14:52.36 | darkdrgn2k | is core a dynamic module by any chance? |
| 14:54.21 | pabelanger | darkdrgn2k, where do you see core? |
| 14:54.29 | pabelanger | on the filesystem? |
| 14:55.03 | darkdrgn2k | hmm good point |
| 14:55.12 | darkdrgn2k | i need to modify ./main/dsp.c |
| 14:55.17 | darkdrgn2k | thats proably the main exec |
| 14:56.54 | darkdrgn2k | does asterisk detect all dtfm tones or is there a way to change the "country" |
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| 15:10.48 | sekil | Tuju: working now? |
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| 15:28.13 | WIMPy | darkdrgn2k: That depends on the technology used and also on configuration. |
| 15:41.32 | *** join/#asterisk iulhk (~iulhk@182.189.96.66) |
| 15:41.50 | iulhk | i am using asterisk-11 , focus just sip to sip call, audio/video, in allowed codecs 'gsm;h264' , Issue peer 1000 calling to 2000 audio call, after ack, at receiver end video screen automatically opened, although caller started audio call, if i just enabled gsm, codec then both-end its audio screen, but in that case i can't start video as video codec not enabled. i want when user starting |
| 15:41.51 | iulhk | video call, it should start video at both-end when user starting audio call then at both end it should be audio call, is it possible ? |
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| 16:14.05 | madax | guys, i have mapped the Queue function to an exten (100).It also launches a script via AGI. I'd like to send as argument the number of the extension who finally hangs up and not 100. Anyone knows how to do that ? |
| 16:16.08 | [TK]D-Fender | madYour description is confusing. Show us what you have created alrady |
| 16:16.15 | [TK]D-Fender | madax: Your description is confusing. Show us what you have created alrady |
| 16:19.51 | Penguin | Extensions don't call others, phones do. So the only extension involved here is the one that runs your application. |
| 16:20.55 | Penguin | If you want to know the extension number associated with the phone calling extension 100, use the caller ID number. This assumes the callerID number is configured in a sensible way. |
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| 16:52.37 | madax | i basically do that in the extensions.conf |
| 16:52.51 | madax | exten => 100,1,Verbose(1,Caller ${CALLERID(all)} entering the "operateur" queue) |
| 16:53.11 | madax | same => n,Queue(operateur,rRtTxXwW,,,120,"test.php,${CALLERID(num)},${ARG2}") |
| 16:53.40 | madax | my question is for ${ARG2} |
| 16:53.58 | madax | if i put ${EXTEN} i will have --> 100 |
| 16:54.41 | madax | but i want the real number of the telephone who hangs up |
| 16:56.02 | [TK]D-Fender | madax: Queue(queuename[,options[,URL[,announceoverride[,timeout[,AGI[,macro[,gosub[,rule[,position]]]]]]]]]) |
| 16:56.10 | [TK]D-Fender | madax: So far I see no option for passing parameters in the first palce |
| 16:56.28 | [TK]D-Fender | madax: Next, that gets run right as the call is CONNECTED... not once it is HUNG UP |
| 16:56.40 | [TK]D-Fender | madax: You don't seem to be thinking clearly in terms of timing |
| 16:57.21 | [TK]D-Fender | madax: And you have also not showed us what kind of members you are using. |
| 16:59.22 | madax | in queues.conf: member => SIP/101 |
| 16:59.22 | madax | member => SIP/102 |
| 16:59.54 | [TK]D-Fender | madax: You still have all those other issues to answer first... |
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| 17:41.53 | drmessano | [12:19:51] <Penguin> Extensions don't call others, phones do. <-- False, people do |
| 17:42.51 | WIMPy | No, they just tell their phones. |
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| 18:00.53 | Qwell | No, they tell their phones to ask the PBX to. |
| 18:01.39 | WIMPy | You need a PBX to do it? |
| 18:09.58 | madax | i'm trying to identify who has taken the call on a queue, and launch a macro (or a script AGI) with some args based on who takes the call |
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| 18:10.30 | madax | is it possible to give args to "membermacro" ? |
| 18:11.13 | [TK]D-Fender | madax: use LOCAL CHANNELS as members and do the dial yourself and pass it there. |
| 18:16.39 | aruntomar | i need to design a solution/architecture that could dial in excess of 7600 calls per minute. i've been using asterisk for couple of years but most of them are small & medium call centers. need suggestions/pointers while designing such a redundant solution from our community experts. |
| 18:17.42 | madax | [TK]D-Fender : ok i will do that, thanks (i can't right now..) |
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| 18:33.33 | *** join/#asterisk Elv1313 (~etudiant@2607:fad8:4:6:a6ba:dbff:feba:26d8) |
| 18:34.39 | Elv1313 | Hello, I am trying to use TLS. In wireshark, I see something that look like a full TLS handshake. However, both client and asterisk then timeout and nothing appear in rasterisk (I have sip set debug on) |
| 18:48.14 | *** join/#asterisk ThoMe (~tm@tm.muc.de) |
| 18:48.15 | ThoMe | hiho |
| 18:48.24 | ThoMe | have when i reboot my firewall the follow problem |
| 18:48.25 | ThoMe | [Sep 22 20:47:39] NOTICE[4173]: chan_sip.c:15067 sip_reg_timeout: -- Registration for '777z3bugxj@fpbx.de' timed out, trying again (Attempt #26) |
| 18:48.28 | ThoMe | i use asterisk 11.12.1 |
| 18:48.53 | [TK]D-Fender | ThoMe: You're not getting an answer |
| 18:49.10 | ThoMe | and now i cant register my asterisk at my trunk |
| 18:49.24 | ThoMe | [TK]D-Fender: hm. is this problem at my sip provider or me? |
| 18:50.26 | ThoMe | this problem is only if I reboot/redial my firewall (also my dsl interface= |
| 18:50.30 | [TK]D-Fender | ThoMe: That message proves nothing as to why. So far you aren't even really looking at what you're sending. |
| 18:50.44 | *** join/#asterisk gusto (~gusto@2a02:810d:8640:1dd0:221:6aff:feb8:e0b2) |
| 18:50.47 | [TK]D-Fender | ThoMe: You've giving us nothing to go on |
| 18:50.53 | [TK]D-Fender | given* |
| 18:51.11 | Chainsaw | Your offering does not please the fender. |
| 18:51.19 | ThoMe | [TK]D-Fender: which infos you need for help me? |
| 18:51.21 | Chainsaw | It requires unadulterated, complete logs on pastebin |
| 18:51.39 | Chainsaw | You may not provide snippets without context. |
| 18:52.00 | [TK]D-Fender | ThoMe: You aren't even bothering to look at the SIP debug of your registration attempt to prove where it is trying to contact, and what you are sending them for retrun adress, etc.... |
| 19:00.32 | ThoMe | [TK]D-Fender: hmm. ok |
| 19:01.03 | ThoMe | [TK]D-Fender: http://pastebin.com/raw.php?i=7zxNaG08 sip debug. |
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| 19:02.08 | ThoMe | 192.168.100.4 = my asterisk, 82.135.63.218 = my router, dsl box and 62.134.52.212 is my trunk sip provider |
| 19:02.24 | [TK]D-Fender | ThoMe: Go double check any port forwarding you should have, make sure it's hitting whatever routers/firewalls it needs to along the way, and that the IP is sending them in your Contact: is accurate |
| 19:02.44 | [TK]D-Fender | ThoMe: And while you're at it ... that the IP you are sending to is proper |
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| 19:38.27 | woopstar | Hi People. So, I'm testing the WebRTC to Asterisk thingy. Got sipml5 to connect through WSS to Asterisk. Setup DTLS succesfully and I'm able to do a call. But there is no audio at all, and the call is disconnected with "for lack of RTP activity in 11 seconds", because rtptimeout is set to 10. So it seems there is a problem with RTP ? |
| 19:42.24 | woopstar | is there a specific codec i need to use maybe? :) |
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| 20:23.08 | mjordan | woopstar: which version of Asterisk? |
| 20:23.35 | woopstar | mjordan: asterisk-11.12.1 |
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| 20:26.54 | mjordan | Hm. |
| 20:27.11 | mjordan | Well, if you're getting an RTP timeout, it would be because RTP isn't able to make it from the browser to Asterisk and vice versa. |
| 20:27.27 | mjordan | Which means, despite the best efforts of ICE, a media path couldn't be found. |
| 20:28.12 | woopstar | For that we can agree. But I just don't get it. As all UDP ports are forwarded... But maybe it's a client issue... |
| 20:29.32 | mjordan | possibly. If you're just looking to 'play around with it', I'd make the puzzle as simple as possible and do it without a NAT in the way. ICE/STUN/TURN do the best they can to get through NATs, but there's no guarantee that they will be able to punch through everything. |
| 20:30.17 | woopstar | Yeah. I guess i'll put the asterisk server directly to the internet to prevent nat on the server side... preventing nat on client side is much more of a hazzle, right? |
| 20:34.13 | mjordan | NAT is pretty much just always a pain. Generally, configured correctly, ICE/STUN/TURN should find a way through. You may just need to tweak the settings some |
| 20:38.37 | woopstar | Yeah... Thing is.. Seems like it's the RTP connection the the sip provider... It initiates a RTP connection to them, but times out... |
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| 21:07.55 | *** join/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
| 21:08.51 | marceloamorim | hello, there is anyway to send the ${DIALSTATUS} to another channel? |
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| 21:10.06 | marceloamorim | I`m using the app Dial with option F(context^exten^priority) |
| 21:19.52 | Penguin | Why would you need to send the DIALSTATUS of one channel to another channel? |
| 21:20.50 | marceloamorim | Because I`m trying to build my own cdr, and so far I couldn`t do so much when the caller hang up the call |
| 21:21.46 | marceloamorim | I use the option g, and I`m fine when the called phone hang up, but when the caller hangup the best I could do so far is using the F option |
| 21:22.29 | Penguin | Did you consider the h exten and/or hangup handlers? |
| 21:24.46 | marceloamorim | well, I didn`t consider, but H is for allow a dtmf sequence |
| 21:25.13 | Penguin | h exten! Not H option for Dial() application. |
| 21:26.25 | marceloamorim | uhmm, let me check, I thought that options was for press some buttons and this buttons hangup the call |
| 21:31.03 | marceloamorim | Penguin: that featuremap for disconect? |
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| 21:31.27 | Penguin | What part of EXTEN are you having trouble understanding? |
| 21:31.35 | WIMPy | He said extension. Read about special extensions. |
| 21:32.20 | Penguin | You're probably going to want a hangup handler, so read about that as well. |
| 21:32.38 | marceloamorim | ok |
| 21:36.13 | marceloamorim | sorry, I was thinking about this " h: Allow the called party to hang up by sending the DTMF sequencedefined for disconnect in "features.conf". " |
| 21:36.33 | Penguin | I specifically told you no. |
| 21:36.40 | Penguin | I specifically told you the h extension. |
| 21:36.59 | Penguin | I specifically told you not the h option to the Dial() application. |
| 21:37.46 | marceloamorim | I realized that now |
| 21:47.43 | marceloamorim | thx guys, that was what I needed |
| 21:48.01 | marceloamorim | cooool \o/ yay |
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