| 00:00.00 | kc8pxy | how do i kick sip peers out? |
| 00:00.04 | kc8pxy | DOH |
| 00:00.07 | kc8pxy | :) |
| 00:03.38 | *** join/#asterisk Mhaddog (~Mhaddog@z65-50-118-232.ips.direcpath.com) |
| 00:04.55 | kc8pxy | is it normal for nat'd sip channels to show in sip sho peer channelname as having an addr->IP showing as undetermined? |
| 00:05.24 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 00:09.05 | xheliox | Undetermined or unspecified? |
| 00:09.14 | xheliox | And yes, if it says unspecified, that's normal. |
| 00:09.35 | xheliox | It would only display if you had the IP hard coded in sip.conf. E.g. No registration. |
| 00:11.47 | pabelanger-lap | kc8pxy: example? |
| 00:12.03 | xheliox | Actually, I lied. Scrap all that. |
| 00:12.17 | xheliox | goes back to his cave. |
| 00:14.51 | kc8pxy | is grabbing the data |
| 00:19.10 | xheliox | kc8pxy de w4gpl.. |
| 00:20.50 | kc8pxy | xheliox: qso :) |
| 00:21.45 | kc8pxy | ok, this is getting weirder. |
| 00:22.44 | kc8pxy | why would my addr->IP not have the same ip as my softphone? |
| 00:23.33 | xheliox | is one the external facing IP and the other the internal IP? |
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| 00:37.52 | kc8pxy | ok, so it seems to be my external IP on the softphone.. but that brings up a question in my config. i now know i have the server behind a nat, and it seems the softphone is also behind one, and connecting on funky ports. |
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| 00:42.22 | xheliox | are they both behind the same nat device? |
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| 00:55.38 | pabelanger-lap | kc8pxy: Rather then us guessing, post a debug log |
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| 01:42.02 | kc8pxy | pabelanger-lap: would help if i knew how to generate one, and pastebin it |
| 01:42.35 | pabelanger-lap | ~collectdebug |
| 01:42.36 | infobot | hmm... collectdebug is a method of collecting logs allowing others help troubleshoot an issue. Refer to doc/HOWTO_collect_debug_information.txt or http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt |
| 01:42.49 | pabelanger-lap | kc8pxy: ^^ |
| 01:43.27 | kc8pxy | xheliox: pabelanger-lap thx.. will do.. i have a really weid set of results... |
| 01:44.29 | kc8pxy | pabelanger-lap: problem i think i have, is i'm pretty sure i have a PICNIC error, just not sure which one :) |
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| 02:38.46 | plut0 | is there a way to detect when the remote end has answered the call? |
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| 02:43.29 | WIMPy | yes |
| 02:45.04 | plut0 | WIMPy: care to share? |
| 02:45.32 | WIMPy | Not unless you ask more specific. |
| 02:46.10 | plut0 | i'd like to dial a number, after it answers, senddtmf |
| 02:47.20 | WIMPy | Take a look at option D to Dial(). |
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| 02:56.38 | Deeps | hiya |
| 02:56.48 | Deeps | am using the current trunk, got it all compiled and happy |
| 02:57.00 | Deeps | have loaded chan_mobile and got that seemingly working |
| 02:57.23 | Deeps | however only the first clal appears to have worked correctly |
| 02:57.35 | Deeps | ever since that first call, i get lots of static on varying ends of the call |
| 02:57.57 | Deeps | i've tried alignmentdetection=yes on the adaptor, but instead it made both sides noisy |
| 02:58.48 | Deeps | from what i understand, this is most likely to be a problem relating to my bluetooth adaptor? |
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| 03:02.41 | plut0 | WIMPy: that works, thanks! |
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| 03:55.44 | DogBoy | I discovered last night the secret is to read "the book" |
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| 03:56.08 | DogBoy | as if I didn't know that before |
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| 04:25.44 | boodu | bye |
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| 05:30.03 | thansen | can anyone tell or point me to how I can have 'OR' logic in a gotoif condition? |
| 05:31.18 | thansen | I want to check of ${RECORD_STATUS} == "DTMF" OR "HANGUP" |
| 05:32.08 | carrar | see | |
| 05:32.57 | thansen | carrar: got it, thanks |
| 05:33.10 | carrar | GotoIf($[$["${RECORD_STATUS}" = "DTMF"] | $["${RECORD_STATUS}" = "HANGUP"]]]?true) |
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| 05:33.48 | carrar | err s/]]]/]]/ |
| 05:34.16 | thansen | why can't you just throw the pipe in a single set of [] |
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| 05:34.32 | carrar | try it |
| 05:35.13 | carrar | probably can |
| 05:35.17 | p3nguin_ | such as ["${RECORD_STATUS}" = "DTMF" | "HANGUP"] ? |
| 05:36.10 | thansen | p3nguin_: yes |
| 05:36.21 | thansen | actually..no |
| 05:36.34 | carrar | heh |
| 05:36.46 | thansen | ["${RECORD_STATUS}" = "DTMF" | "${RECORD_STATUS}" = "HANGUP"] |
| 05:37.08 | p3nguin_ | oh |
| 05:37.23 | thansen | *should* that work? |
| 05:37.45 | carrar | test it |
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| 06:05.11 | thansen | carrar: yeah, syntax is valid it appears |
| 06:05.53 | p3nguin_ | But does it work as expected? |
| 06:09.36 | thansen | p3nguin_: yes |
| 06:09.45 | p3nguin_ | Great! |
| 06:09.55 | thansen | very |
| 06:10.21 | thansen | just had to add the k option to my record function :) then everything was working correctly |
| 06:10.32 | thansen | had me scratching my head for a minute |
| 06:11.02 | p3nguin_ | record function? Surely you mean application. |
| 06:11.23 | ChannelZ | funcucation |
| 06:11.53 | ChannelZ | Or I guess that'd be funclication. |
| 06:12.06 | thansen | p3nguin_: yeah :) I'm a noob |
| 06:12.10 | p3nguin_ | I also don't have a k option for Record(), so I really don't know. |
| 06:12.16 | ChannelZ | Which could be very dirty if pronounced incorrectly. |
| 06:12.55 | thansen | p3nguin_: k: Keep recording if channel hangs up. |
| 06:13.17 | p3nguin_ | Must be a 1.6.x option. |
| 06:13.21 | ChannelZ | Indeedy. |
| 06:13.27 | thansen | is on 1.6 |
| 06:13.38 | p3nguin_ | I'll probably never use 1.6.anything. |
| 06:13.55 | thansen | and skip to 1.8 or something? |
| 06:14.02 | p3nguin_ | most likely |
| 06:14.24 | thansen | has kept up to date since 1.2 |
| 06:14.36 | p3nguin_ | I'm up-to-date, as well. |
| 06:14.49 | thansen | nods |
| 06:14.51 | p3nguin_ | I'm using a pretty recent 1.4 release. |
| 06:15.12 | thansen | if there was ever a series to skip that was it |
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| 06:15.26 | p3nguin_ | hmm? |
| 06:15.27 | thansen | with the weird options |
| 06:15.39 | thansen | like | instead of , or whatever it was |
| 06:16.48 | p3nguin_ | As far as I know, all of the 1.4 versions still allow old habits of | instead of , . |
| 06:16.48 | ChannelZ | wasn't | in there since the beginning of time? |
| 06:17.46 | thansen | don't remember, just that they switched away from it right after they added it |
| 06:17.48 | p3nguin_ | I think I was using a | as recently as 1.4.21, if I remember my version number correctly. |
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| 06:18.21 | ChannelZ | I think | was the normal delimiter, then , was added, and now | is finally deprecated |
| 06:18.30 | p3nguin_ | I think I changed them all after tk yelled at me a couple times. |
| 06:18.56 | thansen | ChannelZ: I'm pretty sure it was the opposite |
| 06:19.17 | p3nguin_ | The pipe was used in 1.2. |
| 06:19.41 | p3nguin_ | The pipe still worked in 1.4, but the comma was the new preferred delimiter. |
| 06:20.18 | p3nguin_ | I doubt that the pipe has been reintroduced in 1.6 branches. |
| 06:29.03 | thansen | p3nguin_: http://www.voip-info.org/wiki/view/Asterisk+cmd+VoiceMail |
| 06:29.20 | thansen | looks to me like it was added in 1.4, doesn't really matter I guess though |
| 06:31.15 | ChannelZ | That looks like more of a note to that specific application |
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| 07:07.32 | ranjan | hi all. i want to know how we can call to a normal telephone with asterisk...is it possible to do so?? |
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| 07:09.45 | ranjan | what is t38modem ?? Can it replace a hardware modem?? |
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| 07:12.05 | ranjan | ChannelZ, thansen, do you know what t38modem is? |
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| 07:23.58 | ChannelZ | yeah, a contradiction in terms |
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| 07:26.43 | gavimobile | folks, im having trouble with music on hold. http://www.pastebin.org/385068 im using asterisknow version 1.7, could someone give me a hand please |
| 07:27.24 | gavimobile | i tried to upload both .wav and .mp3 files, without any "-" or "_" in it. seems to add a "_" regardless. i noticed that asterisk things there are 2 slashes before the file name "//orig_LetsGo1" also i dont see an extention name eg".mp3" or ".wav" listed in the log, however in /var/lib/asterisk/moh/test/orig_LetsGo1 has the extention name of .mp3. |
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| 07:36.02 | ChannelZ | looks like maybe you have a trailing / on the pathname for the MOH files |
| 07:37.51 | ChannelZ | * can't play .mp3 without the add-on handler, but it's a waste of CPU anyway. Easier to do 8khz mono .wav files, or if you are using a specific codec all the time for your channels (gsm, g729 etc) to have them in the native format. |
| 07:39.35 | gavimobile | ChannelZ: I didn't addthe trailing /, maybe its freepbx, but freepbx says its not my issue and that it's a bug with asterisknow |
| 07:39.48 | gavimobile | ChannelZ: please see http://www.freepbx.org/forum/freepbx/beta-program-issues/moh-doesnt-work#comment-27376 |
| 07:39.56 | ChannelZ | This is why I hate FreePBX |
| 07:40.34 | gavimobile | ChannelZ: lol... trixbox? |
| 07:41.03 | ChannelZ | how about just asterisk |
| 07:41.10 | DogBoy | I couldn't even figure out what tribox was |
| 07:41.15 | DogBoy | lame site |
| 07:41.16 | ChannelZ | PBX for whores |
| 07:41.32 | DogBoy | it should say that on an about link |
| 07:41.36 | gavimobile | elastix? |
| 07:41.39 | DogBoy | but it don't say nothing |
| 07:42.00 | ChannelZ | re: how about just asterisk? |
| 07:42.08 | DogBoy | that's how I roll |
| 07:42.20 | gavimobile | gurus |
| 07:42.24 | ChannelZ | If you want to learn asterisk, you're not going to by using these gay GUIs |
| 07:42.35 | DogBoy | same is true of any guis |
| 07:42.36 | ChannelZ | which are purposely designed to completely obfuscate how the thing works |
| 07:43.08 | ChannelZ | in any case, the link to the discussion you provided seems specific to mp3 files. wav's should be working. |
| 07:43.12 | gavimobile | ill bet yourright... |
| 07:43.37 | gavimobile | but I learn much better with hands on then taking it apart |
| 07:43.59 | gavimobile | so I would really like to get this gay asteriskbox of mine working |
| 07:44.01 | ChannelZ | ~book |
| 07:44.02 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 07:46.51 | ChannelZ | re: the wav files should work so long as they are the right format |
| 07:47.32 | ChannelZ | 8khz, MONO, 16-bit is fine - called blah.wav etc |
| 07:48.05 | gavimobile | ChannelZ: ill try to convert them now |
| 07:48.06 | gavimobile | thanks |
| 07:48.47 | ChannelZ | You can grab the stock files from http://downloads.asterisk.org/pub/telephony/sounds/asterisk-moh-opsound-wav-current.tar.gz |
| 07:49.45 | ChannelZ | and for the record, the double slashes seem not to be a problem |
| 07:50.05 | gavimobile | ChannelZ: I believe these were working. but I want to custom sounds |
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| 07:53.22 | ChannelZ | ok so you know it's you then |
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| 08:13.50 | joako | I have a queue setup. Is there a way that I can set different paramters depending on where in the dialplan it was called? I want to be able to enter some people into the queue with periodic annoucements and some without |
| 08:14.33 | joako | I think I might need to create 2 queues define agents as a 3rd queue? |
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| 08:24.19 | ttwhy | Hi, can i limit the slots for a possible queue connection? (i am using zoiper which offers more than one parallel connection and so does not report the queue list, that the agent is already in a call whats stupid for a priority based queue list ;) ) |
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| 08:54.42 | ttwhy | a method to limit the parallel calls of a agent from a queue list? someone got any ideas? |
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| 10:27.51 | vlt | Hello. Is there a way to "redirect" call that is currently ringing on one extension to another one from the CLI? |
| 10:27.59 | vlt | *a call |
| 10:33.56 | nettie | Hi guys, anyone using Sangoma B700 FlexBRI card please? |
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| 11:01.20 | Bartockbatz | Hey Ladies and Germs - just flew in from LA - cripes my arms are tired! |
| 11:01.27 | Bartockbatz | I know - old joke |
| 11:02.38 | Bartockbatz | Hey - anyone point me to a good, opensource voice-recognition library for Asterisk? I see a few commercial ones out there, but my Scottish roots make me too cheap to purchase the license. |
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| 11:08.47 | jblack | <PROTECTED> |
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| 12:01.20 | telnettech | good morning everyone....how is the channel doing this fine morning?\ |
| 12:01.46 | Faustov | ~seen gr0mit |
| 12:01.47 | infobot | gr0mit <~tim@81.187.67.134> was last seen on IRC in channel #asterisk, 15d 15h 41m 52s ago, saying: 'hey guys - can anyone recommend a provider who can provide a 212 area DID please?'. |
| 12:02.03 | Faustov | lies |
| 12:03.06 | tzafrir | where does it? |
| 12:03.54 | *** join/#asterisk AlHafoudh (~alhafoudh@195.46.69.4) |
| 12:03.57 | AlHafoudh | hi all |
| 12:04.06 | AlHafoudh | guys, does someone have worked with Nuance TTS/ASR? |
| 12:04.51 | AlHafoudh | when I try to synthesize text in other language than english, i get this error: Jul 07 11:40:21.50|TUCPU=62|TKCPU=31|TID=3980|0|9102|SID=EMDEELIFAAAFJDBLAAAAAAAB|Vocalizer Plugin|Cannot process speak data. TTS_ERROR \= TTS_E_SSML_PARSE_ERROR(Code\=180) |
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| 12:10.49 | *** join/#asterisk jnicola (~jnicola@186.124.123.2) |
| 12:13.33 | jnicola | hello all! |
| 12:14.07 | jnicola | im trying with free fax for asterisk.. asterisk v1.6.2.9 |
| 12:14.38 | jnicola | but i cant receive fax. |
| 12:14.45 | jnicola | the error is: |
| 12:15.01 | jnicola | just a minute |
| 12:16.42 | *** join/#asterisk Pazzo (~ugelt@reserved-225136.rol.raiffeisen.net) |
| 12:18.47 | jnicola | ERROR[18673]: res_fax.c:685 set_fax_t38_caps: channel 'SIP/XXX-00000007' is in an unsupported T.38 negotiation state, cannot continue. |
| 12:18.59 | jnicola | im tryng with t38. |
| 12:19.05 | jnicola | without g711 |
| 12:19.38 | jnicola | its a problem of sender or its a problem of my asterisk!?¿ |
| 12:20.33 | [TK]D-Fender | You need G.711 for T.38 |
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| 12:27.09 | evilbit | jnicola: are you following a particular how-to? fax over t38 is next on my todo list |
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| 12:27.46 | jnicola | fax for asterisk digium manual |
| 12:28.17 | jnicola | i have downloaded a free license.. |
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| 12:29.25 | jnicola | i set max and min rate. |
| 12:29.53 | jnicola | i set the peer in sip.conf... set all in [general] section in sip.conf |
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| 12:32.46 | evilbit | ah, ok |
| 12:33.00 | *** join/#asterisk DelphiWorld (~Delphi@196.20.95.121) |
| 12:33.35 | DelphiWorld | ANY PROVIDER THAT ofer Iax2?* |
| 12:36.13 | *** join/#asterisk itiliti (~itiliti@74.222.43.219) |
| 12:36.21 | itiliti | good morning all. |
| 12:36.34 | itiliti | I am trying to forward calls through asterisk, and record the audio stream. |
| 12:36.35 | [TK]D-Fender | DelphiWorld: teliax, les.net |
| 12:36.55 | DelphiWorld | [TK]D-Fender: for wholesale or retail please? |
| 12:37.02 | itiliti | my prolem is that when I tell the box to answer the calls, my dialer get confused, and thinks the call is answered. |
| 12:37.24 | itiliti | So I am trying to forward the calls without answering the call, and then start recording after it is answered by the called party. |
| 12:38.43 | itiliti | Here is my dialplan.... |
| 12:38.44 | itiliti | http://pastebin.com/b8Pe2qVk |
| 12:39.13 | itiliti | I removed the answer, and now I get: chan_sip.c:20063 handle_request_invite: Call from 'patton-3' to extension '3124988982' rejected because extension not found. |
| 12:39.18 | [TK]D-Fender | DelphiWorld: YES |
| 12:39.30 | DelphiWorld | [TK]D-Fender: thank you a lot |
| 12:39.35 | itiliti | patton-3 is my device on the inside that conencts via PRI to my dialer... |
| 12:39.49 | [TK]D-Fender | itiliti: You have no priority "!" <- |
| 12:39.52 | [TK]D-Fender | "1" |
| 12:40.11 | itiliti | ahhh... so put a 1 on the first line? |
| 12:40.23 | AlHafoudh | anyone has experience with Nuance TTS/ASR? |
| 12:40.23 | [TK]D-Fender | itiliti: Yes, the first has to be "1" |
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| 12:46.37 | pabelanger | itiliti: *CLI> dialplan show 3124988982@dialer-out |
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| 12:50.30 | itiliti | sounds good. it worked, thx.... |
| 12:51.16 | itiliti | dont know why I didnt see that..... duh.... |
| 12:52.36 | [TK]D-Fender | itiliti: Alway start be the basics. Confirm your very first priority is 100% right. Then make sure that the call is even landing in that CONTEXT <_ |
| 12:53.18 | [TK]D-Fender | itiliti: This means looking at the call. Your debug doesn't prove what peer got matched so you may think "Yeah it says to go to ABC!", but if the auth fails it'll fall elsewhere. |
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| 13:00.46 | itiliti | I got you. It is working properly. Thx four your quick help... |
| 13:01.28 | itiliti | BTW, what does the "_" mean anyway. |
| 13:01.37 | evilbit | I use teliax, and they are pretty good |
| 13:02.35 | evilbit | _ is a pattern matcher |
| 13:03.04 | evilbit | or rather it says that what comes after is a pattern to be matched |
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| 13:18.32 | itiliti | anyone know what the switch is to turn off the "beep" on Mixmonitor? |
| 13:18.54 | itiliti | It just plays once at the beginning of the mixmonitor app to the caling party... |
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| 13:31.34 | ncopa | hi |
| 13:32.02 | ncopa | are there any rss feed or something i can subscribe to for new releases of asterisk and related packages? |
| 13:33.27 | tzafrir_laptop | ncopa, http://lists.digium.com/mailman/listinfo/asterisk-announce |
| 13:33.38 | ncopa | perfect |
| 13:33.41 | ncopa | thanks |
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| 13:35.37 | *** mode/#asterisk [+o malcolmd] by ChanServ |
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| 13:39.57 | *** join/#asterisk bragon (~Alexandre@81.93.247.165) |
| 13:40.00 | bragon | Hi |
| 13:40.49 | bragon | Could we have with asterisk 1.2.27 have many extension.conf and sip.conf ? |
| 13:41.16 | *** join/#asterisk cesar_CR (~cesar@201.192.86.30) |
| 13:41.28 | pabelanger | bragon: #include? |
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| 13:41.43 | bragon | pabelanger: where can i include that ? |
| 13:41.49 | bragon | pabelanger: in extension.conf ? |
| 13:42.04 | bragon | pabelanger: do you have the syntaxe ? |
| 13:42.11 | bragon | Or an exemple. |
| 13:42.24 | pabelanger | bragon: I doubt 1.2 supports it |
| 13:42.42 | bragon | i can't upgrade. |
| 13:43.11 | pabelanger | bragon: then backport the code |
| 13:45.22 | bragon | :'( |
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| 14:08.21 | [TK]D-Fender | bragon: Yes 1.2 supports includes |
| 14:08.50 | bragon | [TK]D-Fender: \o. |
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| 14:09.27 | bragon | thanks |
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| 14:21.43 | Bullterd | Hey All - Anyone know if localphone let you change the asterisk box a geo DDI is pointed at? |
| 14:26.32 | *** join/#asterisk wcselby (~wcselby@216.110.88.194) |
| 14:30.26 | AlHafoudh | using Nuance Speech server, when I try to synthesize text in other language than english, i get this error: Jul 07 11:40:21.50|TUCPU=62|TKCPU=31|TID=3980|0|9102|SID=EMDEELIFAAAFJDBLAAAAAAAB|Vocalizer Plugin|Cannot process speak data. TTS_ERROR \= TTS_E_SSML_PARSE_ERROR(Code\=180) |
| 14:30.34 | AlHafoudh | what should i do? |
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| 14:47.21 | pabelanger | AlHafoudh: Contact Nuance? |
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| 15:00.30 | *** join/#asterisk _pepo_ (c837ea09@gateway/web/freenode/ip.200.55.234.9) |
| 15:00.37 | _pepo_ | hi friends |
| 15:04.35 | *** join/#asterisk [T]ank (~ckwall@206.71.78.158) |
| 15:07.35 | [T]ank | I am getting an error when dialing out using h323. It says that the host does not exist. I have my context in the config, I dont understand why i am getting this error: I have posted some output and configs. Could someone double check my work? http://pastebin.com/MBMBmreC |
| 15:08.10 | Katty | good morning all you beautiful people!!!!! |
| 15:08.30 | markfeatherston_ | Howdy! |
| 15:08.47 | [TK]D-Fender | [T]ank: exten => _XXXXX,1,Dial(H323/Avaya/${EXTEN}) |
| 15:09.02 | [T]ank | hmmm... ok. |
| 15:09.53 | [T]ank | gives me the same output. Its saying "No such host: Avaya" |
| 15:10.37 | _pepo_ | I am with problems in my queue, some calls are rejected after hearing the message from the queue when it tells me my place in queue, the call is closed |
| 15:12.46 | _pepo_ | what can be wrong? |
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| 15:14.04 | [TK]D-Fender | _pepo_: Absolutely anything. Especially because you are showing absolutely nothing. |
| 15:15.45 | [T]ank | am i using the correct config files? |
| 15:16.07 | wcselby | Katty - you're far too happy at this time of the day |
| 15:16.13 | [T]ank | I cant understand why it wouldnt find the host [Avaya] since it is in the h323.conf |
| 15:16.14 | *** join/#asterisk Faithful (~Faithful@202.6.145.116) |
| 15:17.19 | _pepo_ | the use of my call is: exten => 100, n, Queue(callcenter-100-152,,,,600) |
| 15:17.56 | _pepo_ | is there some configuration in queues.conf that cause thiw desconection? |
| 15:18.16 | wcselby | [T]ank - is there an "h323 show peers" option of some sort? |
| 15:18.26 | [TK]D-Fender | _pepo_: Look at the ENTIRE CALL. Not just 1 silly line of dialplan. |
| 15:19.18 | *** join/#asterisk devmod (~devmod@c-76-100-208-204.hsd1.md.comcast.net) |
| 15:19.24 | [T]ank | wcselby: Im not seeing anything.... at least anything similar to like a sip show peers... currently google'ing. brb |
| 15:21.57 | [T]ank | there is an ooh323 show peers. but that is an entirely different channel type, isnt it? |
| 15:22.54 | [T]ank | if i dial using the 00h323 channel type, i dont get any host errors, but it gives me a congestion error. |
| 15:25.12 | *** part/#asterisk sekil (~sekil@80.93.247.26) |
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| 15:27.07 | [T]ank | http://pastebin.com/3Z9CVwta |
| 15:27.32 | Joe_CoT | so when compiling asterisk and choosing options with make menuconfig, I don't have the option to compile app_meetme, because it depends on dahdi. I have dahdi compiled and loaded into the kernel. Any idea what asterisk is really checking? |
| 15:28.12 | pabelanger | Joe_CoT: re-run ./configure in Asterisk |
| 15:28.14 | WIMPy | Did you run configure after installing dahdi? |
| 15:28.28 | pabelanger | Joe_CoT: If still a problem pb your config.log |
| 15:28.32 | [T]ank | start with a make distclean, then configure |
| 15:30.10 | *** join/#asterisk ^Winampbot^ (~JH2000@p5B3D8AB0.dip0.t-ipconnect.de) |
| 15:30.41 | markfeatherston_ | Joe_CoT: configure seems to try to build a test file linking against dahdi. If it can't find the headers or the libs it wouldn't work |
| 15:30.57 | ^Winampbot^ | how can i run a command or script when a telefon logs in to sip? |
| 15:31.30 | [T]ank | ^Winampbot^: look into the asteriskmanager and AGI |
| 15:31.40 | Joe_CoT | ran make distclean (which did more than make clean did, so i was hopeful), then ./configure, then make menuconfig, still XXXed out |
| 15:31.58 | [T]ank | which packages have you isntalled? |
| 15:32.06 | [T]ank | there are two dahdi packages. |
| 15:32.37 | Joe_CoT | I have installed dahdi, based off what mark said looks like I need dahdi-source as well. trying that |
| 15:33.07 | markfeatherston_ | Joe_CoT: does "/usr/include/dahdi/dahdi_config.h" exist? |
| 15:33.19 | ^Winampbot^ | hm, ok, i´ll look there |
| 15:33.41 | ChannelZ | ^Winampbot^: you can try 'regcontext' and 'regexten' in sip.conf to make it start executing a dialplan context, depending on what you need done. (I guess you could run an AGI too.) |
| 15:34.03 | Joe_CoT | markfeatherston_, it does now. Installing the dahdi-source fixed it, thanks! |
| 15:34.22 | [T]ank | awesome |
| 15:35.38 | ^Winampbot^ | ah okay, thats exactly what i´ve looked for :D |
| 15:35.40 | ^Winampbot^ | thank you |
| 15:37.35 | *** join/#asterisk Godfather_ (~Godfather@35.Red-88-7-6.staticIP.rima-tde.net) |
| 15:38.11 | [T]ank | here is the debug output when using ooh323 http://pastebin.com/yr46tBZe |
| 15:38.18 | Godfather_ | hi |
| 15:38.41 | aphexer | is it possible that the M() option in the Dial command only works when calling local SIP users? |
| 15:38.53 | aphexer | I can only get it to work when dialing local sip users... |
| 15:39.10 | *** join/#asterisk hfb (~hfb@pool-98-119-197-22.lsanca.dsl-w.verizon.net) |
| 15:39.43 | ChannelZ | it shouldn't really know the difference. |
| 15:39.55 | [TK]D-Fender | aphexer: No. |
| 15:39.57 | [T]ank | http://pastebin.com/vf7bhyPi here are my registered channel types. does everything look ok for ooh323 and h323? |
| 15:40.53 | [T]ank | devicestate shows "no" shouldnt those be "yes"? |
| 15:44.38 | aphexer | ChannelZ, [TK]D-Fender: could you take a look at this? http://pastebin.com/CV1bkhU9 |
| 15:49.29 | *** join/#asterisk bsaxon (~bsaxon@12.107.149.61) |
| 15:50.36 | tzafrir_laptop | Joe_CoT, you need dahdi-source and libtonezone-dev |
| 15:50.47 | [TK]D-Fender | aphexer: Perhaps you should actually look at the ENTIRE CALL |
| 15:51.04 | Joe_CoT | tzafrir_laptop, already go it, thanks |
| 15:51.20 | tzafrir_laptop | I don't think you need to have dahdi itself installed merely for building asterisk |
| 15:52.08 | [T]ank | im freaking stumped... how is this not finding the host from h323.conf? Or, why cant i get the call to go out via ooh323? |
| 15:52.18 | aphexer | [TK]D-Fender: how do you mean? |
| 15:52.24 | *** join/#asterisk s14ck (~jtorres@srvveccs01.onuva.net) |
| 15:55.55 | markfeatherston_ | aphexer: not sure if this is what he meant, but try turning up the verbosity and watch the failing call in the console or /var/log/asterisk/full |
| 15:56.26 | *** join/#asterisk UQlev (~yuriy@212.50.100.76) |
| 15:57.59 | [TK]D-Fender | aphexer: Showing a catalog sales sheet for a gun won't help investigate a murder. You may want to consider looking at the VIDEO RECORDING of the perpetrator shooting the victim in the head with it <- |
| 15:58.06 | *** join/#asterisk RobH (~robh@wikimedia/RobH) |
| 15:58.32 | [TK]D-Fender | aphexer: Picture of car "new" = worthless. Video of "car crash" useful". |
| 15:58.51 | paulc | pabelanger-lap: Delayed reply but thanks for that heads up on asterisk test suite from yesterday - much appreciated :) |
| 15:58.55 | aphexer | [TK]D-Fender: hm, I'll need a translation into tech talk :) |
| 15:59.21 | aphexer | markfeatherston_: I'll try increasing debug output... |
| 16:00.53 | *** join/#asterisk nightwalk (~nightwalk@daimon.vixel.org) |
| 16:02.37 | *** join/#asterisk Whtsup (~sssi@WimaxUser376-208.wateen.net) |
| 16:02.43 | Whtsup | helllo how r u |
| 16:02.47 | Whtsup | <PROTECTED> |
| 16:02.58 | Whtsup | i m getting this message in my asterisk console what does it means |
| 16:03.23 | *** join/#asterisk raden_work (~jon@69-179-99-17.stat.centurytel.net) |
| 16:05.30 | Whtsup | hello |
| 16:05.35 | Whtsup | any one alive ? |
| 16:05.39 | markfeatherston_ | nope |
| 16:08.11 | aphexer | markfeatherston_: the logs don't tell me anything knew... it does look like it only executes the .sh script only once though |
| 16:08.47 | *** join/#asterisk DelphiWorld (~Delphi@196.20.95.121) |
| 16:08.51 | DelphiWorld | [TK]D-Fender: sory! |
| 16:09.03 | DelphiWorld | [TK]D-Fender: all the providers that you gave me require a cc;) |
| 16:09.33 | Whtsup | <PROTECTED> |
| 16:09.43 | Whtsup | wht does it means |
| 16:09.47 | Whtsup | need help plz |
| 16:10.40 | Chainsaw | DelphiWorld: Is it a problem to be personally identifiable? Are you planning on selling certain pharmaceuticals? |
| 16:10.53 | *** join/#asterisk jesselang (~jesse@c-24-131-130-197.hsd1.mn.comcast.net) |
| 16:11.04 | DelphiWorld | Chainsaw: be calm and know what you are saying |
| 16:11.07 | *** join/#asterisk ariel_ (~chatzilla@c-24-127-196-248.hsd1.fl.comcast.net) |
| 16:11.15 | markfeatherston_ | Whtsup: it looks like there are dozens of things that can cause that error. firewall messing with rtp, a bad pstn, codec issues, specific sip client issues. |
| 16:11.26 | Chainsaw | DelphiWorld: Classic feet shuffle. Okay. |
| 16:11.45 | DelphiWorld | Chainsaw: in here there is no credit card |
| 16:12.08 | markfeatherston_ | aphexer: I'm not sure what that means in this context, what sh script? Could you pastebin some of the logs? |
| 16:13.11 | aphexer | markfeatherston_: http://pastebin.com/CV1bkhU9, that's a part of the dialplan |
| 16:13.16 | aphexer | I call a sh script there to get some dat |
| 16:13.18 | aphexer | data* |
| 16:13.41 | aphexer | the log file only shows one AGI "run" |
| 16:15.09 | p3nguin_ | delphiworld: VoIP.ms has IAX2 support and you can pay by PayPal. |
| 16:15.24 | DelphiWorld | yes, p3nguin_ |
| 16:15.33 | DelphiWorld | p3nguin_: i am with it allready :D |
| 16:15.40 | DelphiWorld | p3nguin_: but i can't see the config |
| 16:16.40 | markfeatherston_ | aphexer: that may be the problem. pastebin the call logs though if you can. it should be running it twice |
| 16:16.41 | p3nguin_ | "the config" <-- what EXACTLY are you wanting to see? |
| 16:18.51 | aphexer | markfeatherston_: http://pastebin.com/AUxwH6RX => that's all I can find in the 'full' log |
| 16:19.09 | aphexer | (except some messages when the config is reloaded etc) |
| 16:23.11 | markfeatherston_ | aphexer: that's not what I need to see, that's just the sh script executing. I need to see the actual call log where it tells it to execute the agi script |
| 16:24.19 | *** join/#asterisk s14ck (~jtorres@srvveccs01.onuva.net) |
| 16:24.37 | markfeatherston_ | aphexer: core set verbose 4, and find the call again. That will show the dialplan executing as well |
| 16:24.46 | aphexer | ok thanks, i'll try that |
| 16:25.54 | troy42 | yeah, also agi set debug |
| 16:26.17 | troy42 | you'll see args to each command, at least with fastagi and i assume with local agi |
| 16:28.45 | aphexer | [Jul 7 20:25:16] VERBOSE[28067] logger.c: == Spawn extension (macro-sendcustomername, s, 3) exited non-zero on 'SIP/To-Provider-09938820' in macro 'sendcustomername' => that might be it... though I don't understand why it happens |
| 16:29.06 | aphexer | s,3, that's the Festival command |
| 16:29.21 | aphexer | why would it exit non-zero? |
| 16:29.32 | markfeatherston_ | aphexer: are you using sip reinvite? |
| 16:29.38 | aphexer | no |
| 16:29.42 | aphexer | not that i know :) |
| 16:31.27 | *** part/#asterisk bzing2 (~dr105@dhcp-194-66-208-235.canterbury.ac.uk) |
| 16:33.08 | *** join/#asterisk nicknick (~administr@host213-123-201-13.in-addr.btopenworld.com) |
| 16:34.50 | aphexer | markfeatherston_: a more complete log is here: http://pastebin.com/071CrsPk |
| 16:37.28 | *** join/#asterisk Gary_B (~IceChat7@85.211.169.212) |
| 16:37.48 | Gary_B | are there different versions of Q.931 for ISDN? (uk) |
| 16:38.29 | Gary_B | i just heard our telecom providers line is V0.8, is this different from V1, is there a standard |
| 16:39.01 | *** join/#asterisk bmg505 (~leon@196-209-7-27.dynamic.isadsl.co.za) |
| 16:42.32 | *** join/#asterisk idespinner (~idespinne@cpe-76-93-115-243.socal.res.rr.com) |
| 16:43.40 | *** part/#asterisk DelphiWorld (~Delphi@196.20.95.121) |
| 16:43.50 | leifmadsen | for those buying voip adapters in Canada, who do you use? |
| 16:44.28 | leifmadsen | aha, I think I found who I used before (canadianvoipstore.com) |
| 16:49.09 | markfeatherston_ | aphexer: I'm really not sure what is happening. Check to make 100% sure sip reinvite isn't enabled though. |
| 16:51.52 | aphexer | markfeatherston_: I just checked, everyone it says canreinvite=no... but what has that to do with this problem? |
| 16:53.02 | *** join/#asterisk jesselang (~jesse@c-24-131-130-197.hsd1.mn.comcast.net) |
| 16:54.15 | markfeatherston_ | aphexer: if you use a sip trunk and it was forwarding the rtp stream there, i don't think agi can do anything with it |
| 16:55.38 | aphexer | markfeatherston_: it's rather Festival which is trying to output some audio |
| 16:55.51 | aphexer | markfeatherston_: why wouldn't that be possible? just sending some audio over a channel, and after that connect it... |
| 16:57.48 | *** join/#asterisk spiceycurry (~mcurry@proxy.hostopia.com) |
| 16:57.57 | markfeatherston_ | aphexer: it should work as far as I can tell. I'm not sure what else would cause it. |
| 16:58.28 | spiceycurry | Is 'h' a reserved character in the first parameter of an exten=> h,1,... meaning hang-up? |
| 16:58.34 | spiceycurry | (When hung up)? |
| 16:58.57 | spiceycurry | Better yet, is 'h' an extension for "hung-up" ? |
| 16:59.15 | Bullterd | Exchange can suck my cock tbh |
| 16:59.22 | Bullterd | just totally fucked me out of my evening |
| 16:59.42 | aphexer | markfeatherston_: ok, well thanks for looking at it :) |
| 17:00.00 | markfeatherston_ | aphexer: no problem. Good luck with the agi |
| 17:00.07 | aphexer | thanks |
| 17:00.18 | p3nguin_ | spiceycurry: h is the hangup extension. |
| 17:00.35 | spiceycurry | ok thanks, thats what I was praying to jesus about. |
| 17:00.45 | spiceycurry | Where is Jesus anyhow? |
| 17:01.02 | spiceycurry | [TK]-D is my jesus |
| 17:01.19 | [TK]D-Fender | walks on water... and holds heads under |
| 17:01.45 | [TK]D-Fender | spiceycurry: "h" = Asterisk Standard Extension. They're like Pokemon... gotta catch'em all |
| 17:01.53 | spiceycurry | haha |
| 17:02.06 | spiceycurry | ok great |
| 17:02.22 | spiceycurry | ~ |
| 17:02.34 | *** part/#asterisk Bullterd (~Bullterd@bullbnc.org) |
| 17:02.36 | *** part/#asterisk Gary_B (~IceChat7@85.211.169.212) |
| 17:06.03 | p3nguin_ | ~asterisk standard extensions |
| 17:06.04 | infobot | [asterisk standard extensions] http://www.voip-info.org/wiki/view/Asterisk+standard+extensions |
| 17:06.36 | [TK]D-Fender | spiceycurry: And no, it isn't so much "reserved" as it is "jumped to by default on hangup". You could jump there explicitly, pass it as the exten to dial in a REGISTER statement, dial it direct on a phone with alpha capabilities, etc. |
| 17:08.01 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 17:15.37 | *** join/#asterisk neurosys (~neurosys@173-9-159-177-miami.txt.hfc.comcastbusiness.net) |
| 17:15.50 | *** part/#asterisk spiceycurry (~mcurry@proxy.hostopia.com) |
| 17:17.06 | *** join/#asterisk spiceycurry (~mcurry@proxy.hostopia.com) |
| 17:18.32 | spiceycurry | I'm using Fax For Asterisk, I am receiving faxes perfectly, and I have the tiff files being placed in a directory when they are completed. There are a lot of variables I would like to store when the fax is received (such as the caller id, the number called, etc). How could this be done easily? |
| 17:19.53 | *** join/#asterisk aidinb (~Aidin@wsip-98-190-45-225.oc.oc.cox.net) |
| 17:20.35 | spiceycurry | Perhaps there is methods to write SET variables to a text file ? |
| 17:21.15 | *** join/#asterisk [sr] (~Unknowned@pal-213-228-140-150.netvisao.pt) |
| 17:21.16 | [sr] | howdy |
| 17:21.18 | tzafrir_laptop | System() |
| 17:21.49 | spiceycurry | ah ok great |
| 17:21.52 | spiceycurry | thanks |
| 17:26.48 | *** part/#asterisk neurosys (~neurosys@173-9-159-177-miami.txt.hfc.comcastbusiness.net) |
| 17:27.28 | *** join/#asterisk neurosys (~neurosys@173-9-159-177-miami.txt.hfc.comcastbusiness.net) |
| 17:28.40 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.167.9.dsl.dyn.forthnet.gr) |
| 17:33.04 | Bartockbatz | okay - silly question time again |
| 17:34.16 | Bartockbatz | I want to configure my dial plan, so that if I dial a particular code, such as 898 from any particular extension, I can dial out using my SIP trunk |
| 17:35.49 | Bartockbatz | I am a little clueless - I am thinking that I need to use the Dial() application - but I am wondering if anyone can give me boost? |
| 17:36.22 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-50.hsd1.nj.comcast.net) |
| 17:36.38 | tzafrir_laptop | Bartockbatz, what do you mean by "code"? Number to dial? |
| 17:36.44 | *** join/#asterisk m_tadeu (~quassel@89.181.43.198) |
| 17:36.51 | Bartockbatz | oh - sorry |
| 17:36.57 | lirakis | BarthezZ, ... exten _898XXXXXXX => Dial(sip/provider/${EXTEN:3}) |
| 17:37.05 | lirakis | or however you remove the first 3 |
| 17:37.11 | *** join/#asterisk DJF5 (~dennisdeg@84-105-183-83.cable.quicknet.nl) |
| 17:37.16 | thehar | 9:00 < Lenolium> det3: Yeah, he's out today, heading to SF, then returns tomorrow morning and leaves for LV |
| 17:37.20 | thehar | oops |
| 17:37.44 | Bartockbatz | kind of like on a traditional PBX, dial 9, then your phone number for an 'outside' line? |
| 17:38.05 | *** join/#asterisk githogori (~githogori@adsl-66-123-22-146.dsl.snfc21.pacbell.net) |
| 17:38.09 | lirakis | BarthezZ, --------------^ |
| 17:38.19 | lirakis | oh |
| 17:38.22 | lirakis | Bartockbatz, |
| 17:38.23 | lirakis | sorry |
| 17:38.27 | lirakis | tab completed wrong |
| 17:38.34 | lirakis | see above |
| 17:38.36 | Bartockbatz | Okay - I see lirakis :) |
| 17:38.43 | Bartockbatz | thanks much! |
| 17:39.41 | Bartockbatz | so, just so I fully understand lirakis , the exten_898xxxxxxxxx=> is any extension who dials 898 |
| 17:40.01 | p3nguin_ | Why would anyone need to dial 898 on the front of the phone numbers? |
| 17:40.11 | *** join/#asterisk leif[mobile] (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 17:40.11 | *** mode/#asterisk [+o leif[mobile]] by ChanServ |
| 17:40.51 | Bartockbatz | then the Dial(sip/provider/${EXTEN:3}) means dial using sip provider the number from the extension, dropping the first 3 numbers? |
| 17:41.17 | Bartockbatz | lirakis: ------------------------^ |
| 17:41.24 | lirakis | Bartockbatz, not just 898. 898 followed by 10 digits |
| 17:41.29 | leif[mobile] | The first 3 chars of the EXTEN var yes |
| 17:41.46 | lirakis | that gets everything but the first 3 i thought |
| 17:42.04 | p3nguin_ | It is a 3-character offset. |
| 17:42.12 | lirakis | p3nguin_, right |
| 17:42.14 | Bartockbatz | okay - it is a European SIP trunk - so , more than 10 digits - so I would have to wildcard that |
| 17:42.19 | lirakis | so it gets all except the first 3 |
| 17:42.23 | lirakis | Bartockbatz, change as needed |
| 17:42.27 | lirakis | i gave you an example.... |
| 17:42.28 | lirakis | run with it |
| 17:42.34 | p3nguin_ | But why would anyone need to dial 898 on the front of the phone numbers? |
| 17:42.42 | p3nguin_ | Why not just dial the phone number you want to call? |
| 17:42.43 | markfeatherston_ | p3nguin_: we have a prefix for dialing out with a different caller id. |
| 17:43.09 | p3nguin_ | Different prefixes = different caller id numbers? |
| 17:43.29 | Bartockbatz | thank you lirakis |
| 17:43.29 | lirakis | sensing disaster, quietly walks away |
| 17:43.32 | p3nguin_ | Will you ever call one phone number and use more than one caller id number? |
| 17:43.52 | p3nguin_ | What are the conditions for changing the caller id number? |
| 17:44.36 | markfeatherston_ | p3nguin_: I run a collection agency's IT. Some companies want us to act as internal collections for them, so we call out as that business when we collect on those accounts |
| 17:45.05 | markfeatherston_ | so just depending on the time of day for that agent, or what their manager says t hey should work on |
| 17:45.06 | *** join/#asterisk jetlag (jetlag@pool-173-61-208-114.cmdnnj.east.verizon.net) |
| 17:46.01 | p3nguin_ | So dialing a prefix of 55 might set your caller id number to 3214567890, and dialing a prefix of 88 might set the caller id number to 8775551111? |
| 17:46.10 | markfeatherston_ | yea |
| 17:47.20 | p3nguin_ | Your application of it makes at least a little sense. bartockbatz, on the other hand, doesn't seem to have any good reason other than he's living in the age of an old PBX where he has to dial a code to get an "outside line." |
| 17:47.42 | markfeatherston_ | ahh |
| 17:48.24 | p3nguin_ | Screw all that. Just pick up the phone and dial the number you wish to be connected to. |
| 17:50.06 | *** join/#asterisk clintc (~clintc@n128-227-105-14.xlate.ufl.edu) |
| 17:51.03 | *** join/#asterisk sgimeno (~chatzilla@95.122.8.54) |
| 17:55.45 | *** join/#asterisk bmg505 (~leon@196-209-7-27.dynamic.isadsl.co.za) |
| 18:04.51 | *** join/#asterisk rustyclarkson (~rusty@u53.sutus.com) |
| 18:06.46 | Bartockbatz | okay - next question folks |
| 18:07.54 | Bartockbatz | maybe I fat-fingered something |
| 18:08.13 | markfeatherston_ | that's not a question |
| 18:08.41 | Bartockbatz | exten _898XXXXXXX => Dial(sip/provider/${EXTEN:3}) - exten = or exten => ??? |
| 18:09.12 | [TK]D-Fender | Bartockbatz: VERY wrong |
| 18:09.23 | [TK]D-Fender | barkGo read any other code sample. Seriously |
| 18:09.40 | p3nguin_ | ~answers |
| 18:09.41 | infobot | [~answers] Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & Docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: /path/to/src/asterisk/UPGRADE*.txt |
| 18:10.18 | *** join/#asterisk Arsenick (~y@modemcable230.231-70-69.static.videotron.ca) |
| 18:11.46 | Bartockbatz | Okay [TK]D-Fender - best place for me to look this up - if I read about it , then try it , I won't have to ask you folks all these silly questions! |
| 18:11.50 | Bartockbatz | ?? |
| 18:12.21 | [TK]D-Fender | Bartockbatz: exten => pattern,priority,application(data) |
| 18:12.30 | *** join/#asterisk SiNGLer (~singler@78-60-54-125.static.zebra.lt) |
| 18:12.48 | *** join/#asterisk xheliox (~jeff@i216-58-41-253.cybersurf.com) |
| 18:12.52 | [TK]D-Fender | Bartockbatz: Your's was split in bits with "=>' jsut shevaed anywhre, and no priority. |
| 18:13.10 | [TK]D-Fender | shoved* |
| 18:13.11 | [TK]D-Fender | GAH |
| 18:13.16 | [TK]D-Fender | can't type for beans today |
| 18:13.34 | Bartockbatz | No talking about shoving - you know how long it's been ?? |
| 18:13.36 | Bartockbatz | :) |
| 18:13.40 | *** join/#asterisk darksk1ez (~mhb@darkskiez-1-pt.tunnel.tserv5.lon1.ipv6.he.net) |
| 18:14.00 | Bartockbatz | anywho - let me look that up - see if I can make sense |
| 18:24.16 | *** join/#asterisk iscario (~quassel@24.244.71-86.rev.gaoland.net) |
| 18:25.39 | *** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de) |
| 18:26.25 | iscario | hello, i just wanted to know which value to assign to the parameter 'allow' in iax.conf in order to be able to record a conversation in wav format instead of gsm ? |
| 18:28.30 | [TK]D-Fender | iscario: IRRELEVENT |
| 18:28.45 | [TK]D-Fender | iscario: * records in the format you tell it to when you call the app in the DIALPLAN |
| 18:28.59 | *** join/#asterisk REdOG (~REdOG@gentoo/user/redog) |
| 18:29.42 | iscario | [TK]D-Fender: oh, is it mixmonitor who decide that ? |
| 18:30.07 | [TK]D-Fender | iscario: No, it is YOU who decides when you call that app |
| 18:30.20 | markfeatherston_ | iscario: http://www.voip-info.org/wiki/view/Asterisk+cmd+Record |
| 18:30.31 | markfeatherston_ | iscario: you say what format you want when you call record |
| 18:31.14 | [TK]D-Fender | markfeatherston_: Extremely wrong application |
| 18:31.20 | markfeatherston_ | ? |
| 18:31.33 | markfeatherston_ | didn't know what |
| 18:31.38 | [TK]D-Fender | markfeatherston_: Didn't think thqat left any room for misinterpretation. |
| 18:31.46 | *** join/#asterisk outtolunc (~me@c-98-248-105-248.hsd1.ca.comcast.net) |
| 18:31.52 | [TK]D-Fender | markfeatherston_: he wants to record the CALL |
| 18:32.33 | markfeatherston_ | so, monitor? |
| 18:33.05 | iscario | [TK]D-Fender: sorry, i don't understand. In my Dialplan, i call mixMonitor(), when do i have to tell it the format ? |
| 18:33.24 | [TK]D-Fender | iscario: "core show application mixmonitor" <- |
| 18:37.25 | *** join/#asterisk mercutioviz (~michaelco@freeswitch/developer/msc) |
| 18:37.57 | REdOG | woo hoo I have calls coming in! |
| 18:38.37 | *** join/#asterisk QubeZ (~nkasu@110.67.204.68.cfl.res.rr.com) |
| 18:38.43 | QubeZ | hello all |
| 18:38.44 | iscario | [TK]D-Fender: damnit , it is not installed on this computer. Is it just the extension that it is chosen who allow to choose the right format ? if no, i ' ll read at home the man when i'll be back ;) |
| 18:38.45 | REdOG | when I get a call I see "Call from '' to extension ... " in the console, why would the '' be empty? |
| 18:38.59 | REdOG | seems like CID would be there |
| 18:39.06 | [TK]D-Fender | iscario: PARDON? |
| 18:39.12 | [TK]D-Fender | iscario: WHAT isn't "installed"? |
| 18:39.23 | [TK]D-Fender | redunless they don't have one. |
| 18:39.28 | QubeZ | i am running Asterisk 1.4.26.2 and my chanspy module is only allow me to monitor one extension after logging then i have to cancel the session and restart it to monitor another extension. Anyone heard of this issue? Rather than having the ability to go from one ext to another? |
| 18:39.36 | [TK]D-Fender | REdOG: LIke... blocked number. Or just not available |
| 18:39.59 | REdOG | broadvoice seems a bit dodgy |
| 18:40.23 | REdOG | I get non stop timeout attempts |
| 18:40.36 | iscario | [TK]D-Fender: sorry, i meant : http://pastebin.org/385479 |
| 18:40.37 | REdOG | but when I first fire up asterisk my calls seem to come in |
| 18:40.38 | *** part/#asterisk mercutioviz (~michaelco@freeswitch/developer/msc) |
| 18:41.20 | *** join/#asterisk retentiveboy (~pdugas@74-95-28-37-Atlanta.hfc.comcastbusiness.net) |
| 18:41.57 | [TK]D-Fender | iscario: Congratulations. Now how did you install *? |
| 18:44.29 | iscario | [TK]D-Fender: can't remember, it is just my laptop here, i installed it few weeks ago^^ i don't know why it is like that, but it doesn't really matter , i'll have a look on my real server |
| 18:45.10 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 18:46.01 | [TK]D-Fender | iscario: Yes, it definitely helps to read the instructions |
| 18:46.47 | iscario | [TK]D-Fender: ;) thanks anyway |
| 18:49.01 | *** part/#asterisk rossand (~aross@CPE485b390978ce-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 18:50.28 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 18:50.51 | bran | so weird, i can call landline numbers thru the skype extension, but i can't hear anything |
| 18:51.51 | REdOG | ok i am being blocked |
| 18:52.12 | REdOG | what setting would cause asterisk to keep trying to register even after it has? |
| 18:53.19 | bran | how do I install this g729 codec? |
| 18:55.30 | *** join/#asterisk LemensTS (~LemensTS@adsl-70-238-159-189.dsl.stlsmo.sbcglobal.net) |
| 18:56.04 | LemensTS | prefered polycom phone for phone to phone intercom? |
| 18:56.36 | REdOG | is the config insecure=very still valid for *-1.6.2.9 ? |
| 18:56.49 | LemensTS | and do you have to modify the .cfg of the phone for intercom to work |
| 18:57.44 | Naikrovek | LemensTS: all |
| 18:57.58 | Naikrovek | i do need to find a voip phone with a louder headset and/or speaker though |
| 18:58.43 | Naikrovek | and yes you need to tell the phone you want it to answer the intercom type of call |
| 18:58.51 | LemensTS | Nakrovek: is intercom paging something i can code in asterisk 1.6 to work with teh polycoms, or do i have to make a boot server and modify the .cfg of the polycoms to do that? (long time ago seemed i did that to make it work) |
| 18:58.53 | Naikrovek | Ring Answer, it's called |
| 18:58.55 | [TK]D-Fender | redDecices re-register on a demanded frequency you know... |
| 18:59.16 | Naikrovek | yes you need to modify a config somewhere (or create one to override default) |
| 18:59.23 | Naikrovek | but it's a simple change |
| 18:59.27 | [TK]D-Fender | LemensTS: Yes you need to provision them |
| 18:59.37 | [TK]D-Fender | REdOG: Decices re-register on a demanded frequency you know... |
| 18:59.42 | [TK]D-Fender | devices* |
| 18:59.49 | LemensTS | Nakrovek: TK: Thanks, didn't know if thats how it was still or not. |
| 18:59.57 | Naikrovek | still is |
| 19:00.03 | Naikrovek | but its an easy change |
| 19:00.15 | Naikrovek | i put my config up on pastebin some time ago --- wonder if i can still find it |
| 19:00.16 | REdOG | [TK]D-Fender: the company is blocking me because I send too many registrations |
| 19:00.17 | [TK]D-Fender | LemensTS: Hasn't changed, is unlikely to ever do so |
| 19:01.00 | REdOG | can I manipulate the frequency? |
| 19:01.17 | *** part/#asterisk LemensTS (~LemensTS@adsl-70-238-159-189.dsl.stlsmo.sbcglobal.net) |
| 19:01.18 | bran | does g729 come free with my SFA license? |
| 19:01.18 | Naikrovek | how many are you sending... |
| 19:01.41 | REdOG | not that many tbh |
| 19:01.47 | Naikrovek | how many are you sending... |
| 19:01.50 | [TK]D-Fender | REdOG: registertimeout=X |
| 19:01.50 | pabelanger | bran: I believe so |
| 19:01.58 | REdOG | every 20 seconds it looks like |
| 19:02.07 | Naikrovek | how many phones |
| 19:02.16 | [TK]D-Fender | Naikrovek: Not Applicable |
| 19:02.17 | REdOG | just 1 channel |
| 19:02.21 | Naikrovek | oh |
| 19:02.22 | Naikrovek | k |
| 19:02.22 | REdOG | no phones yet |
| 19:02.23 | [TK]D-Fender | Naikrovek: * > ITSP reg interval |
| 19:02.34 | Naikrovek | gotcha |
| 19:02.35 | bran | pabelanger: is g729 automatically installed when I installed SFA? |
| 19:02.36 | [TK]D-Fender | REdOG: See above |
| 19:02.40 | REdOG | ryt |
| 19:02.41 | REdOG | tks |
| 19:03.39 | Naikrovek | in other news, i just discovered half-life and team fortress 2 thanks to the steam summer sale |
| 19:03.42 | Naikrovek | in a word: whoa |
| 19:04.33 | REdOG | thats a time sink for me |
| 19:04.46 | Naikrovek | no kidding |
| 19:05.20 | pabelanger | bran: cannot remember been awhile since I used SFA. Contact Digium |
| 19:05.57 | Naikrovek | the product page should say so |
| 19:06.08 | Naikrovek | and i would guess no because they sell G729 as well |
| 19:06.37 | bran | how do I check which modules are loaded in asterisk? |
| 19:07.07 | [TK]D-Fender | bran: "core show modules" |
| 19:07.19 | bran | No such command 'core show modules' (type 'core show help core show' for other possible commands) |
| 19:07.34 | [TK]D-Fender | bran: "modules show " |
| 19:07.41 | Naikrovek | lol "core show help core show" |
| 19:07.43 | bran | No such command 'modules show' (type 'core show help modules show' for other possible commands) |
| 19:07.48 | Naikrovek | wtf |
| 19:07.57 | Naikrovek | what version are you using |
| 19:07.58 | [TK]D-Fender | bran: "help" |
| 19:08.01 | *** join/#asterisk githogori (~githogori@SJC-Office-DHCP-135.mail-abuse.org) |
| 19:08.01 | [TK]D-Fender | <PROTECTED> |
| 19:08.07 | bran | no help either |
| 19:08.12 | bran | im using 1.6.2.7 |
| 19:08.19 | [TK]D-Fender | braNo help? |
| 19:08.19 | Naikrovek | is asterisk running |
| 19:08.21 | [TK]D-Fender | Pardon? |
| 19:08.33 | bran | yeah no help |
| 19:08.34 | bran | lmao |
| 19:08.37 | bran | this is fucked |
| 19:09.08 | beek | notes the technical term used to describe the problem. |
| 19:09.38 | bran | Connected to Asterisk 1.6.2.7 currently running on ps (pid = 3625) |
| 19:09.38 | bran | Verbosity is at least 10 |
| 19:09.50 | bran | ps*CLI> help |
| 19:09.51 | bran | No such command 'help' (type 'core show help help' for other possible commands) |
| 19:10.19 | bran | sweet core show help is working |
| 19:12.34 | *** join/#asterisk spiceycurry (~mcurry@proxy.hostopia.com) |
| 19:12.40 | *** join/#asterisk outtolunc (~me@c-98-248-105-248.hsd1.ca.comcast.net) |
| 19:12.52 | *** join/#asterisk rocksfrow (~kyle@74-92-146-97-WashingtonDC.hfc.comcastbusiness.net) |
| 19:14.08 | *** part/#asterisk spiceycurry (~mcurry@proxy.hostopia.com) |
| 19:14.37 | *** join/#asterisk DaveCanoe (~Dave@strike.eicat.ca) |
| 19:15.22 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 19:16.08 | rocksfrow | can anybody assist me in confirming that long distance is not working? |
| 19:16.18 | rocksfrow | i had random fax issues |
| 19:16.23 | tzafrir_laptop | bran, /etc/asterisk/cli_aliases.conf may be useful |
| 19:16.23 | rocksfrow | and now random outbound calling issues |
| 19:16.30 | rocksfrow | and now i'm thinking something is up with long distance |
| 19:16.34 | rocksfrow | bc local calls seem to work fine |
| 19:16.38 | rocksfrow | but long distance aren't... |
| 19:16.43 | rocksfrow | how can i confirm this before calling my telco? |
| 19:16.52 | bran | tzafrir_laptop: i dont have that file |
| 19:17.04 | bran | where can i find a document with the latest commands for asterisk 1.6.2 |
| 19:17.05 | rocksfrow | is there anything within asterisk that would cause it to suddenly start giving busy signals on a long distance call? |
| 19:17.09 | tzafrir_laptop | look at the sample one |
| 19:17.14 | bran | all the random snippets i find online are for older version or something |
| 19:18.04 | tzafrir_laptop | IIRC the sample one aliases 'help' to 'core show help' |
| 19:18.09 | rocksfrow | toll free #'s work too |
| 19:18.45 | tzafrir_laptop | bran, configs/cli_alises.conf in the source directory |
| 19:18.50 | bran | tzafrir_laptop: i don't mind typing the whole core show help command lol, as long as it works |
| 19:19.34 | [TK]D-Fender | rocksfrow: CALL |
| 19:20.31 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 19:20.41 | *** join/#asterisk _mwoodj_ (~mwoodj@pdpc/sponsor/digium/hyper-eye) |
| 19:21.47 | *** join/#asterisk veryhappy (~veryhappy@188.28.43.141.threembb.co.uk) |
| 19:25.45 | rocksfrow | [TK]D-Fender, by CALL, do you mean call the telco? |
| 19:25.59 | rocksfrow | i'm creating a trouble ticket now, i'm just always afraid i'm going to report an issue that is being caused by asterisk |
| 19:26.11 | rocksfrow | the error log says 'other hangup' |
| 19:26.18 | [TK]D-Fender | rocksfrow: "how do I know if its working" <- TRY IT |
| 19:26.31 | [TK]D-Fender | rocksfrow: Fuck logs. TRY NOW <- |
| 19:27.49 | rocksfrow | [TK]D-Fender, lol |
| 19:27.54 | rocksfrow | [TK]D-Fender, i can pastebin |
| 19:28.26 | rocksfrow | [TK]D-Fender, i previously thought this issue was an issue with FAX |
| 19:28.41 | rocksfrow | simply bc by random chance, we were faxing long distance more often than we were calling long distance |
| 19:28.49 | *** join/#asterisk voxter (~voxter@76.77.73.130) |
| 19:29.04 | rocksfrow | i'm actually quite pleased to learn its an issue with long distance in general, that probably means my fax will start working again when i fix this |
| 19:29.13 | *** join/#asterisk fofware (fabian@host149.190-31-55.telecom.net.ar) |
| 19:29.44 | [TK]D-Fender | rocksfrow: "thinking" tends to be worthless. LOOKING is important. |
| 19:29.45 | rocksfrow | [TK]D-Fender, http://pastebin.com/itJJTpp0 |
| 19:29.54 | rocksfrow | [TK]D-Fender, you'll see what i mean when you see the log |
| 19:34.55 | *** join/#asterisk jonmasters (~jcm@dallas.jonmasters.org) |
| 19:36.06 | *** join/#asterisk veryhappy (~no@188.28.96.66.threembb.co.uk) |
| 19:37.01 | veryhappy | sorry last time as i tried my internet connection broke |
| 19:37.44 | rocksfrow | [TK]D-Fender, any clue? |
| 19:37.46 | veryhappy | perhaps someone can suggest me a nice easy tutorial for asterisk also with installation of sip xlite |
| 19:39.06 | REdOG | veryhappy: http://asteriskdocs.org ? |
| 19:39.27 | REdOG | It's what ive been using |
| 19:39.58 | [TK]D-Fender | veryhappy: a usable SIP peer definition is about 6 lines in sip.conf. teh rest is dialplan |
| 19:43.03 | [TK]D-Fender | rocksfrow: Presentation: Presentation permitted, user number passed network screening (1) '5110' ] <-- why are you sending a bogus CID # to the PSTN? |
| 19:43.23 | [TK]D-Fender | rocksfrow: Toll-free's should tell you to GTFO, etc. |
| 19:43.28 | [TK]D-Fender | rocksfrow: its asking for trouble |
| 19:43.34 | rocksfrow | toll frees and locals work |
| 19:43.37 | rocksfrow | everything has been working fine |
| 19:43.41 | rocksfrow | i imagine thats some weird config |
| 19:44.02 | rocksfrow | long distance has been working before this |
| 19:49.02 | rocksfrow | [TK]D-Fender, i will configure a default CID on the route to try to avoid that from happening |
| 19:49.08 | rocksfrow | but that is not why long distance is not working |
| 19:50.38 | *** join/#asterisk veryhappy (~Ben@92.40.158.134.sub.mbb.three.co.uk) |
| 19:52.01 | *** join/#asterisk hohum (dcorbe@apollo.corbe.net) |
| 19:53.10 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 19:53.11 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 19:53.20 | Naikrovek | just replaced my aged postfix server in about 30 minutes, from decision to do it to completion |
| 19:53.28 | Naikrovek | not sure if that's quick but it's certainly quick for me |
| 19:53.36 | Naikrovek | seeing as how it's a relay for exchange only |
| 19:54.11 | rocksfrow | [TK]D-Fender, > Presentation: Presentation permitted, user number passed network screening (1) '4108145945' ] |
| 19:54.17 | rocksfrow | [TK]D-Fender, that better? :) |
| 19:54.25 | rocksfrow | [TK]D-Fender, still same busy signal, though |
| 19:54.56 | rocksfrow | [TK]D-Fender, but thanks for that tip, glad i got that straightened out |
| 19:55.13 | rocksfrow | i have a support ticket open with the telco on it |
| 19:56.28 | rocksfrow | q931_hangup: other hangup |
| 19:56.28 | rocksfrow | NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Outgoing Call Proceeding, peerstate Incoming Call Proceeding, hold-state Idle |
| 19:56.28 | rocksfrow | q931.c:4686 q931_disconnect: Call 32771 enters state 11 (Disconnect Request). Hold state: Idle |
| 19:56.50 | rocksfrow | you think i should try powering off/on the PRI equipment |
| 19:56.50 | rocksfrow | ? |
| 19:57.52 | veryhappy | hello ... have a question can you please suggest me an easy tutorial for asterisk installation perhaps with sip and x-lite ... thank you |
| 19:58.11 | [TK]D-Fender | rocksfrow: Could be they are actually BUSY |
| 19:58.25 | [TK]D-Fender | veryhappy: ... |
| 19:58.27 | [TK]D-Fender | ~book |
| 19:58.28 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 19:58.29 | [TK]D-Fender | ^^^^^^^ |
| 19:58.34 | [TK]D-Fender | ~jerjerguide |
| 19:58.34 | infobot | [~jerjerguide ] Jeremy McNamara's quick sample guide to setting up * : http://www.jeremy-mcnamara.com/2007/02/26/how-to-configure-asterisk-your-first-installation/ |
| 19:58.42 | Naikrovek | jerjer? |
| 19:58.47 | [TK]D-Fender | ^^^^^^ For added "inspiration" about a more complete simple setup |
| 19:58.50 | rocksfrow | [TK]D-Fender, they aren't busy |
| 19:58.52 | rocksfrow | 6 diff numbers? |
| 19:59.00 | rocksfrow | that i call on my cell phone immediately after and they all ring |
| 19:59.04 | rocksfrow | heh.. |
| 19:59.14 | rocksfrow | you seem to miss that this has been working for 6 months now |
| 19:59.22 | rocksfrow | an install manual probably isnt my answer at this point |
| 19:59.42 | rocksfrow | could the fact that it happened so randomly mean a hardware issue? |
| 19:59.42 | [TK]D-Fender | rocksfrow: I trust NOTHING about claims to changes made/not made... have to judge what I see in front of me. |
| 19:59.56 | [TK]D-Fender | rocksfrow: I don't do "time travel" or "story time" |
| 19:59.58 | rocksfrow | [TK]D-Fender, so what does that log tell you? what is that 'other hangup' |
| 20:00.03 | rocksfrow | there isnt much documentation on that |
| 20:00.09 | rocksfrow | [TK]D-Fender, ..lol |
| 20:00.24 | veryhappy | thank you i will try it |
| 20:00.49 | rocksfrow | [TK]D-Fender, oops did not see veryhappy chime in there, lol |
| 20:01.00 | rocksfrow | [TK]D-Fender, is there any other info that would help? |
| 20:01.16 | rocksfrow | [TK]D-Fender, but i'm positive the other end is not busy |
| 20:01.23 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 20:01.30 | rocksfrow | toll frees work, locals work, long distance gives me busy |
| 20:01.52 | veryhappy | if this works then im [veryhappy] |
| 20:01.53 | veryhappy | :D |
| 20:02.32 | veryhappy | i'll call myself and wonder why nobody is on the other side... sure NOT :D |
| 20:02.54 | veryhappy | good thank you |
| 20:02.57 | veryhappy | by |
| 20:03.00 | veryhappy | bye |
| 20:03.12 | *** part/#asterisk veryhappy (~Ben@92.40.158.134.sub.mbb.three.co.uk) |
| 20:04.13 | *** join/#asterisk mindCrime (~chatzilla@64.134.184.80) |
| 20:06.15 | *** join/#asterisk eject_ck (~eject_ck@85.223.182.86) |
| 20:06.44 | eject_ck | Hi guys, can someone recommend SIP hardware phone with PPTP client ? |
| 20:07.12 | Naikrovek | pptp? |
| 20:07.13 | eject_ck | ISP banned VoIP |
| 20:07.23 | Naikrovek | what country are you in |
| 20:07.25 | eject_ck | Naikrovek: yes, pptp |
| 20:07.48 | eject_ck | Naikrovek: does it matter ? For suggestion ? |
| 20:07.56 | Naikrovek | just curious |
| 20:08.01 | Naikrovek | doesn't really matter |
| 20:08.01 | eject_ck | UA |
| 20:08.26 | Naikrovek | hmm |
| 20:08.50 | eject_ck | I have cisco 7960 and not able to use it |
| 20:09.15 | Naikrovek | can you tunnel the voip through something else |
| 20:09.28 | eject_ck | there are couple of ways for me: 1) router with pptp client; 2) VoIP phone with pptp client; 3) PATA with pptp client |
| 20:10.09 | eject_ck | yes, I can in theory but really I don`t have devices which support pptp |
| 20:10.34 | eject_ck | so maybe some one know any above for ~ 100$ |
| 20:11.13 | Naikrovek | i don't know of any myself but there might be some |
| 20:13.10 | Naikrovek | gov'ts banning voip is a pretty good benchmark for super-corruption |
| 20:13.13 | Naikrovek | at least in my mind |
| 20:13.20 | Naikrovek | reality always seems to be different |
| 20:14.44 | *** join/#asterisk tsalvador (~tsalvador@207.88.49.98) |
| 20:16.03 | *** join/#asterisk bmoraca_work (~bmoraca@66-242-174-254.ceres.bvn.net) |
| 20:17.40 | [TK]D-Fender | Naikrovek: Not a sign of corruption. Buy influence to circumvent laws would be. |
| 20:17.55 | [TK]D-Fender | Naikrovek: That just makes them opressive :) |
| 20:18.53 | Naikrovek | yeah i guess |
| 20:21.02 | *** join/#asterisk blaines (~blaines@ip70-190-67-126.ph.ph.cox.net) |
| 20:23.24 | *** join/#asterisk elliot98 (~elliot@unaffiliated/elliot98) |
| 20:23.45 | elliot98 | slides into his place |
| 20:23.48 | eject_ck | [TK]D-Fender: do you have in mind device what I need ? |
| 20:25.12 | markfeatherston_ | eject_ck: you're almost definately going to be better off using a router that supports it, or set up a dedicated box that just handles the vpn that you can forward the routes to. |
| 20:25.20 | elliot98 | sometimes when I try to authenticate a device using a static IP address, the device does not properly get idenified...it just gets thrown into the default contet |
| 20:25.26 | elliot98 | *context |
| 20:25.26 | [TK]D-Fender | eject_ck: Plenty fo devices out there. None I would personally consider |
| 20:25.38 | [TK]D-Fender | eject_ck: You could jsut go for a cheap netbook and soft-phone |
| 20:26.12 | elliot98 | what would be causing this behavior? |
| 20:27.01 | [TK]D-Fender | elliot98: Can't say, because we can't SEE |
| 20:29.31 | *** join/#asterisk rustyclarkson (~rusty@u53.sutus.com) |
| 20:29.32 | raden_work | elliot98, pastebin ? |
| 20:29.44 | raden_work | shrinks boss with shrink ray |
| 20:31.08 | [TK]D-Fender | checkout time, BBIAB |
| 20:32.23 | *** join/#asterisk diegomad (~mad@190.146.200.120) |
| 20:32.28 | eject_ck | :( |
| 20:32.44 | eject_ck | plenty != work for me |
| 20:33.00 | eject_ck | thanks anyway |
| 20:34.16 | Katty | :< |
| 20:34.22 | Katty | here squirrely squirrely squirrel:< |
| 20:34.27 | Katty | dinner is served :< |
| 20:35.09 | *** join/#asterisk psilikon (~j@cerberus.vicimarketing.com) |
| 20:35.56 | markfeatherston_ | eject_ck: why is handling the vpn on the router side not an option? |
| 20:36.23 | raden_work | heya Katty |
| 20:36.27 | Katty | hi raden |
| 20:36.35 | raden_work | how are things going ? |
| 20:36.49 | Katty | a bit rocky at the moment |
| 20:36.59 | raden_work | oh ? |
| 20:37.03 | Katty | mhmm |
| 20:37.11 | raden_work | why dat |
| 20:37.40 | Katty | raden_work: -> |
| 20:38.01 | tsalvador | anyone know what file i need to use to config VM |
| 20:38.30 | Katty | voicemail.conf |
| 20:38.48 | *** join/#asterisk freeedrich| (~eeePC@hansaserver.de) |
| 20:38.49 | Katty | Qwell: ping |
| 20:39.52 | *** join/#asterisk retentiveboy (~pdugas@69.169.199.82) |
| 20:40.18 | tsalvador | katty, what about a log file to check why calls are disconnecting right after the VM greeting? |
| 20:40.54 | markfeatherston_ | tsalvador: /var/log/asterisk/full |
| 20:41.14 | markfeatherston_ | you might need to turn up the verbose level to get what you want |
| 20:41.47 | tsalvador | i don't have /full |
| 20:42.32 | SiNGLer | enable it in logger.conf |
| 20:48.41 | markfeatherston_ | How do I list the channel variables from cli? |
| 20:51.31 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 20:52.15 | Trixboxer | for connecting more than 3 sites ( all to each other ) is IAX2 better or SIP trunk? currently both are working |
| 20:52.58 | Katty | hi Chainsaw |
| 20:53.06 | Chainsaw | Hello Katty :) |
| 20:53.33 | *** join/#asterisk Sheeplet (~BuRn@layer0.datahive.org) |
| 20:53.44 | rocksfrow | Chainsaw, do you remember helping me debug my "fax" issues the other day? |
| 20:53.48 | *** join/#asterisk _pepo_ (c837ea09@gateway/web/freenode/ip.200.55.234.9) |
| 20:53.50 | Sheeplet | lo all |
| 20:53.51 | rocksfrow | the 'other hangup' debug message? |
| 20:54.01 | _pepo_ | hi friends |
| 20:54.09 | rocksfrow | Chainsaw, turns out its a long distance issue in general, not fax |
| 20:54.13 | Chainsaw | rocksfrow: To a degree, yes. |
| 20:54.52 | rocksfrow | Chainsaw, local and toll free calls are working fine |
| 20:54.55 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 20:54.58 | rocksfrow | Chainsaw, but long distance is giving me a busy signal |
| 21:00.18 | Kobaz | so... is it bad when the telco tells you it's going to take 30 days to install a T1... and then they ask for the programming details. And then three weeks go by and they ask again for the programming details... and then two weeks later they still haven't actually started the installation |
| 21:02.07 | markfeatherston_ | do you have anything in your contract saying they have to have it completed in 30 days? |
| 21:02.27 | Kobaz | generally we get the contract when the thing is installed |
| 21:02.40 | Kobaz | or right when the guy comes to turn it up |
| 21:02.57 | Kobaz | i think it has to do with this buyout going on |
| 21:03.12 | Kobaz | windstream bought dne communications, and it's been mostly hell |
| 21:03.33 | Kobaz | customers t1 lines are going up and down randomly |
| 21:03.41 | markfeatherston_ | yikes |
| 21:04.12 | markfeatherston_ | find someone else or give them a deadline of today/tomorrow |
| 21:04.16 | Kobaz | heh yeah |
| 21:04.24 | Kobaz | the problem is, verizon is like 200 more a month |
| 21:06.03 | *** join/#asterisk retentiveboy (~pdugas@69.169.199.82) |
| 21:17.01 | rocksfrow | Chainsaw, it was a telco issue |
| 21:17.06 | rocksfrow | they just fixed it! |
| 21:17.12 | Chainsaw | rocksfrow: Awesome :) |
| 21:17.19 | rocksfrow | they made a typo in the billing # |
| 21:17.28 | rocksfrow | 914 instead of 814 |
| 21:17.33 | rocksfrow | idiots. |
| 21:17.52 | rocksfrow | Chainsaw, atleast now i have fax to email setup too :-) heh |
| 21:18.11 | Chainsaw | rocksfrow: *G* So close, yet so far. |
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| 21:27.38 | REdOG | well that didn't help... |
| 21:27.43 | REdOG | fkn broadvoice |
| 21:28.16 | REdOG | 45 minute wait each time I start up * |
| 21:28.48 | markfeatherston_ | ? |
| 21:28.50 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 21:29.03 | markfeatherston_ | I didn't know a 486 would run * |
| 21:29.08 | markfeatherston_ | why is it so slow to boot? |
| 21:29.14 | markfeatherston_ | or is that to register? |
| 21:29.32 | REdOG | it registers right away but then keeps trying to register |
| 21:29.33 | Corydon76-dig | I'm sure you could get a 486 to run Asterisk |
| 21:29.51 | REdOG | after the 5th or so attempt they say they are blocking my ip |
| 21:29.52 | Corydon76-dig | good luck on transcoding or anything else that takes up CPU |
| 21:29.58 | markfeatherston_ | heh |
| 21:30.07 | REdOG | but my softphone still registers |
| 21:30.11 | REdOG | is confused |
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| 21:33.50 | boodu | hello |
| 21:34.02 | markfeatherston_ | howdy |
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| 21:42.45 | joako | I enter asterisk queue and it says " your time on hold should be 1 minute 110 seconds" how does that make sense? |
| 21:43.28 | pabelanger | joako: Because Wookies live on Endor? |
| 21:43.40 | markfeatherston_ | I walked into a fast food restaraunt and they said "please wait sir". Why did they make me wait? |
| 21:43.55 | markfeatherston_ | joako: we need details :P |
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| 21:44.55 | joako | why would it say 1 minute 110 seconds? shouldn't it say 2 minutes 50 seconds? |
| 21:45.25 | markfeatherston_ | wait, i missed that. I've never heard that in my queue |
| 21:45.26 | mocker | joako: It takes 115 seconds for asterisk to calculate a minute. |
| 21:45.34 | REdOG | lol |
| 21:45.40 | markfeatherston_ | lol |
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