| 02:52.43 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 02:52.43 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.0.0-rc2 (2011/11/15), 1.8.7.1 (2011/10/17), dahdi-linux 2.5.0.2 (2011/10/21), dahdi-tools 2.5.0.2 (2011/10/21), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 02:54.41 | p3nguin | And then infobot comes back. |
| 02:55.08 | WIMPy | Just in time for monday morning. |
| 02:56.47 | LiuYan | who is the owner of infobot? |
| 02:57.02 | Russ | bzflag |
| 02:57.07 | Russ | (I would assume) |
| 02:57.08 | p3nguin | rikers |
| 02:58.09 | Russ | p3nguin, its actually just 'riker' |
| 02:58.32 | p3nguin | Yeah, you're right. |
| 02:58.42 | p3nguin | Riker |
| 02:59.25 | LiuYan | thanks |
| 03:00.08 | LiuYan | is res_config_ldap.so missed in offcial yum repository? |
| 03:01.54 | LiuYan | ~ads |
| 03:01.54 | infobot | please don't advertise in #asterisk. it's not effective anyhow. |
| 03:02.20 | LiuYan | ~ActiveDirectoryServer |
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| 05:32.05 | OldMonk | anyone know where i can find out the meaning of the various numbers "queue show FOO" displays? (e.g. W:0, C:12, A:9, SL:0.0% within 0s) |
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| 05:52.46 | phiscribe | would a PII 367mhz 192mb ram be enough for 3-4 phones? |
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| 06:38.43 | kaushal | Hi |
| 06:39.02 | kaushal | I have set inbound facility in asterisk |
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| 06:49.40 | kaushal | when i make an inbound call i get http://pastebin.ubuntu.com/744954/ |
| 06:49.50 | kaushal | Any clue ? |
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| 06:54.29 | [TK]D-Fender | kaushal: what are you doing passing an alaw file .... to MP#player? |
| 06:54.38 | [TK]D-Fender | mp3player <- |
| 07:02.59 | irroot | hi folks |
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| 07:11.44 | kaushal | [TK]D-Fender: please give me a moment |
| 07:12.23 | dym | Mornings ladies. |
| 07:13.56 | ChannelZ | hey hey cocoa-muffins |
| 07:16.27 | dym | <3 |
| 07:16.34 | dym | ellow sweetcheeks |
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| 07:22.12 | schmidts | good morning |
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| 07:40.36 | SeRi | g/n #asterisk. done studying... Calling it a night.... |
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| 07:48.33 | dym | :D |
| 07:48.38 | dym | SeRi: nn |
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| 09:20.22 | polysics | hello |
| 09:20.33 | polysics | scenario: user A calls user B, user B answers |
| 09:20.34 | mandla | hi |
| 09:20.43 | polysics | user A wants to bring user C in the conversation |
| 09:20.53 | polysics | how can A dial C and join them all? |
| 09:21.10 | polysics | i am using Adhearsion but i am not familiar with the concept itself in Asterisk |
| 09:21.20 | mandla | polysics, have you configured conferencing?? |
| 09:21.26 | polysics | not yet |
| 09:21.30 | polysics | i am on asterisk 1.8, btw |
| 09:21.44 | polysics | the call between A and B is a "normal" call though |
| 09:22.13 | mandla | polysics, Dont know dude. |
| 09:22.36 | mandla | polysics, but that i manage it with call conferencing. |
| 09:22.46 | mandla | irroot, ?? |
| 09:24.03 | mandla | WARNING[3534]: app_dial.c:1098 wait_for_answer: Unable to forward voice or dtmf |
| 09:24.15 | mandla | What causes this error, WARNING[3534]: app_dial.c:1098 wait_for_answer: Unable to forward voice or dtmf |
| 09:27.16 | polysics | hmm, to start, how do i detect in-band DTMF? |
| 09:27.34 | polysics | i was thinking of having A press # followed by the ID of the wanted C |
| 09:27.35 | kaldemar | polysics: features.conf |
| 09:28.06 | kaldemar | polysics: use application map to make an extension that puts all three parties to a conference. |
| 09:28.31 | polysics | all this is in 1.8, right? |
| 09:29.48 | kaldemar | yes. this is one way to do it: http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO |
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| 09:33.25 | polysics | can i put an AGI call in the application map somehow, please? |
| 09:33.37 | polysics | i am using Adhearsion and would like to stay in there |
| 09:33.46 | polysics | or just put the AGI app in the macro? maybe it's easier |
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| 09:34.05 | polysics | also, is the application map stanza in features.conf enough to turn on inline DTMF? |
| 09:35.03 | kaldemar | see the IMPORTANT NOTE part in the sample features conf. |
| 09:35.12 | polysics | i would even be happy with just being able to see in band DTMF in AMI events and take its from there |
| 09:35.38 | polysics | the sample shipped with 1.8? |
| 09:35.39 | kaldemar | you need to set the DYNAMIC_FEATURES variable also. |
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| 09:35.58 | polysics | i mean, not in the page you linked me as there aren't any :-) |
| 09:36.21 | kaldemar | "in the sample features conf" |
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| 09:39.46 | polysics | this will make me look like a noob |
| 09:40.04 | polysics | but i if have a working asterisk install, and don't want to make samples because i have my own config |
| 09:40.10 | polysics | where do i find a sample? :-) |
| 09:40.31 | kaldemar | in the asterisk tarball for example. configs/ |
| 09:41.03 | kaldemar | or here: http://svn.digium.com/svn/asterisk/tags/1.8.7.1/configs/features.conf.sample |
| 09:41.56 | polysics | i think it could just be easier to find a way to get in band DTMF tones into AMI events |
| 09:42.06 | polysics | but i will read through this, thanks |
| 09:47.14 | kaldemar | you should get events in AMI if dtmf class is configured for the manager user and events are on. |
| 09:50.18 | polysics | even the in-band ones? without any features.conf intervention? |
| 09:50.44 | kaldemar | if inband dtmf is configured for the channel, yes. |
| 09:54.46 | polysics | and where does that configuration reside, please? |
| 09:55.33 | polysics | by the way |
| 09:55.47 | polysics | i am not even sure "in band" means what i think it means :-) |
| 09:56.14 | polysics | i noticed i do not get DTMF events when inside a bridged call between two SIP users |
| 09:56.22 | polysics | while i do get them in a menu or before they bridge |
| 09:57.50 | polysics | i probably need them to be in band though as some of the users will not be on a SIP soft phone but calling in through our PSTN number through the SIP provider |
| 09:57.52 | kaldemar | manager.conf |
| 09:58.02 | polysics | if i am not making any sense tell me :-) |
| 09:58.09 | kaldemar | inband is sent as audio. |
| 09:58.18 | polysics | i have read= all and write= all |
| 09:58.36 | polysics | maybe all does not include dtmf? :-) |
| 09:58.37 | kaldemar | recognizing it requires asterisk to stay on the media path, you'd need to disable directmedia in sip.conf. |
| 09:59.02 | polysics | and that could bring up some problems somewhere else, i fear |
| 09:59.07 | kaldemar | all includes dtmf. you probably have the audio going directly between the phones in a call, so asterisk has no way of detecting it. |
| 09:59.22 | polysics | as we are using a pretty complex tunneling system to bypass firewalls |
| 09:59.29 | kaldemar | it might. |
| 09:59.43 | polysics | that's something i do not want to mess up :-) |
| 10:00.03 | polysics | so i am basically dead in the water? |
| 10:00.37 | kaldemar | not necessarily. try it. |
| 10:01.03 | polysics | i got lost :-D try what, please? |
| 10:01.15 | polysics | disabling directmedia? |
| 10:01.18 | kaldemar | disable directmedia |
| 10:01.31 | polysics | if it's the only way i can try that |
| 10:01.36 | kaldemar | but you really shouldn't use inband dtmf anyway. |
| 10:02.07 | polysics | the original requirement is "user A calls user B. User A wants to bring user C into the call using the phone digits." |
| 10:02.14 | polysics | the conference itself is another problem |
| 10:02.32 | polysics | let's generalize it as "user A wants to send arbitrary commands to the * server" |
| 10:02.51 | polysics | could even be something totally different like "email something to someone" |
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| 10:04.45 | polysics | i suppose it is not the first time someone wants to do this - might just be to park a call |
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| 10:27.23 | polysics | no ideas, please? |
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| 10:36.00 | wdoekes2 | polysics: didn't kaldemar suggest you switch dtmf mode from inband/rfc2833 to info? |
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| 10:36.30 | polysics | i would say he suggested other things and not that, but i might be thick-skulled |
| 10:37.08 | kaldemar | well, i hinted that you should not be using inband dtmf. |
| 10:42.10 | wdoekes2 | (I don't know if it works though.. I don't know how to tell my phone to do INFO, it will probably do rfc2833 regardless) |
| 10:49.43 | polysics | ok, so, what is a quick-ish thing i could try to see if i can get OOB DTMF, please? |
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| 10:57.44 | field_it | re |
| 10:58.03 | field_it | how ca i tell the voicemail to only look for gsm encoded sound files? |
| 10:58.26 | field_it | actually it tries to play alaw but sound files are in gsm only |
| 10:59.31 | field_it | ah, ok |
| 10:59.50 | field_it | just set the disallow=all and allow=gsm in sip.conf. |
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| 11:11.43 | kaldemar | field_it: you just limited the phone to use gsm codec only, which is lower quality than alaw. is that what you really wanted? |
| 11:13.25 | field_it | kaldemar: hi, thx for the hint. well, yes, i'd try to stick to that and see what's worth it. i'm running * on a rt-n16. have no idea so far how performant this box is regarding codecs.. |
| 11:14.13 | field_it | kaldemar: do you have recommendations about codecs on embedded systems? the rt-n16 isn't bad in it's class i guess. but i'm lacking experience.. |
| 11:17.05 | kaldemar | field_it: only one recommendation, don't do transcoding on the box. have audio files in a format that channels use. |
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| 11:31.03 | mandla | irroot, |
| 11:33.06 | field_it | kaldemar: yes, thx. |
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| 12:03.17 | the_5th_wheel | Hi All. Has anyone had issues with soxmix not mixing entire asterisk mixed tracks together? Im having a strange issue where the mixed recording stops at about 20% into the call. Are there perhaps any alternatives to soxmix? |
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| 12:19.34 | tzanger | hm, has anyone noticed asterisk 1.8.7.1 (maybe earlier too, haven't tried yet) ignoring --without-gtk2 and picking up GTK2 from elsewhere? |
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| 14:14.36 | Katty | morning |
| 14:15.25 | [TK]D-Fender | Katty, Mew |
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| 14:23.04 | Katty | hugs [TK]D-Fender |
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| 15:08.40 | tzanger | morning... does anyone else have troubles with 1.8.7.1 and maybe earlier 1.8 refusing to NOT build the gtk crap? |
| 15:08.49 | tzanger | it sees GTK and ignores my --without-gtk2 |
| 15:08.54 | tzanger | (this is on OE if it matters) |
| 15:09.23 | pabelanger | tzafrir: pb the output |
| 15:10.07 | tzanger | pabelanger: not sure how much it'd help, it's just hte configure script getting it rong |
| 15:10.25 | tzanger | config.log shows --without-gtk2 but then GTK2_INCLUDE and GTK2_LIBS are set |
| 15:10.45 | tzafrir | tzanger, there's gmime which is loosly related to the gtk stack |
| 15:11.04 | tzanger | hm, I can try blowing that away as well |
| 15:12.12 | tzafrir | Is the gtk2 code still in place in 1.8? |
| 15:12.17 | tzafrir | Wasn't it removed? |
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| 15:15.01 | tzafrir | There's a test for GTK2. I guess that this is only for menuselect? |
| 15:15.50 | tzafrir | No module in the tree requires gtk2 |
| 15:17.23 | tzanger | yes, it's menuselect that's failing |
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| 15:35.17 | spotter | can anyone help me understand why an extension of "s", isn't working as a default from in my "from" context (from voip provider) |
| 15:35.49 | spotter | if I put in the username/# it works, but I figured "s" should work too (and shouldn't matter what the # is if I have multiple) |
| 15:35.50 | WIMPy | Because it isn't a default. |
| 15:36.03 | spotter | is there a way to say "default"? |
| 15:36.10 | Gugge | i |
| 15:36.16 | Gugge | thats the invalid extension :) |
| 15:36.20 | spotter | yes |
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| 15:36.31 | WIMPy | "_." but that will probably have bad side effects. |
| 15:36.34 | Gugge | why not use a |
| 15:36.41 | Gugge | _X. extension |
| 15:36.55 | WIMPy | i might work. |
| 15:36.58 | *** part/#asterisk sekil (~sekil@78.24.104.73) |
| 15:36.58 | spotter | hmm |
| 15:37.02 | spotter | maybe |
| 15:37.10 | spotter | I guess callcentric's instructions are wrong |
| 15:37.11 | spotter | http://www.callcentric.com/support/device/asterisk/1_6 |
| 15:37.23 | tzanger | _. does funny things, it's always neat to see the call hang up and then the console complain that it's trying to call h :-) |
| 15:37.24 | spotter | or did it work like this in 1.6, but no longer in 1.8? |
| 15:38.09 | WIMPy | No, s has never been a default. s matches nothing. |
| 15:38.25 | spotter | ok |
| 15:38.27 | spotter | so always wrong |
| 15:38.38 | Gugge | s matches s :) |
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| 15:38.51 | spotter | thanks |
| 15:39.02 | WIMPy | If you have a classig phone and lift the receiver to dial, it hits s. |
| 15:39.12 | [TK]D-Fender | WIMPy, "s" matches "s" :) |
| 15:39.16 | WIMPy | s/g/c/ |
| 15:39.40 | WIMPy | Yes and dialling s will off course also match s. |
| 15:40.06 | [TK]D-Fender | WIMPy, Funny, when I used to lift the receiver it never hit anything and Id dial and it'd hit what I dialed :) |
| 15:40.27 | spotter | here's a different more complicated question. I've done a bit of reading of queues and it seems I can get a queue variable that tells me how many people are in the queue |
| 15:40.59 | spotter | what I'm wondering, is it possible to determine if there's an "member" (agent) available for the queue right away or if the caller will have to hold |
| 15:41.26 | spotter | the idea is to ask the caller some Qs if they have to hold, otherwise go directly to agent |
| 15:42.33 | WIMPy | [TK]D-Fender: So y mananged to combine the drawbacks of the old and the new? ;-) |
| 15:43.14 | [TK]D-Fender | WIMPy, Well... when I'd lift a receiver... I've never just automatically landed on "s". I dialed. What about you? |
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| 15:56.57 | Katty | i made a decision today. |
| 15:57.00 | Katty | a very important decision. |
| 15:57.13 | [TK]D-Fender | PAPER! :) |
| 15:57.15 | Katty | if i ever get married, i want to have cases and cases of monster at the reception. |
| 15:57.24 | Qwell | monster? eww |
| 15:57.30 | Katty | yes. monster. |
| 15:57.35 | Katty | red bull would also work |
| 15:57.49 | Katty | and rather than a Wedding Cake |
| 15:57.52 | Katty | i want a wedding Cheesecake |
| 15:57.56 | Qwell | redbull cake! |
| 15:58.00 | Katty | and rather than sappy love music |
| 15:58.07 | Katty | i think i want industrial music, or maybe dubstep |
| 15:58.10 | Qwell | that's several decisions |
| 15:58.15 | Katty | yes. yes it is. |
| 15:58.43 | tzanger | dubstep? good god my dear, you need an intervention! |
| 15:59.42 | Katty | i love you too tzanger |
| 16:00.20 | [TK]D-Fender | Katty, Why don't you just GO to rave? |
| 16:00.48 | Katty | because the rave should come to me |
| 16:01.03 | Katty | i would be cool with rave at my wedding. |
| 16:01.04 | *** part/#asterisk AmirBehzad (~behzad@31.184.187.1) |
| 16:01.40 | WIMPy | [TK]D-Fender: Yes, I do. |
| 16:02.01 | tzanger | Katty: :-) |
| 16:02.08 | [TK]D-Fender | WIMPy, Ok, so for shared perspective, what are your circumstances? |
| 16:02.38 | WIMPy | With dahdi, immediate=yes. For other channels it's default. |
| 16:03.31 | WIMPy | Or fixed. |
| 16:03.33 | [TK]D-Fender | WIMPy, Ok, and what percent of the analog FXS DAHDI user-base do you believe your implementation represents? |
| 16:03.53 | [TK]D-Fender | User w/ bat-phones... vs ... not |
| 16:04.05 | WIMPy | NFI. Analog was before my times. |
| 16:04.35 | [TK]D-Fender | Good to know :) |
| 16:05.05 | WIMPy | An yes, I know it's easier the ugly way, but if you try to do a PBX that just sets the quality too low. |
| 16:05.38 | spotter | you need to pick up the phone to be connected to an operator to bridge your call to the right # :) |
| 16:06.43 | WIMPy | Yes, turn the crank to get an operator. |
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| 16:12.46 | [TK]D-Fender | misses the days when he could send 10,000 volts down the line and fry the mofo who just pissed him off. |
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| 16:37.24 | LemensTS | If I am in a call, on a sip phone to an outside line, is there an easy way to join another sip phone so it is a 3 way call? |
| 16:37.46 | LemensTS | Asterisk 1.8 btw |
| 16:38.07 | [TK]D-Fender | LemensTS, Rest of the call details please... |
| 16:38.27 | [TK]D-Fender | LemensTS, Most SIP phones have 3-way calling. |
| 16:39.18 | LemensTS | They are Polycom 321's. I will look it up for the phones, I figured I would have to do soemething server side. |
| 16:39.27 | WIMPy | [TK]D-Fender: Charles Bronson style? |
| 16:39.41 | [TK]D-Fender | WIMPy, Yes your phones has plety of means fo doign this. |
| 16:39.45 | [TK]D-Fender | LemensTS, rather |
| 16:40.10 | [TK]D-Fender | Bronson ... wow... I don't do guns ... any more... |
| 16:41.08 | WIMPy | [TK]D-Fender: If you don't know it you should watch "Telefon" with him. |
| 16:41.26 | LemensTS | TK: And that is Polycom making the 3 way call, no 3 way calling on the POTS line needed correct |
| 16:43.00 | [TK]D-Fender | LemensTS, Well the phone has 3-way ... to the server. |
| 16:43.10 | [TK]D-Fender | now taht other call is it's own entity |
| 16:43.30 | [TK]D-Fender | LemensTS, Is there something you might want to tell us about your use of lines perhaps? |
| 16:43.32 | Katty | who's bunny is Johnny Gray Giant? |
| 16:43.52 | Qwell | Katty: Corydon |
| 16:44.07 | Katty | infobot: seen corydon |
| 16:44.15 | infobot | Katty: i haven't seen 'corydon' |
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| 16:44.26 | Qwell | he doesn't come here |
| 16:44.32 | tzanger | infobot: seen my stapler |
| 16:44.33 | infobot | tzanger: i haven't seen 'my stapler' |
| 16:44.46 | Katty | Qwell: also not registered with nickserv |
| 16:44.50 | Katty | Qwell: which channel is he usually in? |
| 16:44.55 | Qwell | -dev |
| 16:44.58 | Katty | k |
| 16:45.02 | Qwell | and -social |
| 16:46.37 | LemensTS | TK: Ok I read the other day about the 3way on the phone being a flash hook to the telco, I found a different PDF now of doing this using the CONF button. I will add that button to the poly cfg file and try. Thank you for the help. |
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| 16:48.10 | [TK]D-Fender | LemensTS, Each are valid. on your analog line you may have 3-way service. |
| 16:48.31 | [TK]D-Fender | LemensTS, And most SIP phones can juggle multiple calls (each of which is distict to the server). |
| 16:48.43 | [TK]D-Fender | LemensTS, So you should be clear which one you are loking to do |
| 16:49.58 | WIMPy | Basic telephony features |
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| 17:08.15 | ctjctj | I have two sip peers for trunking. Normally when I dial out I just use one of those peers. I'm looking for suggestions on how to best handle it when one of those trunking peers is unreachable and to bounce back and forth so that I don't use the same trunk every time. |
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| 17:10.10 | [TK]D-Fender | ctjctj, Well you have to decide when to start trying again for one you want to avoid for a while |
| 17:10.41 | [TK]D-Fender | ctjctj, Everything in there is jsut a little dialplan to flip between your peers for outbound calls. |
| 17:11.23 | ctjctj | Is there an application that will return the state of a peer? |
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| 17:12.44 | WIMPy | DEVICE_STATE if you have qualify enabled. |
| 17:14.09 | ctjctj | WIMPy: It sounds like the solution is to have a dialplan like "outbound-freenum2" that does some processing to decide which trunk to use based on the status. Then actually bother to check the return from the Dial() applicaiton. |
| 17:15.03 | WIMPy | Sounds like a good strategy. |
| 17:15.53 | [TK]D-Fender | might want both |
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| 17:16.24 | ctjctj | Just because I'm here. I've been looking for the phrase "If you know your party's extension you can dial it at any time" or words to that affect. |
| 17:16.27 | WIMPy | I interpreted "Then" as both in that order. |
| 17:16.59 | ctjctj | [TK]D-Fender: Thank you. I've got to program/write the dialplan to take into account checking for dead/unreachable trunks and flip flopping. |
| 17:17.14 | [TK]D-Fender | yup |
| 17:17.56 | [TK]D-Fender | ctjctj, There is a pre-made sound file for that . take a look in the folder. It should stand out. |
| 17:18.24 | ctjctj | It is amazing how much more sophisticated asterisk dialplans have become since I wrote my first dialplan 5+ years ago.... |
| 17:18.41 | ctjctj | [TK]D-Fender: I was looking and just didn't find it. I'll look again. |
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| 17:20.51 | ctjctj | [TK]D-Fender: Yeah, found it. |
| 17:21.04 | [TK]D-Fender | ctjctj, glad to hear |
| 17:21.21 | ctjctj | which is sad because I've looked before. |
| 17:23.53 | Qwell | ctjctj: there's a .txt file included with sounds releases, that contains the phrases in each prompt |
| 17:26.42 | ctjctj | Qwell: thanks |
| 17:26.54 | ctjctj | Allyson had a hard time saying "All your base are belong to us." |
| 17:36.03 | dailylinux | how is * today as it was criticized for not being modular? |
| 17:36.26 | dailylinux | Does * as a competing product which is open source? |
| 17:37.08 | WIMPy | cannot parse the 2nd question |
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| 17:37.27 | WIMPy | And for my taste it is too modular. |
| 17:37.45 | dailylinux | oh? |
| 17:38.00 | WIMPy | Too many situations where you need to know which modules you are acting with to get what you want. |
| 17:38.22 | dailylinux | i see |
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| 17:38.58 | Qwell | criticized for not being modular? wtf |
| 17:39.22 | [TK]D-Fender | dailylinux, What is your goal? And what are your criticisms? |
| 17:39.32 | WIMPy | 0.9 perhaps? :-) |
| 17:39.39 | dailylinux | i used * around 7-8 years back |
| 17:39.41 | [TK]D-Fender | dailylinux, And who are these other people you are alluding to? |
| 17:39.53 | dailylinux | no critics at all! |
| 17:40.03 | [TK]D-Fender | dailylinux, 7 years ago is pretty much pre 1.0 |
| 17:40.11 | dailylinux | just read pros and cons of * 5+ years back |
| 17:40.14 | dailylinux | yeah |
| 17:40.15 | [TK]D-Fender | <dailylinux> no critics at all! <- you said there were |
| 17:40.25 | dailylinux | i have personally none |
| 17:40.36 | [TK]D-Fender | Or is this 7+ years ago you're talking about? |
| 17:40.40 | dailylinux | but i've read those critics 5+ years back |
| 17:40.45 | dailylinux | yep |
| 17:41.02 | Qwell | They were just as wrong then as they are now. |
| 17:41.12 | [TK]D-Fender | dailylinux, Come up with your own opinions, not the vastly outdated and unvalidated ones of others |
| 17:41.28 | dailylinux | have not used * lately |
| 17:42.00 | dailylinux | but gonna install it next year for personal usage |
| 17:42.23 | [TK]D-Fender | dailylinux, Go for it |
| 17:42.31 | dailylinux | :) |
| 17:43.09 | dailylinux | i remember i used it with a free service, which i don't remember name of anymore |
| 17:43.18 | dailylinux | worldfreecall or something |
| 17:43.31 | [TK]D-Fender | Free World Dialup |
| 17:43.31 | dailylinux | wonder if that service still exists |
| 17:43.33 | [TK]D-Fender | probably |
| 17:43.41 | [TK]D-Fender | "FWD" |
| 17:43.42 | dailylinux | yep, you got it! |
| 17:44.04 | [TK]D-Fender | That was originally not a PSTN termination service, just P2P SIP |
| 17:44.07 | dailylinux | exactly! |
| 17:44.11 | dailylinux | yep |
| 17:44.17 | [TK]D-Fender | (well.. technicallly through their switch w/ reinvites, etc) |
| 17:44.24 | dailylinux | i was able to call many VOIP customers in the world |
| 17:44.35 | dailylinux | like Telio here in Norway |
| 17:44.45 | dailylinux | does FWD still exist? |
| 17:44.48 | [TK]D-Fender | There were a few free ones that started off as experiments and quickly closed up as regular paid services |
| 17:45.16 | dailylinux | hmm :( |
| 17:45.19 | dailylinux | typical |
| 17:45.22 | [TK]D-Fender | Yes FWD still exists, as do other similar free networks like Ekiga.net, and ENUM serves a certain simlar role |
| 17:46.17 | dailylinux | i see. Good to know! |
| 17:46.53 | dailylinux | my main goal with * is to be able to call free to VOIP phones around the world |
| 17:47.11 | dailylinux | and get a free inbound number |
| 17:47.34 | [TK]D-Fender | Thre are a few limited free inbound options out there. |
| 17:47.47 | dailylinux | mm |
| 17:47.55 | [TK]D-Fender | calling free to "voi phones" ... pretty much a given. Just a question ofhow you get to know about and reach them |
| 17:48.03 | dailylinux | i'll come back for more talk about that and more |
| 17:48.13 | [TK]D-Fender | If they are friends then you can arrange to common means of reaching. |
| 17:48.42 | dailylinux | even if they are connected on other VOIP provider? |
| 17:50.48 | [TK]D-Fender | who is "they"? |
| 17:50.56 | [TK]D-Fender | NVM |
| 17:50.58 | [TK]D-Fender | misread.. |
| 17:51.11 | [TK]D-Fender | wel.. depends what they are connected to |
| 17:51.21 | [TK]D-Fender | Some providers let you call their clients directly. |
| 17:51.36 | [TK]D-Fender | Maybe your clients will connect to multiple resources |
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| 17:57.28 | carp1 | <PROTECTED> |
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| 18:39.46 | carp1 | . |
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| 20:08.14 | BlackBishop | using realtime auth, I have the table "secret" with the password stored in clear. works ok. |
| 20:08.53 | BlackBishop | changed the name of the table to "md5secret", and encrypted "a" in md5 ( like echo -n 'a' | md5sum .. results in the same thing as md5('a') in mysql ) |
| 20:09.03 | BlackBishop | now I can't auth, anything I'm doing wrong ? |
| 20:09.29 | BlackBishop | 10.0.0-rc2 http://d3xt3r01.tk/~dexter/ast_auth_problem.txt |
| 20:11.29 | plundra | BlackBishop: Arent you supposed to use the peername:realmname:secret as input for your hash? (or whatever order it is) |
| 20:11.36 | BlackBishop | OW FSCK |
| 20:11.37 | BlackBishop | yeah |
| 20:11.44 | BlackBishop | totally forgot about that part |
| 20:12.48 | BlackBishop | peer:realm:password, right ? |
| 20:13.38 | plundra | Not 100% sure, havent done it in a while and I'm on my phone, so can't check for you :-P |
| 20:13.53 | BlackBishop | that's the one. |
| 20:14.04 | BlackBishop | totally missed this part |
| 20:14.42 | plundra | I hope you didn't drop the old data :-) Or forgot to keep a backup. |
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| 20:17.07 | BlackBishop | neah, I'm ok :) |
| 20:17.47 | BlackBishop | solved ! :) |
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| 20:54.23 | beek | At Astricon I heard someone say that Asterisk has "n" percentage of the PBX market right now. IIRC, it was 20% but I may be "misremembering." Can anyone tell me what that number is? |
| 20:57.21 | afink | I would guess a smaller number than that |
| 21:01.09 | timbreck | hi, I'm getting the error "Unable to open pseudo device" when using MeetMe(), I've read that it is related to /dev/dadhi which I don't seem to have on my system. What do I have to install to have the pseudo device ? |
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| 21:03.23 | r0m|u | timbreck, you need to recompile asterisk with dahdi. just make menuconfig and chose your new modules. once your done run "make" and copy your new modules in place including your new binary's and or lib's |
| 21:03.46 | r0m|u | timbreck, make sure you do not run make install |
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| 21:06.03 | beek | afink: thanks. Me too. |
| 21:06.46 | bacon4leif | I saw a study a couple years ago showing Asterisk having a fairly large slice of the pie |
| 21:06.57 | r0m|u | timbreck, are you sure you are missing dahdi? |
| 21:07.19 | drudge` | when i do sip show channels, i see a 192 address being nat'd to an outside address i dont recognize |
| 21:07.24 | timbreck | r0m|u: I'm not sure, I installed asterisk using the package in debian stable |
| 21:07.34 | r0m|u | I am sure is there |
| 21:07.37 | drudge` | usually looks like this "192.168.1.79 (None) 8af7b290-ff 00101/00001 unkn No Rx: REGISTER" |
| 21:08.03 | timbreck | r0m|u: how can I check ? |
| 21:08.50 | timbreck | r0m|u: I appreciate the help, btw |
| 21:09.05 | r0m|u | timbreck, maybe dahdi show status? |
| 21:09.41 | r0m|u | timbreck, to see if the pseudo channel is up do dahdi show channels |
| 21:09.52 | timbreck | r0m|u; No DAHDI found. Unable to open /dev/dahdi/ctl |
| 21:10.40 | r0m|u | well thats odd that it was in the pkg.... try searching for the dahdi pkg in apt-get? |
| 21:10.46 | r0m|u | not* |
| 21:10.49 | drudge` | when i do sip show channel and enter in teh channel ID, i dont recognize the IP and it doesnt belong to anyone registered with 'sip show peers' |
| 21:11.15 | drudge` | is that someone attmepting to use it or could it be some phone somewhere attemptign to register |
| 21:11.25 | bacon4leif | beek: in a presentation from about a year ago I linked to a pie chart that showed open source PBX's had 18% market share, but the link no longer works (they removed the article or something). IT was on nojitter.com |
| 21:11.37 | drudge` | the channel id also changes so quickly that i dont have time to copy/paste it for a 'sip show channel' before it changes |
| 21:12.03 | timbreck | do you guys know if I really need MeetMe (and dahdi) to use ICES() ? I "just" want Asterisk to pick up the call and send it to ice cast. It kind of works without meet me, it's just that it's streaming an empty stream |
| 21:12.03 | beek | Thanks bacon4leif. So 20% isn't too far off base. |
| 21:12.15 | bacon4leif | beek: right, and 18% figure was over 2 years ago |
| 21:12.26 | beek | I need to find some authorative documentation for that. |
| 21:12.32 | [TK]D-Fender | beek, that was OSS telephony, not just * |
| 21:12.44 | beek | Okay [TK]D-Fender. |
| 21:12.46 | [TK]D-Fender | beek, And of course, 1 place's claim |
| 21:13.06 | timbreck | r0m|u: i installed the package already, but I suspect this distribution is tweaked in some way (provided by my vps host) |
| 21:13.07 | beek | I'm going to have to go on a Google hunt to find this. |
| 21:14.07 | timbreck | r0m|u: I read dadhi must be installed as a kernel extension too, can I do that now or should it have been done by my provider when installing the server |
| 21:14.31 | timbreck | r0m|u: kernel driver (?) |
| 21:15.51 | [TK]D-Fender | timbreck, Your provider is giving you the server? |
| 21:16.21 | *** part/#asterisk umay (~chris@67-6-172-210.hlrn.qwest.net) |
| 21:16.33 | bacon4leif | beek: http://www.zdnet.com/blog/open-source/does-open-source-control-18-of-the-pbx-market/3391 <-- although links pointing to data are dead |
| 21:16.44 | timbreck | [TK]D-Fender: I chose a distribution when ordering it, and they installed it |
| 21:17.01 | [TK]D-Fender | timbreck, An what did you pick? |
| 21:17.34 | timbreck | [TK]D-Fender: Debian squeeze |
| 21:17.47 | [TK]D-Fender | timbreck, And how was * installed? |
| 21:18.23 | timbreck | [TK]D-Fender: apt-get … is that ok ? |
| 21:18.31 | timbreck | should I compile my own? |
| 21:18.38 | [TK]D-Fender | timbreck, the use apt-get to install DAHDI |
| 21:18.39 | beek | thanks bacon4leif |
| 21:18.47 | bacon4leif | np |
| 21:18.50 | pabelanger | timbreck: apt-get install asterisk-dahdi |
| 21:19.54 | timbreck | [TK]D-Fender: I did install this: dahdi - utilities for using the DAHDI kernel modules |
| 21:20.19 | timbreck | pabelanger: I tried that after searching online, and the package is unknown |
| 21:20.37 | *** join/#asterisk kylix (~trek@ppp-157-69.27-151.libero.it) |
| 21:21.08 | kylix | Hi! Anybody using Jabber and Asterisk? |
| 21:22.23 | [TK]D-Fender | timbreck, Not just utilities |
| 21:22.27 | [TK]D-Fender | DAHDI-LINUX |
| 21:22.30 | kylix | Is it possible to make Asterisk execute a certain AGI script when receiving an XMPP proper formatted? |
| 21:23.06 | timbreck | [TK]D-Fender: it's installed too .. I pretty much installed anything that mentioned dahdi |
| 21:23.21 | [TK]D-Fender | stop * |
| 21:23.23 | kylix | and after executing the script send back the Jabber user a text message |
| 21:23.30 | [TK]D-Fender | run dahdi_cfg -vvvv |
| 21:23.45 | [TK]D-Fender | then start * manually. Test that chan_dahdi is actually loaded |
| 21:24.06 | timbreck | [TK]D-Fender: line 0: Unable to open master device '/dev/dahdi/ctl' |
| 21:24.19 | [TK]D-Fender | timbreck, You stopped * first? |
| 21:24.47 | timbreck | [TK]D-Fender: I did now, same thing |
| 21:25.21 | timbreck | Let's say I want to start over from scratch. What distro/packages should I use |
| 21:26.12 | *** join/#asterisk timeshell (~chatzilla@gw.lusi.on.ca) |
| 21:26.49 | [TK]D-Fender | checkout time here, BBIAb |
| 21:26.58 | *** join/#asterisk umay (~chris@67-6-172-210.hlrn.qwest.net) |
| 21:27.12 | *** join/#asterisk Carlos_PHX1 (~Carlos@ip24-56-6-80.ph.ph.cox.net) |
| 21:28.02 | drudge` | welcome carlos |
| 21:28.57 | timbreck | Again, I don't see anything in the doc telling me I do need to use MeetMe to be able to stream to icecast |
| 21:30.20 | kylix | no Jabber user for sending messages through Asterisk? |
| 21:31.09 | kylix | or somebody that uses XMPP messaging with Asterisk |
| 21:32.16 | drudge` | man, i added an access-list deny statement and after 5 seconds of applying it, 1 IP has over 688 matches |
| 21:47.07 | *** join/#asterisk sofh (~sofh@ner-as30696.alshamil.net.ae) |
| 21:47.27 | sofh | hi all |
| 21:47.43 | sofh | i want to have custom cdr reporting for Asterisk 1.8... |
| 21:48.12 | sofh | problem is when i match dialed number with rates table , i've no way to filter exact prefix matching the dst number |
| 21:48.13 | *** join/#asterisk bchia (~Adium@nat/digium/x-fdxnpzydsyqucukq) |
| 21:48.30 | sofh | for eg : 441 is dialed then in query i get both 441 as uk landline and 44 as uk |
| 21:48.47 | sofh | is there a way in CDR function or something for accounting purpose ? |
| 21:57.21 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 21:59.49 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 21:59.58 | umay | sofh: a low-level SQL function to walk thru the matches is one way to do it |
| 22:00.41 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 22:03.09 | sofh | umay: i've read all SQL functions they are all giving me all matching codes against prefix |
| 22:03.47 | sofh | if 4474 is dialed SQL query should match it with 4474 but it mathces against both 4474 and 44 and infact 447 too |
| 22:04.15 | umay | the SQL query you wrote ? |
| 22:05.43 | sofh | <PROTECTED> |
| 22:05.43 | sofh | <PROTECTED> |
| 22:05.43 | sofh | <PROTECTED> |
| 22:05.43 | sofh | <PROTECTED> |
| 22:05.57 | sofh | sorry its wrong one |
| 22:06.19 | sofh | is if i take lenght(prefix or dst) it gives only 1 record |
| 22:06.26 | sofh | while i need to apply code names on all cdrs i've |
| 22:06.28 | sofh | no matter 1000 |
| 22:06.41 | sofh | but lenght function in SQL wants me to limit it to 1 |
| 22:07.51 | sofh | can CEL help in any way ? |
| 22:09.58 | umay | probably no on the cel |
| 22:10.27 | umay | i suggest a SQL function - not a query |
| 22:10.42 | umay | you can do more in functions, eg. loops |
| 22:11.10 | umay | that query you pasted will run s l o w |
| 22:11.30 | umay | take a look at PostgreSQL Trigger Functions for a bit |
| 22:12.02 | sofh | am using mysql 5.1 that also supports trigger |
| 22:12.36 | sofh | so u mean i should parse on SQL level to match code with dialed number ? |
| 22:15.40 | *** join/#asterisk TimeRider (steve@188-220-33-123.zone11.bethere.co.uk) |
| 22:15.47 | umay | database functions in PostgreSQL can be written in SQL and several other languages |
| 22:15.52 | umay | not sure about mysql |
| 22:16.49 | Nugget | will never understand why people choose to use mysql |
| 22:21.40 | timbreck | Can anyone tell me why this http://pastie.org/2900318 streams a blank stream ? Echo() at the same place works |
| 22:23.13 | *** join/#asterisk Russ (~russ@206.29.182.181) |
| 22:26.21 | *** join/#asterisk caveat- (hoax@gateway/shell/bshellz.net/x-ojgqnnnbjkjclmgd) |
| 22:42.18 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 23:02.14 | *** join/#asterisk citywok (~citywok@67-134-194-33.dia.static.qwest.net) |
| 23:26.55 | *** join/#asterisk IsUp (5b8e8e8f@gateway/web/freenode/ip.91.142.142.143) |
| 23:27.06 | IsUp | hello |
| 23:27.24 | IsUp | does bad timing causes strange sounds? |
| 23:27.42 | bacon4leif | no, packet loss and jitter do |
| 23:27.50 | bacon4leif | unless you're in a conference |
| 23:28.13 | bacon4leif | but you shouldn't be able to get into a confernece with a timing module, because app_meetme wouldn't build |
| 23:28.19 | carrar | can cause pregnancy too |
| 23:28.24 | IsUp | i am using DAHDI |
| 23:28.45 | IsUp | my PRI spans was working well, then my telco link gone for 10 minutes |
| 23:29.01 | IsUp | after they restart their side, i have strange sounds, robotic voices |
| 23:29.42 | IsUp | i have GSM gateways connected with PRI, i am using telco clock as ref clock |
| 23:30.10 | IsUp | so span 0 is telco and i am using span 0 for timing on span 1,2,3 |
| 23:31.25 | IsUp | i suspect that something wrong with telcos timing |
| 23:32.24 | *** join/#asterisk LemensTS (~matthew@adsl-70-238-163-254.dsl.stlsmo.sbcglobal.net) |
| 23:32.24 | *** join/#asterisk tuxxie (~Ryan@rrcs-70-63-90-226.midsouth.biz.rr.com) |
| 23:34.01 | IsUp | how can i check if timing is correct or now? |
| 23:34.05 | IsUp | *not |
| 23:37.21 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 23:45.27 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-yuyuynnmamheuccz) |
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| 23:47.48 | jmwpc | is there a dialplan application to check if a google talk line is in use? I tried ChanIsAvail but that does not appear to work for this purpose. |
| 23:49.51 | [TK]D-Fender | core show function DEVICE_STATE |