| 00:02.08 | TripleFFFF | hmm |
| 00:02.09 | TripleFFFF | SELECT * FROM routes WHERE '234534234' RLIKE pattern ORDER BY LENGTH(pattern) DESC; |
| 00:02.15 | TripleFFFF | means the patern by default sucks |
| 00:02.43 | Qwell | RLIKE? |
| 00:03.38 | TripleFFFF | oh |
| 00:03.40 | TripleFFFF | fuckers |
| 00:03.57 | Qwell | Is that valid? Never heard of it |
| 00:04.02 | TripleFFFF | yeah |
| 00:04.04 | TripleFFFF | BNUT |
| 00:04.10 | TripleFFFF | astcc is fucked up in svn |
| 00:04.23 | TripleFFFF | default patern on install should be ^1.* |
| 00:04.25 | TripleFFFF | not 1* |
| 00:04.32 | TripleFFFF | 1* matches anything even BOB |
| 00:04.37 | Qwell | ahh, regex? |
| 00:04.56 | TripleFFFF | yeah |
| 00:05.03 | Qwell | cool |
| 00:05.13 | TripleFFFF | sucks.. i made ^1.* and now all works even invalid numbers |
| 00:05.22 | TripleFFFF | the sucker wasd ailing out on anything even 13423 |
| 00:05.30 | Qwell | heh |
| 00:05.38 | TripleFFFF | not good for installs.. hacked and pakistanese dis will be dialed |
| 00:05.39 | TripleFFFF | lol |
| 00:24.00 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 00:24.01 | *** topic/#asterisk is Asterisk: The Open Source PBX -=- http://www.asterisk.org -=- Asterisk 1.2.10 and Zaptel 1.2.7 released! (July 14, 2006) -=- FreePBX/AMP/Asterisk@Home/Trixbox support in #freepbx |
| 00:28.48 | Spla4t1 | is there a problem with the 729 codec that causes dead air. |
| 00:28.54 | Spla4t1 | other codecs work fine |
| 00:31.00 | *** join/#asterisk CANO-1982 (i=alejandr@190.48.66.106) |
| 00:35.43 | Spla4t1 | also as far as the licensing goes for g729 if I have 1 trunk and 1 phone does that require licenses for 2 channels? |
| 00:36.07 | file | you pay per simultaneous channels, so if you want 2 calls up at a time - you need 2 licenses |
| 00:36.45 | Spla4t1 | Ok so I should be fine with 1 phone using 729 calling a phone provider with 1 channel using 729 |
| 00:38.56 | Qwell | sorry, but...this is a pretty stupid idea |
| 00:38.56 | Qwell | http://www.geeks.com/details.asp?InvtId=USBFD-WB-512-BK&cpc=SUGG |
| 00:39.06 | ariel_ | Spla4t1, using the pass through options sometimes works. make sure you have setup canreinvite=yes |
| 00:39.48 | ariel_ | Qwell, well yes and no. I will get allot of geeks would get it. |
| 00:40.01 | Qwell | yeah, but like...that's too bulky |
| 00:40.04 | *** join/#asterisk Mw3 (i=mw3@national.t-error.hu) |
| 00:40.26 | ariel_ | I would not use it. |
| 00:40.32 | ariel_ | but I guess it has it's use |
| 00:40.50 | *** join/#asterisk topping (n=topping@207.47.6.136.static.nextweb.net) |
| 00:41.00 | ariel_ | have you seen the verizon gps route... the display gets you there but sucks. |
| 00:41.03 | *** join/#asterisk tempest1 (n=asf@adsl-144-60-200.chs.bellsouth.net) |
| 00:43.14 | Spla4t1 | I get a out of license error with only the trunk configured for g729 |
| 00:44.19 | *** join/#asterisk tempest1 (n=asf@adsl-0-238-115.chs.bellsouth.net) |
| 00:45.08 | ariel_ | Spla4t1, if you have not installed either the testing one from intel or the paid one from digium your not going to get it working if that what it said. |
| 00:45.55 | Spla4t1 | I just went through the registration process. |
| 00:48.07 | Spla4t1 | I see it licensed 1 channel when I do a show g729 |
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| 00:51.23 | ariel_ | Spla4t1, ok one but you really need 2 to get started. |
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| 00:53.17 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 00:53.23 | Spla4t1 | ariel: It looks like its not freeing up the decoder |
| 00:53.41 | Spla4t1 | on a show g729 it shows 0/1 encoder/decoders in use. |
| 00:55.57 | ariel_ | Spla4t1, normal |
| 00:56.19 | ariel_ | when you setup asterisk with g729 you really need 2 lic |
| 00:56.22 | ariel_ | not one. |
| 00:56.32 | Spla4t1 | huh.. I have 1 license and 1 channel configured for g729. I get the error saying that the it is out of licenses for the decoder.. This is normal. |
| 00:57.13 | ariel_ | yes |
| 00:57.34 | ariel_ | when you do show channels don't you see 2 up for a call |
| 00:58.05 | Spla4t1 | I do but the connection from my phone to asterisk is ulaw not g729 |
| 00:58.42 | ariel_ | asterisk needs to do the transcoding |
| 00:59.07 | ariel_ | if you setup the phone to do g729 and setup canreinvite=yes it should work |
| 00:59.18 | Spla4t1 | lemme try that real quick. |
| 01:02.27 | Spla4t1 | Ok that did it.. |
| 01:02.42 | Spla4t1 | although the sound quality was worse than ulaw or gsm I think. |
| 01:03.04 | Spla4t1 | that might be my phone though.. Im not sure it has the horsepower to do wpa and g729 at the same time. |
| 01:03.08 | [TK]D-Fender | you don't need reinvites for G729 to passthrough, and can cause routing issues... |
| 01:04.52 | Spla4t1 | now to see if I can get it to work over gprs/edge.. |
| 01:08.28 | Spla4t1 | I removed the reinvites and it still worked. |
| 01:11.26 | *** part/#asterisk TripleFFFF (n=TripleFF@147-102.mc.cite.net) |
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| 01:13.56 | *** mode/#asterisk [+o russellb] by ChanServ |
| 01:14.15 | Spla4t1 | although its not freeing up the encoder/decoder after the call is ended. |
| 01:16.07 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
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| 01:22.42 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 01:23.42 | Spla4t1 | what codec does vonage use by default? |
| 01:25.40 | *** join/#asterisk techie (n=gus@voipops.net) |
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| 01:33.56 | Bobcat_1966 | Hello all, does anybody know if there is a trick to install the trunk version of asterisk-addons? I keep getting an make: *** [all] Error 2 |
| 01:34.09 | anonymouz666 | well guys |
| 01:34.15 | anonymouz666 | good luck for you |
| 01:34.22 | anonymouz666 | it's time to drink my black label |
| 01:37.29 | *** join/#asterisk Synyn_ (n=Synyn@cpe-72-181-72-81.houston.res.rr.com) |
| 01:37.36 | Synyn_ | hola |
| 01:37.49 | anonymouz666 | hola |
| 01:38.51 | russellb | Bobcat_1966: what is the error before that |
| 01:39.06 | Bobcat_1966 | let me check |
| 01:39.08 | russellb | pastebin it |
| 01:39.15 | Bobcat_1966 | sure will give me a sec |
| 01:41.05 | Bobcat_1966 | russellb: Here it is http://pastebin.ca/96645 |
| 01:41.10 | Bobcat_1966 | I appricate the help |
| 01:41.23 | russellb | no problem |
| 01:41.37 | russellb | if it's broken, it's likely my fault :) |
| 01:41.50 | Bobcat_1966 | :) |
| 01:42.10 | russellb | alright, easy fix, actually |
| 01:42.19 | russellb | you need to do a "make install" from asterisk before trying to build asterisk-addons |
| 01:42.30 | russellb | actually, no |
| 01:42.37 | russellb | you probably did do that |
| 01:42.48 | russellb | it's broken, and not to much surprise, it *was* my fault |
| 01:42.49 | Bobcat_1966 | yep I did a make clean; make install |
| 01:43.27 | russellb | i'll let you know after i commit the fix |
| 01:43.43 | Bobcat_1966 | thanks Ill be here |
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| 01:48.13 | russellb | Bobcat_1966: ok, it should be fixed in revision 263 |
| 01:48.23 | Bobcat_1966 | cool let me try |
| 01:48.26 | russellb | res_config_mysql and chan_ooh323 were broken due to recent API changes |
| 01:48.40 | russellb | only the chan_ooh323 breakage was actually my fault :) |
| 01:48.51 | russellb | it may take up to 15 minutes or so for it to make it to the mirrors |
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| 01:49.13 | Bobcat_1966 | yep its still hsowing 262... Ill be paitent |
| 01:49.42 | Bobcat_1966 | Thanks again russellb |
| 01:50.04 | russellb | you're welcome |
| 01:50.17 | russellb | thanks for pointing out it was broken :) |
| 01:51.00 | *** part/#asterisk techie (n=gus@voipops.net) |
| 01:51.05 | Bobcat_1966 | russellb: can I ask you a quick question, I noticed that after installing the latest trunk that my calls work fine but when I hang up the call, flash operating panel still shows the trunk active...any ideas on that one |
| 01:51.36 | Bobcat_1966 | has something changed that requires a change to FOP? |
| 01:51.44 | russellb | possibly, i'm not sure |
| 01:51.54 | russellb | FOP probably just hasn't been updated to the trunk |
| 01:52.09 | Bobcat_1966 | thats what I figured,,,thanks |
| 01:52.29 | Spla4t1 | is it possible to connect a client from behind nat if it does not support stun or ice? |
| 01:55.32 | Spla4t1 | would like to connect from hotspots. |
| 01:58.14 | Bobcat_1966 | russellb: Ok that did it and I can now run make install on the asterisk addons, but im noticing that a module I was loading in my old 1.2 stable is no longer working. I have to comint it out of my module.conf for asterisk to start. The module is called format_au.so any ideas on this one. |
| 02:00.00 | Bobcat_1966 | hmmm I dont even see it in the modules directory. I wonder if its obsolete and what it was used for |
| 02:03.59 | *** join/#asterisk russellb_ (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 02:03.59 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 02:08.35 | Bobcat_1966 | Hello All, Just upgraded from 1.2 stable to the newes asterisk trunk and now my ENUM is not working. Is there a trick to getting it backup and running? |
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| 02:36.20 | bkw_ | ~seen strom_c |
| 02:36.25 | jbot | strom_c <n=strom@gateway.digium.com> was last seen on IRC in channel #asterisk, 2d 6h 54m 51s ago, saying: 'very welcome'. |
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| 02:40.07 | n9urk | ~test |
| 02:40.09 | jbot | Test Failed! |
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| 03:31.18 | Asterisk_Newbie | hi all, from portugal |
| 03:31.50 | tempest1 | welcome |
| 03:32.07 | Asterisk_Newbie | I'm trying to learn how to use AGI |
| 03:32.14 | Asterisk_Newbie | with java |
| 03:32.28 | Asterisk_Newbie | Do I need to install any modules? |
| 03:34.13 | *** join/#asterisk trelane (i=trelane@unaffiliated/trelane) |
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| 03:57.52 | harryvv | myspace.com poweroutage in data center. woops :) |
| 03:59.26 | *** join/#asterisk angler_ (n=angler@12-219-146-128.client.mchsi.com) |
| 03:59.37 | harryvv | Seems it would be best to have two seperate data centers or some kind of redundent power system. I worked for one of the largest electrical measuring companies in the united states and the data center with its one hundred plus servers went off line. |
| 04:00.29 | harryvv | seems the fire sprinker techs that normally test the system do it on sundays..that was a friday and the entire electrical system collapsed. |
| 04:06.28 | nestar | 99.9 is easy |
| 04:06.33 | nestar | the .1 is hard/expensive |
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| 04:38.01 | asterisk_Newbie | Hi all from Portugal |
| 04:38.18 | asterisk_Newbie | Anyone knows anything about AGI? |
| 04:46.38 | tzafrir_laptop | maybe. ask your question and find out |
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| 05:09.41 | *** join/#asterisk microw (n=mike@218.18.38.249) |
| 05:10.39 | microw | hello, asterisk can't show the callid |
| 05:11.06 | microw | the callid from zap pstn line can't be showed, anyone knows what happenned? |
| 05:11.26 | microw | my x-lite said "unknown" |
| 05:11.46 | microw | anyone could give me a clue? |
| 05:12.24 | microw | ??anyone not sleep? |
| 05:14.31 | microw | hello? |
| 05:17.34 | tzafrir_laptop | microw, where do you expct the caller ID to show up? |
| 05:17.46 | tzafrir_laptop | a call from pstn? |
| 05:18.24 | tzafrir_laptop | if so: do you have 'callerid=asrecieved' in zapata.conf for that channel? |
| 05:18.37 | microw | yes, callerid is set right |
| 05:19.02 | microw | i want it to show on my x-lite screen |
| 05:19.27 | microw | x-lite is a extension softphone |
| 05:19.47 | microw | i am wondering what cidsignalling does |
| 05:20.08 | microw | what should i set for cidsignalling? |
| 05:20.43 | microw | tzafrir_laptop: you there? |
| 05:20.54 | tzafrir_laptop | sort of |
| 05:21.03 | microw | what should i set for cidsignalling? |
| 05:21.44 | tzafrir_laptop | how do you know it is set right? does it show correctly in any other phone? or in the CDR data? |
| 05:22.30 | microw | it doesn't show on any other phone, what is CDR data? |
| 05:22.49 | microw | i only have a x100p |
| 05:23.15 | tzafrir_laptop | if not: set debug to 10 or so, and look at the logs... |
| 05:23.59 | tzafrir_laptop | make sure debug is going to some log. look at "full" in logger.conf |
| 05:24.33 | microw | here is trixbox, how to set debug :(? |
| 05:25.13 | tzafrir_laptop | set debug 10 |
| 05:25.17 | tzafrir_laptop | from the cli |
| 05:26.18 | tzafrir_laptop | BTW: I would really recommend to connect with ssh and asterisk -r |
| 05:26.34 | tzafrir_laptop | you'd probably get mor debugging clues, generally |
| 05:27.52 | microw | amportal stop; and asterisk -r; right?? |
| 05:38.01 | hads|home | ~trixbox |
| 05:38.07 | jbot | [trixbox] NOT supported here! People using it should join #trixbox or #freepbx (FreePBX is the new name of AMP) |
| 05:43.32 | tzafrir_laptop | no no no! |
| 05:43.35 | L|NUX | hads|home |
| 05:43.38 | L|NUX | hads|home : hey |
| 05:43.57 | hads|home | hello |
| 05:44.01 | L|NUX | hads|home : i still have no success :( |
| 05:44.15 | tzafrir_laptop | microw, don't stop asterisk |
| 05:44.23 | L|NUX | hads|home : can you help me |
| 05:44.24 | L|NUX | ? |
| 05:44.46 | tzafrir_laptop | asterisk -r open a remote session to the running asterisk server |
| 05:44.57 | hads|home | L|NUX: Maybe, what was your problem again? |
| 05:45.15 | tzafrir_laptop | that said, #freepbx may be a better place |
| 05:46.02 | L|NUX | see i have DID which is being used by * all i want when some one call to that DID then it will ask for number/extension and then it will dial out using my provider to USA/Canda |
| 05:47.53 | hads|home | OK I remember now. You were pretty close from memory. What can't you get to work? |
| 05:48.46 | L|NUX | well |
| 05:48.51 | L|NUX | when i press numbers |
| 05:48.54 | L|NUX | it will not work |
| 05:49.40 | L|NUX | http://pastebin.ca/95652 |
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| 05:52.36 | L|NUX | any idea |
| 05:52.41 | trumee | guys, sometimes asterisk stops making outgoing calls for me, and i get Avoided initial deadlock for '0x81b11b8', 10 retries |
| 05:52.52 | trumee | is this some sort of bug? |
| 05:56.18 | hads|home | L|NUX: Don't PM. |
| 05:56.48 | L|NUX | ok |
| 05:57.01 | hads|home | L|NUX: So it seems that it's working and you dialled 7 which didn't match anything, so it went to the i extension and hungup. |
| 05:57.17 | L|NUX | well i dialed 4193017228 |
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| 05:57.29 | *** part/#asterisk Kerry_G (n=Kerry_G@ip70-187-129-227.oc.oc.cox.net) |
| 05:58.02 | hads|home | L|NUX: OK, well that won't match _1NXXNXXXXXX either. |
| 05:58.20 | L|NUX | i also dialed 14193017228 |
| 05:58.21 | L|NUX | :( |
| 05:58.26 | L|NUX | but give same |
| 05:58.52 | trumee | is "Avoided initial deadlock for '0x81b11b8', 10 retries" a bug? |
| 05:59.02 | hads|home | So try hard coding an extension such as 800 and testing with that first. |
| 05:59.04 | Qwell | trumee: Is it a warning? |
| 05:59.10 | Qwell | (tip; it is) |
| 05:59.15 | L|NUX | ok |
| 05:59.23 | trumee | Qwell:yes |
| 05:59.29 | Qwell | Then you can ignore it |
| 05:59.58 | trumee | Qwell:but asterisk stops making outgoing calls. and i get a 503 Service Unavailable |
| 06:00.19 | trumee | Qwell:on a production system, it will be ugly |
| 06:00.34 | trumee | Qwell:to have such a downtime |
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| 06:13.01 | L|NUX | hads|home : i have tested it on local |
| 06:13.32 | L|NUX | hads|home : and it shows that when i press keys it will find last digit |
| 06:13.37 | L|NUX | and dial it |
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| 06:24.01 | EyeCue | do meetme confs need context and stuff? |
| 06:28.54 | *** join/#asterisk RippPPppE (n=ripppppp@202.63.174.13) |
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| 06:41.53 | L|NUX | any one good with AGI |
| 06:41.54 | L|NUX | ? |
| 06:55.29 | RippPPppE | anyone |
| 06:55.29 | RippPPppE | <PROTECTED> |
| 06:55.40 | RippPPppE | sorry |
| 06:56.06 | RippPPppE | anyone know why this error come in ABE |
| 06:56.52 | RippPPppE | No application 'Set' for extension (macro-superdial, s, 1) |
| 06:57.19 | Qwell | RippPPppE: I would call Digium.. Probably not many people here know much about ABE |
| 06:57.37 | Qwell | Though, to answer your question (possibly), you might try using SetVar |
| 06:57.46 | RippPPppE | oeky thanks |
| 06:57.59 | RippPPppE | not much diff in the end result i think |
| 07:00.34 | L|NUX | can some one help me with AGI |
| 07:01.05 | L|NUX | i have write an AGI which is working fine with softphone but when i try it with hard phone it will not work |
| 07:02.26 | RippPPppE | what does the AGI script do |
| 07:02.41 | RippPPppE | soft or hard phone normally should not matter (much) |
| 07:02.49 | L|NUX | http://www.voip-info.org/wiki-Asterisk+AGI |
| 07:02.54 | L|NUX | well brother |
| 07:03.02 | L|NUX | when i dial a number it will ask me for number |
| 07:03.11 | L|NUX | from softphone when i dial number it works |
| 07:03.24 | L|NUX | but from hard phone or regular PSTN phone it will not working |
| 07:03.24 | L|NUX | :( |
| 07:05.54 | RippPPppE | can you send me the agi script |
| 07:06.02 | L|NUX | wait |
| 07:07.24 | *** join/#asterisk EyeCue (n=eyecue@unaffiliated/eyecue) |
| 07:09.24 | L|NUX | http://pastebin.ca/96843 |
| 07:11.09 | *** join/#asterisk Assid (i=assid@203.115.83.214) |
| 07:11.12 | Assid | heya |
| 07:11.50 | L|NUX | RippPPppE : http://pastebin.ca/96843 |
| 07:12.37 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 07:30.00 | *** join/#asterisk af_ (n=af@ip-164-6.sn2.eutelia.it) |
| 08:08.44 | *** join/#asterisk DarKnesS_WolF (n=wolf@62.114.187.123) |
| 08:19.50 | *** join/#asterisk af_ (n=af@ip-164-6.sn2.eutelia.it) |
| 08:29.46 | *** part/#asterisk CANO-1982 (i=alejandr@190.48.66.106) |
| 08:56.03 | *** join/#asterisk Koshatul (n=evangeli@ip157-65-132.cust.bit.net.au) |
| 09:02.23 | *** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net) |
| 09:11.27 | EyeCue | hmm , prolly silly question, but can i have my iax/sip users in a mysql backend? |
| 09:11.31 | *** join/#asterisk Koshatul (n=evangeli@ip157-65-132.cust.bit.net.au) |
| 09:13.51 | *** join/#asterisk speedwagon (n=Ariel@dsl-20-177.cofs.net) |
| 09:16.15 | hads|home | EyeCue: You can with "realtime" |
| 09:16.22 | EyeCue | realtime? |
| 09:16.32 | hads|home | BTW your quit message is pretty harsh. |
| 09:16.47 | EyeCue | it's probably warranted. |
| 09:16.51 | EyeCue | but nonetheless :) |
| 09:17.08 | EyeCue | i cant even recall what it is, unless its about pissing into the wind |
| 09:17.41 | hads|home | Yes, it is. |
| 09:17.48 | mitcheloc | well if you are good at it, you can do it into the wind ;) |
| 09:17.52 | EyeCue | heh. |
| 09:17.58 | EyeCue | now, this realtime, a mod/plugin? |
| 09:19.08 | hads|home | I don't know anything much about it, I don't have a need to run phone systems on a database. |
| 09:19.18 | hads|home | There should be info on the wiki etc. |
| 09:19.25 | EyeCue | sok, got it :) |
| 09:20.20 | *** join/#asterisk kristalino (n=kristali@84-50-84-146-dsl.trt.estpak.ee) |
| 09:31.22 | *** join/#asterisk bjohnson_ (n=bjohnson@i216-58-9-37.cybersurf.com) |
| 09:52.16 | Assid | realtime rocks |
| 09:52.24 | Assid | im planning to move my stuff to realtime as well |
| 09:55.14 | robin_sz | anyone tried the latest grandstream stuff, v 1.1.1.17? |
| 09:55.50 | robin_sz | and in particular, know how to downgrade to 1.1.0.16? |
| 09:56.22 | robin_sz | (yes, I know, the version numbering system is screwed up ...) |
| 10:06.39 | *** join/#asterisk RoyK (n=roy@122.80-203-78.nextgentel.com) |
| 10:14.24 | kay2 | has someone successfully used Sphinx with asterisk ? |
| 10:14.39 | *** join/#asterisk Splat (n=Splat@220-253-101-189.TAS.netspace.net.au) |
| 10:18.20 | *** join/#asterisk evisu (i=hIRC@bzq-88-155-208-171.red.bezeqint.net) |
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| 10:22.45 | microw | asterisk can't show the caller id from my x100p |
| 10:22.55 | trumee | guys, can somebody recommend a wireless headset with a microphone |
| 10:23.00 | microw | <PROTECTED> |
| 10:23.16 | microw | anyone knows what happenned? |
| 10:23.20 | trumee | i can find wireless headsets but no wireless microphones |
| 10:24.39 | microw | usecallerid = yes |
| 10:25.32 | microw | when incoming calls come, the caller ID is unknown, but this line do have caller id service. |
| 10:25.55 | microw | anyone know what happenned? |
| 10:27.27 | kay2 | microw: yeah |
| 10:27.35 | microw | hi kay2 |
| 10:27.55 | microw | kay2: could you help me? |
| 10:28.03 | kay2 | microw: no unfortunatelly |
| 10:28.07 | kay2 | because it's ur card |
| 10:28.09 | kay2 | that is not OK |
| 10:28.18 | microw | what? |
| 10:28.28 | kay2 | dmesg |
| 10:28.33 | kay2 | and then pastbin |
| 10:28.34 | kay2 | ~pb |
| 10:28.42 | jbot | pb is probably a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/ |
| 10:28.42 | microw | x100p can't show call id? |
| 10:29.00 | kay2 | give me the info about it, pastbin the "dmesg" |
| 10:29.03 | kay2 | and I'll tell u why |
| 10:29.12 | microw | ok, kay2 |
| 10:30.34 | *** join/#asterisk felipex (n=dsfdsf@85-18-250-142.ip.fastwebnet.it) |
| 10:31.56 | microw | http://pastebin.ca/97048 |
| 10:32.06 | microw | it is here, kay2, http://pastebin.ca/97048 |
| 10:32.45 | microw | kay2: did you get it? |
| 10:32.45 | kay2 | yeah |
| 10:32.49 | kay2 | lol it's weird |
| 10:32.54 | kay2 | where did u get ur x100p card ? |
| 10:33.34 | microw | from openvox |
| 10:33.42 | microw | it is a a100p card |
| 10:34.06 | kay2 | yeah |
| 10:34.09 | kay2 | cuz it's a clone |
| 10:34.15 | kay2 | and clones don't show the callerid |
| 10:34.22 | kay2 | unless u replace the 2 resistors |
| 10:34.50 | microw | oh?? |
| 10:35.22 | microw | which 2 resistors? |
| 10:36.10 | microw | i don't have schemetic of this card, but I work with iron all the day |
| 10:36.44 | microw | it is a very simple card, with only a motorola chip |
| 10:37.15 | microw | kay2: which 2 resistors i have to replace? |
| 10:38.24 | microw | kay2: r21 looks weird |
| 10:40.10 | microw | kay2: there? |
| 10:41.57 | kay2 | 2. The other alternative is to make the same modification to the card that Digium did when they turned them into "genuine" X100P cards. Simply carefully remove R13 and R19 with a soldering iron. R13 & R19 are pull down resistors that affects the vendorID number that is read from the card. With these two resistors gone, you now have a card that will appear as a genuine Digium X100P card to the Asterisk software. |
| 10:43.16 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 10:43.16 | *** mode/#asterisk [+o denon] by ChanServ |
| 10:43.16 | microw | what is option 1?? |
| 10:43.34 | microw | kay2: I didn't see you option 1. |
| 10:44.36 | kay2 | I past it from the web |
| 10:44.39 | kay2 | go look for it |
| 10:44.55 | microw | where is it? |
| 10:57.21 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 10:57.21 | *** mode/#asterisk [+o denon] by ChanServ |
| 10:57.55 | *** join/#asterisk clive- (n=pirch@dsl-145-1-175.telkomadsl.co.za) |
| 11:16.17 | Assid | you just remove the 2 resistors? |
| 11:27.33 | *** join/#asterisk vpanayotov (n=vdp@213.91.154.185) |
| 11:31.50 | *** join/#asterisk Vinsik (n=vinsik@dsl-145-16-216-83.maxinetti.fi) |
| 11:31.56 | Vinsik | hola! |
| 11:32.37 | *** join/#asterisk vpanayotov (n=vdp@213.91.154.185) |
| 11:32.46 | Vinsik | qdk: does asterisk support SIP-T? If not, will it ever? |
| 11:32.54 | Vinsik | damn |
| 11:32.55 | clive- | whats sip-t? |
| 11:33.26 | Vinsik | SIP-T: Session Initiation Protocol for Telephones RFC3372 |
| 11:33.45 | Vinsik | encapsulated in sip message |
| 11:34.16 | clive- | supports SIP, never heard of the "t" part before |
| 11:35.09 | Vinsik | in my scenario, i was trying to connect astrisk server as trunk, but there is no rtp media before reinvite. The gateway supplier says that asterisk does not support SIP-T thats why it does not work |
| 11:35.59 | clive- | what gateway is this |
| 11:36.13 | Vinsik | airspan itone |
| 11:37.22 | Vinsik | has anybody heard about SIP-T? |
| 11:38.06 | clive- | why dont you try it and see, most stuff using SIP does work with asterisk |
| 11:39.13 | Vinsik | clive-: im sorry, my first question was not as informative. |
| 11:39.37 | Vinsik | clive-: i have the trunk already up. |
| 11:39.54 | Vinsik | clive-: and when i dial out through it, there is no voice for 90sek |
| 11:39.58 | clive- | and its not working ? |
| 11:40.06 | Vinsik | clive-: untill RE-INVITE comes in |
| 11:40.17 | clive- | 90 seconds,,,,silence,...and then it works after 90 seconds? |
| 11:40.21 | Vinsik | yes |
| 11:40.28 | clive- | strange... |
| 11:40.33 | Vinsik | only after another INVITe message from the softswitch |
| 11:40.39 | clive- | well just set the re-invite to be after 1 second |
| 11:40.54 | Vinsik | clive-: they cant |
| 11:41.04 | Vinsik | clive-: and the reinvite must come from the ssw |
| 11:41.35 | Vinsik | clive-: and they say that its asterisk fault. It does not support SIP-T.. so i just wondered maybe someone else had a similar problem or encounter |
| 11:41.54 | clive- | my advice to you,,,just pay someone $100 to fix it for you in asterisk....although I have never heard of such a thing like this before |
| 11:42.33 | clive- | maybe try another version of chan_sip |
| 11:44.10 | Vinsik | chan_sip2 |
| 11:44.12 | Vinsik | ? |
| 11:44.15 | Vinsik | is it stable? |
| 11:44.18 | clive- | yes, its out there |
| 11:44.37 | clive- | if it works, then it works |
| 11:44.38 | clive- | lol |
| 11:44.56 | Vinsik | ;) |
| 11:51.18 | Vinsik | ok |
| 11:51.38 | Vinsik | chan_sip does not support multipart messages |
| 11:51.51 | Vinsik | thats why the problem occures |
| 11:59.22 | clive- | does chan_sip2? |
| 12:11.02 | *** join/#asterisk [Airwolf] (n=airwolf@dsl51B67BC2.pool.t-online.hu) |
| 12:16.16 | Vinsik | clive-: i have to test ;) |
| 12:18.23 | Vinsik | anybody know does current asterisk support multipart/mixed sdp messages? |
| 12:57.35 | *** join/#asterisk ToyMan (n=stuq@sdsl-216-220-117-141.dsl.bway.net) |
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| 13:32.40 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 13:34.22 | *** join/#asterisk ManxPower (i=ewieling@161.sub-70-210-30.myvzw.com) |
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| 13:48.23 | *** join/#asterisk nanotalk (i=Nano@61.94.125.7) |
| 13:49.36 | nanotalk | i want to use my voice modem + mic + headphone to connect to pstn.. do I need to use asterisk, or is there simpler solution? |
| 13:53.32 | ManxPower | nanotalk, Asterisk does not support voice modems |
| 13:53.54 | ManxPower | nanotalk, Use one of the MANY softphones out there. |
| 13:53.55 | nanotalk | ManxPower, any recommendation? |
| 13:54.10 | ManxPower | come to think of it, the softphones don't do that either. |
| 13:54.38 | *** join/#asterisk krausen (n=krausen@2002:18aa:3e3f:0:0:0:0:1) |
| 13:54.49 | ManxPower | Because there is no standard for using the resources on voicemodems basically nothing out there uses them. |
| 13:56.48 | nanotalk | i c |
| 13:57.05 | nanotalk | I thought it's standard :( |
| 13:59.30 | ManxPower | There is enough of a standard API for data and fax, but not for voice. |
| 14:00.32 | *** join/#asterisk uwe (n=uwe@dogbert.palnet.com) |
| 14:00.35 | nanotalk | thanks for your help, ManxPower |
| 14:02.03 | uwe | hello, ive been trying to move asterisk and freepbx from one system to another with identical asterisk and freepbx versions! still i always need to change something in the extensions |
| 14:02.40 | krausen | just copy the extensions.conf file |
| 14:02.43 | uwe | to get them working ! i dont see why although i moved the /etc/amportal, /etc/asterisk the database @ |
| 14:02.44 | uwe | ! |
| 14:02.46 | krausen | and the sip.conf, etc. |
| 14:03.25 | krausen | does it use mysql for some of the storage? If so, you'll want to do a db dump and restore on the new box |
| 14:04.36 | uwe | im not sure of that, mayb it does with freepbx! but i moved the database too, i didnt use mysql dump, but i copied the whole db and fixed permissions! isnt that the same thing? |
| 14:05.16 | krausen | not always |
| 14:05.24 | krausen | depends what db engine you're using |
| 14:05.28 | uwe | mysql |
| 14:05.39 | krausen | if you're using innodb, you'll want to do the dump |
| 14:06.06 | krausen | mysql uses one or more of about 6 engines |
| 14:06.40 | krausen | plus with the dump, you get all the db usernames / passwords / permissions replicated |
| 14:06.41 | uwe | oh ... ic |
| 14:06.46 | kay2 | yo ManxPower |
| 14:07.37 | uwe | well, i will try that, although i tried to restore from a backup created with freepbx on the old machine and it didnt work ether! |
| 14:08.01 | krausen | what is freepbx? |
| 14:09.16 | uwe | krausen, a web interface to ease the management of asterisk, was called amportal |
| 14:09.47 | krausen | never used either |
| 14:09.52 | krausen | just straight * |
| 14:10.38 | krausen | guess I'm a masochist |
| 14:12.06 | kay2 | any PRI guru here? |
| 14:12.33 | *** join/#asterisk postel_ (n=jp@unaffiliated/postel) |
| 14:23.14 | *** part/#asterisk krausen (n=krausen@2002:18aa:3e3f:0:0:0:0:1) |
| 14:24.13 | *** join/#asterisk DaveHope (n=dave@internal.davehope.co.uk) |
| 14:26.12 | DaveHope | Hello all. Am I right in assumingthat "Call from user '200' is 1 out of 0" generally indicates a SIP ReInvite problem ? |
| 14:35.51 | *** join/#asterisk nibbler_de (n=nibbler@some.host.name) |
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| 14:38.33 | *** mode/#asterisk [+o mog_home] by ChanServ |
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| 15:16.37 | Synyn_ | morning folx |
| 15:17.18 | *** join/#asterisk mog_home (n=mogorman@68.62.237.103) |
| 15:17.18 | *** mode/#asterisk [+o mog_home] by ChanServ |
| 15:53.22 | *** join/#asterisk af_ (n=af@ip-164-6.sn2.eutelia.it) |
| 15:57.24 | *** join/#asterisk RoyK (n=roy@gprs-ggsn5-nat.mobil.telenor.no) |
| 15:58.48 | RoyK | <PROTECTED> |
| 15:59.53 | tsurk0 | hello, I have a question |
| 16:00.05 | tsurk0 | an iax client is behind nat and have to connect to asterisk |
| 16:00.29 | tsurk0 | should I redirect any ports to accomplish this? |
| 16:00.50 | EyeCue | nat=yes for the iax client user directive |
| 16:01.00 | EyeCue | if the server is behind nat as well, you'll want to port forward 4569 |
| 16:01.03 | EyeCue | to the server |
| 16:01.44 | tsurk0 | the server is also behind nat, but other clients from the same net connect without trouble (i've redirectet udp 5060 and 4569) |
| 16:02.01 | *** join/#asterisk droops (n=root@wsip-70-184-44-177.pn.at.cox.net) |
| 16:02.02 | tsurk0 | i forgot about this option, i'll check it out |
| 16:02.03 | tsurk0 | thank you |
| 16:03.12 | Sedorox | if the server and clients are behind a nat, and all are on, say 10.2.3.0/24, then you don't need to add the nat option, only its if going through a nat server, like if a client if out on the internet, connecting to the server behind nat |
| 16:05.23 | tsurk0 | Sedorox, yes, the server and the client are "on the both sides" of the NAT |
| 16:08.06 | *** join/#asterisk tempest1 (n=asf@adsl-153-41-19.chs.bellsouth.net) |
| 16:09.09 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 16:09.09 | *** mode/#asterisk [+o denon] by ChanServ |
| 16:20.55 | *** join/#asterisk matteof (n=matteof@217-133-115-71.b2b.tiscali.it) |
| 16:21.00 | matteof | hi all |
| 16:21.15 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 16:21.26 | matteof | I've a one-way audio problem, can someone help me? |
| 16:24.07 | *** join/#asterisk Muppis_ (i=chuck-th@55.56.227.87.static.sylt.siw.siwnet.net) |
| 16:24.37 | matteof | I've a one-way audio problem, can someone help me? |
| 16:24.42 | tsurk0 | EyeCue, works perfectly |
| 16:24.44 | tsurk0 | thank ypu a lot |
| 16:27.15 | Muppis_ | Hello there fellow asterisk users. Im new to asterisk and Im trying to register my thomson modem with MGCP to asterisk. |
| 16:27.45 | Muppis_ | ive googled a bit but its quite quiet about mgcp :( |
| 16:28.19 | eKo1 | That's because MGCP sucks. |
| 16:28.43 | eKo1 | matteof: You're probably having NAT issues. |
| 16:28.55 | Muppis_ | eKo1: yeah, i figured that :P |
| 16:29.03 | Muppis_ | I get alot of Maximum retries exceeded for transaction 6 on [ST716_A14D13] in asterisk |
| 16:29.28 | eKo1 | Muppis_: If and when you do get it working, please feel free to add it to the wiki. |
| 16:29.57 | Muppis_ | does asterisk have its own wiki? |
| 16:30.11 | Muppis_ | or are you referring to voip-info.org ? |
| 16:31.32 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 16:31.39 | matteof | eKo1: it's not possible everything is on the same subnet |
| 16:33.29 | matteof | eKo1: can you help me? |
| 16:35.23 | *** join/#asterisk SpaceBass (n=sp@static-71-251-230-6.rcmdva.fios.verizon.net) |
| 16:36.44 | matteof | eKo1: now it is ok...I've changed canreinvite from yes to no in sip.conf |
| 16:37.56 | eKo1 | Great. |
| 16:38.16 | eKo1 | Muppis_: I'm refering to voip-info.org. |
| 16:38.48 | matteof | thank you for support |
| 16:38.50 | *** part/#asterisk matteof (n=matteof@217-133-115-71.b2b.tiscali.it) |
| 16:40.06 | *** join/#asterisk wundaboy (n=asdf@c-24-21-100-201.hsd1.or.comcast.net) |
| 16:41.18 | Muppis_ | hm, i think i read that asterisk just does mgcp in call manager mode. my modem says call agent address in its config, i figure call agent and call manager is not compatible? |
| 16:43.21 | *** join/#asterisk Terlouw (n=Terlouw@80.126.223.172) |
| 16:46.49 | ManxPower | Call Manager is a Cisco product. |
| 16:46.59 | Muppis_ | ah, okay |
| 16:47.06 | Muppis_ | my bad |
| 16:47.23 | ManxPower | Last I heard Asterisk supports MGCP as a SERVER, but not as a CLIENT. |
| 16:47.29 | russellb | Muppis_: but you're right ... asterisk can only be the master in the MGCP relationship |
| 16:47.36 | russellb | ManxPower: jinx |
| 16:47.49 | ManxPower | Therefore you can connect MGCP phones to Asterisk, but you cannot connect Asterisk to a MGCP Gateway. I don't know if this has changed or not. |
| 16:48.02 | russellb | nope |
| 16:48.41 | Muppis_ | i want to connect a analog phone trough a PSTN-port in my modem that speaks mgcp, so that should be okay? |
| 16:49.15 | ManxPower | Muppis_, Modems do not talk MGCP. Modems talk RS232. |
| 16:49.32 | Muppis_ | this is a adsl2+-modem over ethernet |
| 16:49.34 | ManxPower | Perhaps you mean an MGCP adapter. I.e. a device that connects Analog<->MGCP. |
| 16:49.36 | Muppis_ | i should have told you that |
| 16:49.49 | ManxPower | Muppis_, We don't care what else the device doesn. |
| 16:50.09 | ManxPower | However, those All-In-One devices are frequently locked. |
| 16:50.20 | ManxPower | Now what is your specific issue? |
| 16:52.11 | Muppis_ | my modem says authentication error in its gui |
| 16:52.19 | Muppis_ | not so informative |
| 16:52.24 | ManxPower | Muppis_, What does the Asterisk console show? |
| 16:52.30 | Muppis_ | Verb: 'RSIP', Identifier: '306151622', Endpoint: 'aaln/1@ST716_A14D13', Version: 'MGCP 1.0' |
| 16:52.48 | Muppis_ | lots of those, lots of `retrans_pkt: Maximum retries exceeded for transaction 135 on [ST716_A14D13]` too |
| 16:53.11 | Muppis_ | RSIP means RestartInProgress of what i read in the rfc of mgcp |
| 16:53.33 | ManxPower | Max retries exceeded is usually a NAT issue. |
| 16:53.52 | ManxPower | Is the Asterisk box also your firewall/NAT box? |
| 16:53.58 | *** join/#asterisk sumasuma (n=sumase@cm222.omega183.maxonline.com.sg) |
| 16:54.16 | Muppis_ | asterisk is on my server 192.168.1.64, the modem has 192.168.1.254 |
| 16:54.30 | sumasuma | hi what is the way to play mp3 in asterisk using Playback application ? |
| 16:55.02 | sumasuma | it looks conversion from mp3 to wav is tiresome |
| 16:55.04 | ManxPower | Muppis_, Does your Asterisk server have more than 1 network interface? |
| 16:55.14 | russellb | sumasuma: install format_mp3 from asterisk-addons |
| 16:55.15 | Muppis_ | ManxPower: nope, just one |
| 16:55.38 | ManxPower | Muppis_, is "iptables" or "ipchains" installed on your Asterisk server? |
| 16:55.38 | sumasuma | russelb: where can i get asterisk addons ? |
| 16:55.43 | sumasuma | checkout ? |
| 16:55.46 | russellb | sumasuma: the same place you get asterisk |
| 16:56.00 | russellb | http://www.asterisk.org/downloads |
| 16:56.02 | Muppis_ | ManxPower: im using freebsd, and ive run without firewall and got the same results |
| 16:56.14 | sumasuma | russelb: you saved my time, thanks a million |
| 16:56.24 | ManxPower | Muppis_, run without firewall until you get it working |
| 16:56.30 | russellb | sumasuma: you can send that million to paypal :) |
| 16:56.53 | sumasuma | http://www.asterisk.org/downloads/ <---- page not found |
| 16:57.02 | sumasuma | even without / also |
| 16:57.04 | russellb | well, just click the downloads link at the top of the page |
| 16:57.24 | sumasuma | russelb: thanks it is download |
| 16:57.31 | russellb | ah, oops |
| 16:57.48 | *** join/#asterisk saftsack (n=oliver@p54A7EF22.dip.t-dialin.net) |
| 16:58.42 | Muppis_ | ManxPower: im running without firewall now, it says the same things. do you want log output or something? |
| 16:58.57 | ManxPower | Muppis_, gads no. |
| 16:59.05 | ManxPower | I cannot help you further. |
| 16:59.37 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 16:59.45 | sumasuma | svn checkout http://svn.digium.com/svn/asterisk-addons/trunk asterisk-addons |
| 16:59.54 | sumasuma | is this checkout correct? |
| 17:00.00 | sumasuma | cursor sleeps on my system |
| 17:00.10 | russellb | that's correct for the trunk version |
| 17:00.17 | ManxPower | sumasuma, why not just download the tar.gz? |
| 17:00.37 | sumasuma | oh where is it ? |
| 17:00.52 | sumasuma | ok thanks |
| 17:00.54 | sumasuma | sorry about it |
| 17:00.58 | sumasuma | i was checking it out |
| 17:02.28 | *** join/#asterisk tuxick (n=userMurf@tuxick.xs4all.nl) |
| 17:02.31 | tuxick | lo |
| 17:04.05 | Muppis_ | russellb: sorry to disturb, but do you know anything about mgcp? |
| 17:05.01 | russellb | some, but I can't really help with it right now. |
| 17:05.50 | Muppis_ | russellb: okay, you don't know if it has happened anything to mgcp in the newest asterisk? im using 1.2.9.1 right now |
| 17:06.26 | russellb | that code is very rarely changed |
| 17:06.35 | sumasuma | russelb: million, i will talk with bill gates, since he is going to be out of microsoft, that might help me also ;) |
| 17:08.58 | Muppis_ | russellb: is there a problem with mpq321 hanging around after a bad shutdown of asterisk? |
| 17:09.05 | Muppis_ | mpg321 |
| 17:09.19 | russellb | well, you shouldn't be using mpg321 at all |
| 17:09.34 | Muppis_ | mpg123 then :P |
| 17:09.42 | russellb | *mpg123*, version 0.59r ONLY |
| 17:10.08 | Muppis_ | Version 0.59r (1999/Jun/15). |
| 17:10.35 | Muppis_ | ive shutdown asterisk with ^c a couple of times, i saw that mpg123 was hanging around after that |
| 17:10.36 | russellb | well there should not be a problem there. You should switch to use files mode MOH, anyway |
| 17:10.57 | russellb | Muppis_: you should be stopping it with "stop now" ... |
| 17:11.42 | Muppis_ | russellb: ah, thanks |
| 17:11.52 | *** join/#asterisk saftsack (n=oliver@p54A7EF22.dip.t-dialin.net) |
| 17:12.15 | *** join/#asterisk saftsack (n=oliver@p54A7EF22.dip.t-dialin.net) |
| 17:12.26 | Muppis_ | russellb: could that be the reason i get lots of `Maximum retries exceeded for transaction 3 on [ST716_A14D13]` too? |
| 17:12.32 | russellb | no |
| 17:12.57 | sumasuma | russelb: that format_mp3 really killed my song ! |
| 17:12.58 | russellb | and that is not indicative of a bug |
| 17:13.11 | Muppis_ | russellb: broken software in my modem? |
| 17:13.14 | russellb | sumasuma: your mp3 should be in 8kHz mono |
| 17:13.14 | sumasuma | do i need to convert to any known format ? |
| 17:13.39 | sumasuma | how will i convert that ? |
| 17:13.44 | russellb | Muppis_: perhaps, or it's not responding |
| 17:13.45 | sumasuma | can i used sox ? |
| 17:13.54 | sumasuma | *use |
| 17:14.12 | russellb | sumasuma: look for a README in the asterisk-addons package somehwere that tells you how ... it's in there somewhere |
| 17:14.17 | russellb | i've got to run, folks ... |
| 17:14.22 | Muppis_ | bye :) |
| 17:14.27 | sumasuma | russelb: thanks |
| 17:14.31 | russellb | you're welcome |
| 17:14.56 | tuxick | i've got sjphone working when calling 1000 for demo, but getting NAT/Firewall: blocked when calling external |
| 17:15.21 | tuxick | sjphone and asterisk on same box |
| 17:15.50 | tuxick | it seems to register with ISP proxy ok |
| 17:20.46 | *** join/#asterisk tsurk0 (n=tsurko@85.187.160.157) |
| 17:23.28 | Muppis_ | why does asterisk list my modem as a gateway? that sounds quote wrong. |
| 17:23.41 | Muppis_ | in the `mgcp show endpints command` |
| 17:23.46 | Muppis_ | * endpoints |
| 17:31.15 | *** join/#asterisk RoyKa (n=roy@c-086ce353.04-15-6f736c3.cust.bredband.no) |
| 17:32.19 | Muppis_ | this was weird, i restarted asterisk and i got chan_mgcp.c:3256 handle_request: Unknown verb 'AUEP' received from 192.168.1.64 |
| 17:32.29 | Muppis_ | 192.168.1.64 is the box running asterisk |
| 17:33.09 | *** join/#asterisk SanketMedhi (n=sanket@221.135.151.215) |
| 17:33.45 | Muppis_ | i don't have any softphones or anything like that |
| 17:34.15 | SanketMedhi | ? |
| 17:38.13 | *** join/#asterisk DrkShdw (n=DrkShdw@unaffiliated/drkshdw) |
| 17:38.25 | sumasuma | format_mp3 has some bugs |
| 17:38.42 | sumasuma | anybody has the same problem ?? |
| 17:38.55 | sumasuma | it drops saying |
| 17:39.13 | sumasuma | III_dequantize_sample: mpg123: Can't rewind stream by 2 bits! |
| 17:39.32 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-9-37.cybersurf.com) |
| 17:39.38 | *** join/#asterisk cHr1Zt1An (n=cHr1Zt1A@200.121.175.199) |
| 17:39.54 | *** part/#asterisk SanketMedhi (n=sanket@221.135.151.215) |
| 17:49.21 | *** join/#asterisk albertito (n=net@host254.201-252-3.telecom.net.ar) |
| 17:55.32 | *** join/#asterisk RoyK (n=roy@ti211310a080-15333.bb.online.no) |
| 18:00.38 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 18:01.10 | Toerkeium | hello everyone |
| 18:02.34 | Muppis_ | i did a tcpdump on port 2427 and 2727, the ports that mgcp uses. i can't see asterisk responding to the modem, i changed to nat=yes in mgcp.conf but the problem still persists :-\ |
| 18:02.47 | *** join/#asterisk malverian (n=malveria@gentoo/developer/malverian) |
| 18:05.44 | Muppis_ | it seems that my modem does something quite weird. it connects to port 2727, but asterix outputs that it talks to port :2427 or something :-\ |
| 18:05.53 | Muppis_ | * asterisk |
| 18:06.09 | Muppis_ | no wonder it behaves weird |
| 18:06.28 | *** join/#asterisk I-MOD (i=opticron@68.62.165.168) |
| 18:11.52 | *** join/#asterisk saftsack (n=oliver@p54A7EF22.dip.t-dialin.net) |
| 18:14.47 | *** join/#asterisk RageMax (n=max@c-24-3-181-140.hsd1.pa.comcast.net) |
| 18:16.56 | *** join/#asterisk fulgas (n=fulgas@a81-84-116-1.cpe.netcabo.pt) |
| 18:19.35 | Skarmeth | hi all |
| 18:21.03 | *** part/#asterisk fulgas (n=fulgas@a81-84-116-1.cpe.netcabo.pt) |
| 18:22.09 | Skarmeth | I have a Asterisk system up and running with about 30 extensions (IP 301) and 4 queues, all polycom phones was configured to have only one line and one call per line, but my users frequently receive more than one call from it's queue. The users are part of only one queue, this way I have q1 (4 users), q2 (4 users), q3 (4 users), q4 (10 users), q3 (2 users).... |
| 18:22.33 | Skarmeth | s/I have a Asterisk/I have an Asterisk/ |
| 18:23.51 | Skarmeth | the last queue it's the recepcionist :) but it receive a lot of calls :) |
| 18:24.04 | Skarmeth | it's on his normal behavior |
| 18:24.24 | Skarmeth | but the 4 first queues not |
| 18:27.40 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 18:29.20 | tuxick | what does "SIP/2.0 404 Not Found" mean? |
| 18:29.30 | tuxick | it doesn't say what isn't found :) |
| 18:30.20 | [TK]D-Fender | tuxick : Means the # dialed doesn't match anything |
| 18:30.58 | tuxick | [TK]D-Fender: so a numberplan/extensions.conf thing? |
| 18:31.37 | [TK]D-Fender | tuxick : Or jsut what you dialed |
| 18:33.37 | tuxick | well the number i dialed exists |
| 18:33.56 | tuxick | it's my own landline # :) |
| 18:34.39 | *** join/#asterisk IOscanner (n=IOscanne@c-67-164-154-209.hsd1.tx.comcast.net) |
| 18:36.48 | *** join/#asterisk starlein (i=star@fo0bar.de) |
| 18:37.10 | uchmando | I have a problem with my asterisk. Every incoming calls hangs up as soon as i pick up the call. The only thing I've done was to upgrade asterisk.. pbx.pean.org/conf/ för config-files. |
| 18:37.35 | RoyK | uchmando: diskettstasjonen din er opp ned |
| 18:38.21 | [TK]D-Fender | tuxick : Not according to your dial-plan then |
| 18:38.28 | uchmando | RoyK, Ajfan. :/ |
| 18:39.36 | *** join/#asterisk mog_home (n=mogorman@68.62.237.103) |
| 18:39.37 | *** mode/#asterisk [+o mog_home] by ChanServ |
| 18:40.44 | Synyn_ | whats the max debugging I can start with *? |
| 18:40.51 | Synyn_ | and verbosity |
| 18:40.57 | Synyn_ | -vvvvgcd? |
| 18:43.06 | tuxick | how to get the "sip debug" noise in logfile? |
| 18:43.17 | tuxick | on console it flies by too fast |
| 18:44.03 | Synyn_ | tuxick have you tried redirecting to file in startup? |
| 18:45.20 | tuxick | i meant using syslog/logger.conf |
| 18:45.36 | tuxick | redirecting output of a consoler isn't really the way :) |
| 18:45.48 | Synyn_ | ah ) |
| 18:46.49 | RoyK | 4x5 is the film format.... in inches.... |
| 19:00.08 | *** join/#asterisk tuxes (i=Deloreor@128.238.140.44) |
| 19:00.14 | tuxick | ugh what was the name of the x-lite executable? |
| 19:00.24 | tuxick | docs don't bother telling |
| 19:01.39 | tuxick | got it |
| 19:16.08 | Toerkeium | anyone knows how can I make a call TO exten jonh from exten paul from CLI ? |
| 19:16.40 | *** join/#asterisk tsurk0 (n=tsurko@85.187.160.157) |
| 19:20.25 | *** join/#asterisk Eggplant (i=No@dsl-72-19-47-190.cascadeaccess.com) |
| 19:20.27 | tuxick | whatever i add to extensions.conf i keep getting 404 |
| 19:20.57 | tuxick | except "100" for my own client :) |
| 19:24.34 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 19:26.07 | tuxick | \o/ |
| 19:33.03 | ManxPower | No. |
| 19:33.13 | ManxPower | I don't believe you can do that in the CLI |
| 19:33.19 | ManxPower | Do it from a softphone |
| 19:33.37 | russellb | you will be able to in 1.4 with the originate CLI command |
| 19:33.40 | russellb | fwiw ... |
| 19:34.05 | ManxPower | russellb, originate a call as a specific device? |
| 19:34.13 | ManxPower | i.e. as a sip.conf section? |
| 19:34.47 | russellb | yeah, something like ... originate SIP/john application Dial SIP/paul |
| 19:35.04 | ManxPower | Ah. Why? |
| 19:35.24 | russellb | same functionality you get with call files, or the originate manager command |
| 19:35.26 | russellb | just from the CLI |
| 19:35.36 | russellb | it's nice for testing |
| 19:36.13 | Toerkeium | thanks russellb |
| 19:37.02 | *** join/#asterisk rajiv (n=irc@gentoo/developer/rajiv) |
| 19:38.27 | russellb | Toerkeium: like i said, this is a future feature for 1.4, it's not in 1.2 |
| 19:40.17 | Toerkeium | when it's going to be released? as you said, that's nice for testing |
| 19:40.20 | *** join/#asterisk SarahEmm (n=sarahemm@MTL-HSE-ppp159791.qc.sympatico.ca) |
| 19:40.34 | russellb | i'm not sure ... when it's ready |
| 19:40.42 | Toerkeium | :) |
| 19:40.44 | russellb | still some big features left to get finished |
| 19:41.29 | *** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it) |
| 19:41.33 | Toerkeium | any place to read about it ? |
| 19:41.40 | Toerkeium | the new features? |
| 19:41.45 | russellb | not yet. i need to start working on that, though ... |
| 19:42.07 | Toerkeium | seems like need some help :) |
| 19:42.16 | russellb | heh, yep |
| 19:42.47 | russellb | svn diff http://svn.digium.com/svn/asterisk/branches/1.2 http://svn.digium.com/svn/asterisk/trunk |
| 19:42.52 | Bobcat_1966 | Hello All, trying to figure out why ENUM Lookup does not work in Trunk, It worked fine until I went from 1.2 stable to svn trunk...any thoughts? |
| 19:42.53 | russellb | i bet that would take a really long time :) |
| 19:43.03 | Toerkeium | let s see |
| 19:43.03 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.71.61) |
| 19:43.04 | file | russellb: do it internally :D |
| 19:43.06 | russellb | Bobcat_1966: i believe the arguments to that function changed |
| 19:43.21 | russellb | file: i wonder what the diffstat would look like |
| 19:43.25 | ManxPower | Bobcat_1966, Setting up a test system? |
| 19:43.44 | Bobcat_1966 | thats what I figured but wanted to check, I have a test system |
| 19:44.11 | Bobcat_1966 | thanks not a big deal, just learning |
| 19:44.21 | ManxPower | "show application enumlookup" |
| 19:44.30 | russellb | file: it's running :) |
| 19:44.35 | Bobcat_1966 | let me try that |
| 19:44.50 | russellb | show function ENUMLOOKUP, actually .... |
| 19:45.29 | *** join/#asterisk Vinsik (n=vinsik@dsl-145-16-216-83.maxinetti.fi) |
| 19:45.34 | Vinsik | hey! |
| 19:45.39 | Bobcat_1966 | Thanks |
| 19:45.51 | *** join/#asterisk X-Gen (n=X-Gen@dsl-145-216-90.telkomadsl.co.za) |
| 19:46.03 | russellb | file: the diff is 8.6 MB |
| 19:46.18 | Vinsik | does anybody know about multipart/mixed SIP messages? |
| 19:46.32 | russellb | from 1.2 to trunk ... 602 files changed, 147968 insertions(+), 67679 deletions(-) |
| 19:47.17 | Vinsik | question is simple. When declaring Content-Type: multipart/mixed; boundary="idstring" .. <= can there be " signs? |
| 19:47.19 | Toerkeium | and still work left |
| 19:47.19 | *** join/#asterisk daysmen3 (n=primus@host86-139-113-254.range86-139.btcentralplus.com) |
| 19:47.23 | Toerkeium | nice |
| 19:47.55 | Vinsik | or does it have to be Content-Type: multipart/mixed; boundary=idstring |
| 19:52.39 | xbmodder_newlapp | Nutzungsvereinbarung |
| 19:52.50 | xbmodder_newlapp | Knoppix5 is out? |
| 19:56.12 | *** join/#asterisk enjay- (i=enjay-@wsip-24-249-169-168.ph.ph.cox.net) |
| 19:56.28 | enjay- | afternoon.. |
| 19:56.54 | *** join/#asterisk Strom_C (n=strom@eaglerock-cuda1-68-67-23-252.vnnyca.adelphia.net) |
| 19:57.00 | Strom_C | yo |
| 19:57.28 | enjay- | sup Strom |
| 19:57.47 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 19:57.51 | Strom_C | do any of you have experience configuring an spa-3102 for voip-to-pstn dialing? Every time I try to place a pstn call from the asterisk box, the spa-3102 returns a 404 |
| 19:59.17 | Strom_C | the documentation is, of course, bad |
| 20:01.57 | *** join/#asterisk JunK-Y (n=junky@modemcable205.175-81-70.mc.videotron.ca) |
| 20:06.08 | *** join/#asterisk Hmmhesays (n=Neg@24-117-135-28.cpe.cableone.net) |
| 20:09.54 | *** join/#asterisk Mercestes (n=Merceste@216.54.143.2) |
| 20:16.15 | enjay- | Is it possible to tell a IAX trunk what zap group to use when dialing out? I.e. I have a trunk coming into my server I want it to use group=5 how do I define in my extensions.conf that ONLY that trunk can use group 5? |
| 20:16.17 | *** join/#asterisk mishehu (i=mishehu@cshells.shavedgoats.net) |
| 20:16.20 | tuxick | Strom_C: i just learn that means there's no match in extensions.conf |
| 20:16.38 | tuxick | the 404 that is |
| 20:17.02 | tuxick | so maybe check if right context= is used |
| 20:17.14 | ManxPower | 404 means "not found" or in Asterisk terms "extension not found" |
| 20:18.23 | tuxick | i'm now trying to figure out why i get "busy" signal when calling my SIP account from landline |
| 20:18.48 | tuxick | sip debug shows some noise when i call so i guess it's not a port forwarding issue? |
| 20:19.22 | tuxick | also suggests ISP is forwarding call |
| 20:23.04 | tuxick | or am i really suppose to forward the all those rtp ports? |
| 20:23.29 | *** join/#asterisk arcy (n=arcanum@ppp140-157.adsl.forthnet.gr) |
| 20:24.20 | Strom_C | no, the 404 comes from the sipura |
| 20:24.31 | Strom_C | not from the asterisk box |
| 20:26.14 | enjay- | Is it possible to tell a IAX trunk what zap group to use when dialing out? I.e. I have a trunk coming into my server I want it to use group=5 how do I define in my extensions.conf that ONLY that trunk can use group 5? |
| 20:27.40 | JunK-Y | do u guys have any script for the melissa db? |
| 20:35.04 | ManxPower | Then the dialplan for the incoming PSTN port is not correct. |
| 20:35.25 | ManxPower | enjay-, only have g5 in the |
| 20:35.25 | Strom_C | ~centosbug |
| 20:35.32 | jbot | methinks centosbug is a problem with the latest Centos kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/`uname -r`-`uname -m`/include/linux/spinlock.h" |
| 20:35.32 | ManxPower | Dial line |
| 20:41.44 | enjay- | ManxPower; I have other stuff dialing on that box as well though, but I want everything on that IAX trunk to go over a specific zap group and nothing else to go over it.. |
| 20:43.57 | ManxPower | enjay-, then put the IAX ""trunk" in a different context. |
| 20:46.23 | *** join/#asterisk RoyK (n=roy@ti211310a080-15333.bb.online.no) |
| 20:46.43 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 20:47.04 | enjay- | ok I'll try that.. |
| 20:47.38 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-9-37.cybersurf.com) |
| 20:55.25 | Synyn_ | using asterisk -vvvvvvvvgcd, all I see is a general timeout for a registered sip using makeing a call, the cdr shows no answer, is there any way to get more debugging info on this? |
| 20:56.15 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-152-23-125.red.bezeqint.net) |
| 20:59.49 | *** join/#asterisk De_Mon (n=de_mon@fl-69-69-142-212.dyn.embarqhsd.net) |
| 21:02.42 | *** join/#asterisk eBody (n=none@adsl-69-153-29-102.dsl.snantx.swbell.net) |
| 21:06.56 | *** part/#asterisk clive- (n=pirch@dsl-145-1-175.telkomadsl.co.za) |
| 21:10.09 | *** join/#asterisk burnproof (n=burnproo@210.213.199.250) |
| 21:12.30 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 21:12.53 | burnproof | hello guys, anyone experience weird problem on attended transfer on asterisk-1.2.10? |
| 21:12.58 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 21:13.18 | burnproof | as soon as i press a single digit it says Jul 24 05:42:37 WARNING[13119]: res_features.c:844 builtin_atxfer: Did not read data. |
| 21:13.24 | burnproof | ? |
| 21:13.47 | burnproof | on SVN trunk i don't experience this problem thou |
| 21:14.29 | Vinsik | any developer online here? |
| 21:17.15 | Qwell | Vinsik: What do you need? |
| 21:18.10 | enjay- | Thanks ManxPower.... |
| 21:18.25 | *** part/#asterisk uchmando (i=peter@pean.org) |
| 21:18.30 | russellb | Qwell: your soul! |
| 21:19.06 | enjay- | vpdtmfsupport disabled by default in zap1.2.7... |
| 21:19.07 | burnproof | Good day russellb: can i ask a question please? |
| 21:19.15 | enjay- | is that due to meetme? |
| 21:19.51 | enjay- | on the wct4xxp mod rather.. |
| 21:20.22 | *** join/#asterisk Assid (i=assid@203.115.83.214) |
| 21:20.23 | Assid | heya |
| 21:22.25 | *** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 21:23.31 | Toerkeium | Guys, does this topology is correct? : |
| 21:24.03 | *** join/#asterisk jmang (i=jordan@wnpgmb09dc1-117-137.dynamic.mts.net) |
| 21:24.42 | jmang | good afternoon all. I find myself frustrated trying to connect two asterisk servers. |
| 21:25.25 | Assid | jmang: why ? they are fun |
| 21:25.35 | Toerkeium | extension 2 > Analog PBX > extension 1 > linksysPAP > Ethernet-to-* > Voip-provider > call termination ? |
| 21:26.11 | Vinsik | qwell: i have quick question about multipart sip messages |
| 21:26.16 | Toerkeium | so, from extension 2 I call to extension 1 (all this with analog PBX), extension 1 connects to linksysPAP, and linksysPAP to *, and then asterisk make the call through VOIP |
| 21:26.18 | Toerkeium | is this correct? |
| 21:26.27 | jmang | I have a server connected in a data center with a very high bandwidth connection. It has several IVR's and handles call queing. I want to have agents that are connected to a server in my office connect to the queues via an IAX2 trunk. |
| 21:27.01 | jmang | My problem is, I don't understand where to start. |
| 21:27.23 | Assid | jmang: i dont think you can have queuing and iax calling.. since the agents would have to be logged into the server in the DC |
| 21:27.50 | Assid | i would think the agents need to register with the server directly |
| 21:28.00 | Assid | but then i could be wrong |
| 21:28.03 | file | jmang & Assid: nope it's possible |
| 21:28.07 | jmang | The documentation on connecting servers seems rather vague to me. My understanding is that this should work if I provide an extension to log them into the server. |
| 21:28.22 | file | what type of agents are you using? |
| 21:28.33 | jmang | The are SIP soft phones. |
| 21:28.41 | file | I mean what sort of agents in Asterisk |
| 21:28.59 | Assid | file: how would the agents be registered online and how would the box know when te agent is busy if the call needs to be transferred to the asterisk box on the end point |
| 21:29.15 | jmang | um...? I use AgentCallbackLogin. |
| 21:29.16 | file | Assid: that's why I'm asking what type of agents |
| 21:29.34 | file | still may be possible |
| 21:30.24 | Assid | okay this is something even i would like to know how it could be done |
| 21:30.42 | jmang | I understand that by setting up peering I would basically be connecting the two server's dialplans. |
| 21:30.57 | Assid | when the the call queue would have remain on the main server |
| 21:31.04 | jmang | So I could dial an extension on DC server from a softphone connected to office server. |
| 21:31.07 | file | Assid: dynamic queue members, chan_local, groups, chan_iax2 |
| 21:31.23 | Toerkeium | guys, anyone could tell me if what I am trying to do is possible? |
| 21:31.24 | Assid | file: even then |
| 21:31.52 | jmang | This whould register the agents, then, the would be calledback via a peer connection from DC server to Office server. |
| 21:32.47 | jmang | The point of this is that we have low bandwidth(1Mbit) connection at the office but need to handle A LOT of incomming calls. DC has 100Mbit connection for queues. |
| 21:33.54 | jmang | file: you seem to think this could be done? |
| 21:34.01 | Assid | normally i would probably handle this by sending it to the queue on the office server.. and what i would do is put a channel limit.. that way anything over a limt say 10 calls.. id send them to voicemail or have them wait in queue ont he DC |
| 21:34.25 | *** join/#asterisk hads (n=hads@mail.nice.net.nz) |
| 21:34.38 | Assid | whena slot opens up in the office quue.. send it to the office queu |
| 21:35.02 | jmang | How would I make a queue in the office, an agent for a queue in the DC? |
| 21:35.46 | file | ugh, the only way to know if this would work is to try it out |
| 21:35.51 | file | in theory I think it would |
| 21:35.52 | Assid | the simplest way would be use a holding / parked area or something.. and then pickup the call from the agents extension |
| 21:35.59 | Assid | using Pickup() |
| 21:36.15 | Assid | thats the simplest way i can think about |
| 21:36.25 | Assid | that is if you want to use a queue |
| 21:36.40 | Assid | else.. if you want to send EXCESS calls to voicemail -- that would be easier |
| 21:36.53 | jmang | I can't send excess calls to VM. |
| 21:37.01 | file | jmang: so you've got two ways of doing this... |
| 21:37.04 | Assid | i mean you really dont want people to wait too long.. |
| 21:37.40 | Assid | i mean you really dont want more than a few calls being run off a 1mbit line. so you need to throttle everything at the DC |
| 21:37.42 | jmang | assid: I know, but I don't make the decisions. |
| 21:38.10 | jmang | assid: That's the whole point of this setup. |
| 21:38.10 | Assid | jmang: ask them if you can send to voicemail, IMHO, that might be a bit better |
| 21:38.22 | Assid | else... put them in parked calls and pick them up |
| 21:38.37 | jmang | assid: 5 min hold time before sending to VM. |
| 21:39.04 | jmang | file: what are these two ways...? |
| 21:39.10 | Assid | okay in that case, use park and pickup |
| 21:39.13 | Assid | and you should be fine |
| 21:39.20 | Assid | that way the agent picksup the call when hes ready |
| 21:40.00 | Assid | your final alternative would be have the sip clients register with the DC directly |
| 21:40.04 | jmang | okay. That could work. Park them in a queue on the DC, and pick them up from there. |
| 21:40.06 | Assid | and use only 1 queue |
| 21:40.42 | Assid | that way you never exceedd your bandwith more than nthe number of active calls |
| 21:40.48 | jmang | The other side of this, is that we actually have 12 different queues being handled by a group of agents. |
| 21:41.34 | Assid | have your DC host the actual queues.. |
| 21:41.49 | Assid | that would take care of everything |
| 21:42.21 | jmang | Yes, but I don't want the clients registering with the server in the DC. |
| 21:42.43 | Assid | cause if you use agentlogin ...... you need agentlogin on the DC for EACH agent n the office anwyays |
| 21:42.52 | Assid | else the queue wont know when to send the call |
| 21:43.40 | jmang | Yes, but I want the queue to send the call to the agent through an IAX trunk link to the server in the office. |
| 21:44.26 | jmang | It seems like it should possible. |
| 21:44.44 | Assid | well.. thats the thing |
| 21:44.51 | Assid | the queue checks for agents which have logged in |
| 21:45.04 | Assid | if you have 1 agent -- viz thew iax link |
| 21:45.08 | Assid | that will always be busy |
| 21:45.19 | Assid | HOWEVER.... |
| 21:45.32 | Assid | you could try it.. since normally.. sip devices return 'busy here' |
| 21:45.46 | Assid | but if you use iax.. ad channel-limit .. you could actually get away with it |
| 21:46.34 | Assid | you just have to make set the agent channel (the one set with agentcallbacklogin) manually to the IAX path |
| 21:46.49 | Assid | err.. make sure to set |
| 21:47.34 | Assid | hello? |
| 21:47.37 | Assid | !ping !pong |
| 21:47.52 | jmang | hello |
| 21:48.09 | Assid | went quiet there |
| 21:48.17 | jmang | just reading. |
| 21:48.18 | file | I'm just trying to warp your idea through my head |
| 21:49.04 | Assid | read the part after however... |
| 21:49.58 | file | I'm doing up... a page... on mine |
| 21:50.08 | Assid | remember.. we work on single registration, multiple lines.. even on sip phones.. that means a single phone can handle multiple lines.. |
| 21:50.22 | Assid | file ? |
| 21:50.27 | file | my idea |
| 21:50.43 | Assid | oh ok |
| 21:50.49 | Assid | i thin this is totally doablew |
| 21:50.56 | Assid | with little effort actually |
| 21:51.13 | jmang | This all seems a little more advanced then my skills at the momement. |
| 21:51.22 | Assid | nah |
| 21:51.23 | jmang | I'm relatively new to asterisk |
| 21:51.35 | Assid | do you know how to setup call queuing? and agents? |
| 21:52.01 | jmang | I have done it, it's what is currently running in our office. |
| 21:52.31 | *** join/#asterisk trbldwine (i=troubled@71.194.161.170) |
| 21:52.53 | *** join/#asterisk zotz (n=zotz@24.244.133.115) |
| 21:53.05 | Assid | okay.. thats all there is to it |
| 21:53.18 | Assid | make an iax connection.. put a channel-limit of say 10.. |
| 21:53.37 | jmang | An IAX connection to where, from where? |
| 21:53.38 | Assid | then make a queue on the DC.. with 5 minute time out and forking off to voicemail after that |
| 21:53.50 | Assid | iax connection between DC and office |
| 21:54.28 | file | http://pastebin.ca/97593 |
| 21:54.55 | jmang | OK, what "type" is that? user at the office end? and peer at the DC? |
| 21:55.24 | Assid | as for the members themselves.. in agents.conf .. USE IAX2/user:pass@officeconnection/extensiontoqueue instead of Agent/1234 |
| 21:55.39 | Assid | in the queues.conf |
| 21:55.44 | Assid | ot agents.conf |
| 21:56.12 | Assid | not even |
| 21:56.52 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 21:57.07 | jmang | ok, I think I'm starting to get it. |
| 21:57.20 | Assid | http://www.voip-info.org/wiki/view/Asterisk+cmd+Queue <-- here |
| 21:57.26 | file | I still find Assid's idea overly complex |
| 21:57.29 | Assid | A member assigned to a queue ("member => Agent/1234" etc above) can be a phone (e.g. "member => SIP/phone1"). |
| 21:57.50 | Assid | therefore.. in queues.conf .. use IAX2/user:pass@officeconnection/extensiontoqueue instead of Agent/1234 |
| 21:57.50 | file | but meh |
| 21:58.12 | Assid | and you use iax2's channel-limit to limit the number of connections.. |
| 21:58.31 | Assid | anything over 10 calls wont allow it.. and i think would return 'busy here' |
| 21:58.58 | Assid | its pretty simple actually |
| 21:59.13 | Assid | hrmm.. i guess that vodka is getting outta my system |
| 21:59.27 | Assid | what did you make me do!?!?!? you made me think again !!! :( |
| 21:59.55 | jmang | So I make queues on both servers, and a trunk with a 10 call limit to transfer b/w queues. I think I get it, agents connect only to office server, and connection b/w dc and office is always setup. |
| 22:00.14 | Assid | yep |
| 22:00.17 | Assid | now you get it |
| 22:00.32 | Assid | queue remains ont he DC and in office.. |
| 22:00.34 | jmang | If I have multiple queues I'll just use multiple IAX channels with lower limits. |
| 22:01.06 | Assid | even if you have 5 agents in the office.. you can have 10 channels, thereby leavin the people in a smaller 2nd queue |
| 22:01.15 | jmang | Right. |
| 22:01.42 | Assid | its like a big line.. then a short break.. and then a smaller line.. |
| 22:02.03 | jmang | I wonder about having this work with multiple queues. |
| 22:02.11 | Assid | big line.. you put a 3 limit time out.. cause really.. they are gonna hangup anwyays.. |
| 22:02.43 | Assid | the smaller line.. you put a small announcemment and make the queue with 5 minutes.. since they are closer to the finish line. |
| 22:02.59 | Assid | jmang: i dont see a problem |
| 22:02.59 | jmang | We have 12 queues being handled by the same group of agents. |
| 22:03.12 | jmang | How do I get the Announce through... |
| 22:03.21 | Assid | you can have 1 big line.. then jumping off to 10 different offices with each office being an agent/agency to the DC |
| 22:03.29 | *** join/#asterisk swytch (n=ezcall@d80-170-73-38.cust.tele2.fr) |
| 22:03.52 | Assid | since your gonna be sending the call.. even a simple play would do fine |
| 22:04.43 | Assid | err. playback |
| 22:05.11 | Assid | okay need mroe water.. and then im hitting the sack |
| 22:05.58 | *** join/#asterisk RoyK (n=roy@ti211310a080-15333.bb.online.no) |
| 22:07.06 | Assid | alrite time for me to go beddy bye |
| 22:07.14 | Assid | gnight file, SarahEmm, jmang |
| 22:07.17 | SarahEmm | nini |
| 22:07.33 | file | night |
| 22:09.05 | *** join/#asterisk robin_sz (n=robin@adsl.redpoint.org.uk) |
| 22:09.21 | jmang | g'night |
| 22:09.46 | *** join/#asterisk Eggplant (n=none@dsl-216-155-213-127.cascadeaccess.com) |
| 22:11.34 | jmang | thanks for the help |
| 22:11.39 | file | good luck! |
| 22:15.58 | *** join/#asterisk Egonis (n=Egonis@207.245.14.10) |
| 22:16.08 | Egonis | What is the command to page an extension / channel? |
| 22:16.09 | *** join/#asterisk RoyK (n=roy@ti211310a080-15333.bb.online.no) |
| 22:16.20 | Egonis | how do I use it? I want to page Console/dsp which autoanswers and playback a file |
| 22:19.27 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 22:19.36 | *** join/#asterisk Amilcar_ (n=Email@201.11.187.241) |
| 22:28.15 | *** part/#asterisk Egonis (n=Egonis@207.245.14.10) |
| 22:30.47 | rajiv | any wanpipe users around ? what should /dev/w1g1 be ? |
| 22:33.42 | xachen | Is asterisk PPP optimizable? |
| 22:34.48 | tuxick | ????? |
| 22:40.31 | *** join/#asterisk CANO-1982 (n=alejandr@190.48.66.106) |
| 22:41.03 | *** join/#asterisk hads|home (n=hads@mail.nice.net.nz) |
| 22:56.44 | Toerkeium | guys, I am using voipstunt to place calls, and I realize that all calls use atleast 74 kbps. Is there anyway I can force to use a lowest bitrate codec for all outgoing calls? or it will be determined by the voip provider? |
| 22:59.29 | *** part/#asterisk CANO-1982 (n=alejandr@190.48.66.106) |
| 23:00.11 | *** join/#asterisk RoyK (n=roy@122.80-203-78.nextgentel.com) |
| 23:00.40 | wunderkin | whatever they allow |
| 23:01.32 | RoyK | <PROTECTED> |
| 23:01.37 | *** join/#asterisk r0d3nt (i=r0d3nt@tinfoilhat.net) |
| 23:02.02 | RoyK | ~nickometer r0d3nt |
| 23:02.18 | r0d3nt | screw you. |
| 23:02.29 | RoyK | :) |
| 23:02.46 | r0d3nt | RoyK, if i wanted any shit from you, I'd squeeze your head. |
| 23:03.02 | RoyK | :) |
| 23:03.25 | RoyK | r0d3nt: say hi :) |
| 23:03.34 | r0d3nt | EAD (tm) |
| 23:05.39 | r0d3nt | I don't have to explain my nick to you and definitely not the stupid bot... I've been the ratman/rodent/khu-nyou/nutria/secretsquirrel and variations for over 15 years.... |
| 23:06.03 | RoyK | whatever :) |
| 23:07.46 | SarahEmm | ~nickometer SarahEmm |
| 23:07.56 | SarahEmm | woo! |
| 23:08.37 | RoyK | hi, SarahEmm |
| 23:08.52 | RoyK | SarahEmm: you're not as lame as r0d3nt, i hear? |
| 23:08.54 | RoyK | :D |
| 23:09.15 | nutria | there |
| 23:09.18 | nutria | you fucking happy ??? |
| 23:09.38 | file | ahum |
| 23:09.41 | file | behave you two |
| 23:09.58 | nutria | ~nickometer nutria |
| 23:10.08 | nutria | RoyK, do you approve ?? |
| 23:10.14 | *** join/#asterisk Bobcat_1966 (n=chatzill@cpe-069-132-139-254.carolina.res.rr.com) |
| 23:10.38 | Toerkeium | what's the usual codecs that VOIP providers are using ? |
| 23:10.58 | Vinsik | g711u/a |
| 23:10.59 | RoyK | nutria: indeed |
| 23:11.21 | Toerkeium | Vinsik, what bitrates uses g711u/a ? |
| 23:11.26 | hohum | lots employ either G.729 or G723 as well |
| 23:11.36 | hohum | ulaw and alaw are 64K |
| 23:12.10 | *** part/#asterisk nutria (i=r0d3nt@tinfoilhat.net) |
| 23:12.20 | Vinsik | ulaw/alaw supports dtmf thats why its most common for pstn termination.. but it depends. In my experience g711 is most used. |
| 23:12.20 | Toerkeium | I was told that there was possible to use a codec at 24kbps, not sure which is |
| 23:12.25 | Vinsik | im off to bed :) |
| 23:13.19 | hohum | Vinsik: RFC2833 |
| 23:15.03 | hohum | Toerkeium: you can get down as low as 5-6k |
| 23:15.09 | hohum | before you add IP overhead |
| 23:15.24 | hohum | but you'll never achieve decent mos ratings either |
| 23:16.36 | Toerkeium | hohum, what would the combination of dtmfmode and codec? |
| 23:16.45 | Toerkeium | to altast go to 24 kbps |
| 23:17.22 | hohum | DTMF rfc2833 and if you want to get the payload size down to 24k then you have lots of options |
| 23:17.29 | hohum | gsm, g723, g729 |
| 23:17.49 | hohum | what are you connecting to? |
| 23:17.55 | Toerkeium | to voipstunt |
| 23:18.13 | Toerkeium | tried with allow=gsm only, but it doesn't low from 74 kbpd |
| 23:18.17 | Toerkeium | kbps* |
| 23:19.03 | hohum | try 723 or 729 |
| 23:19.16 | Toerkeium | let me see |
| 23:19.52 | Toerkeium | chan_sip.c:2552 sip_write: Asked to transmit frame type 4, while native formats is 1 (read/write = 1/1) |
| 23:19.56 | SarahEmm | 729 needs a license tho |
| 23:20.05 | Toerkeium | if I set g623 |
| 23:20.12 | Toerkeium | g723 |
| 23:20.35 | hohum | you need a license for g723 too, don't you? |
| 23:21.34 | Toerkeium | damn |
| 23:22.41 | *** join/#asterisk fugi (n=fugi@ultra.bl.org) |
| 23:23.19 | Toerkeium | g723/729 with licence, and gsm at 74kbps, I am dead |
| 23:23.51 | hohum | there's no way GSM is 74Kbps |
| 23:24.42 | Toerkeium | that's what I see in my bandwidth shaper, while using a voip service with a ISP which uses g729 I see 24kbps |
| 23:24.56 | SarahEmm | last i heard * only supported passthru for g723 |
| 23:25.08 | SarahEmm | and *processing* 729 needed a license, passing it through did not |
| 23:25.11 | SarahEmm | that was awhile back tho |
| 23:25.24 | hohum | Toerkeium: there's no way, that's incorrect |
| 23:25.41 | hohum | GSM is 13Kbps |
| 23:25.49 | hohum | and when you ad IP overhead, 15 at the MOST |
| 23:26.08 | Toerkeium | g729 gives me the same than 723: Jul 23 20:30:50 WARNING[1944]: chan_sip.c:2552 sip_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 256/256) |
| 23:26.46 | *** join/#asterisk CANO-1982 (n=alejandr@190.48.66.106) |
| 23:26.51 | Toerkeium | hohum: Do I need anything alse apart of configuring dtmfmode=rfc2833 and allow=gsm ? |
| 23:27.57 | *** part/#asterisk CANO-1982 (n=alejandr@190.48.66.106) |
| 23:28.02 | hohum | shouldn't |
| 23:28.36 | Toerkeium | <hohum> and when you ad IP overhead < what does this mean? |
| 23:29.05 | hohum | well an RTP packet isn't just the audio stream |
| 23:29.21 | hohum | there's IP and UDP overhead that you have to add into your bandwidth figures |
| 23:29.34 | hohum | headers, etc |
| 23:29.42 | Toerkeium | oh, yes.. |
| 23:29.45 | hohum | generally 2k (if it is compressed) or 4-5k uncompressed |
| 23:30.04 | Toerkeium | yes, I see tha traffic too.. but it's insignifican here |
| 23:30.16 | hohum | so its save to assume that G729 (8Kbps) runs at 12Kbps |
| 23:30.17 | Toerkeium | 2k as you said |
| 23:30.26 | Toerkeium | but gsm at 80kbps |
| 23:30.27 | Toerkeium | heh |
| 23:30.32 | hohum | dude |
| 23:30.36 | Toerkeium | I know I know.. |
| 23:30.42 | hohum | you're reading/doing something incorrect |
| 23:30.44 | wunderkin | ulaw is 80 not gsm |
| 23:30.49 | hohum | GSM does not run at 80+K |
| 23:30.56 | hohum | ulaw is NOT 80k |
| 23:31.02 | hohum | ulaw is 64k |
| 23:31.03 | wunderkin | with overhead |
| 23:31.12 | hohum | no, not even with overhead |
| 23:31.12 | Toerkeium | probably.. asterisk is using another codec |
| 23:31.12 | wunderkin | it is closer to that |
| 23:31.32 | Toerkeium | probably because of voipstunt not having another codec available |
| 23:31.36 | Toerkeium | I should try with another voip |
| 23:31.39 | Toerkeium | provider |
| 23:32.45 | wunderkin | Toerkeium, http://www.asteriskguru.com/tools/bandwidth_calculator.php |
| 23:33.13 | Toerkeium | gsm 13 kbps, nice |
| 23:33.25 | Toerkeium | I don't even expect that much |
| 23:33.45 | Toerkeium | but 80 (totaly) it's absurd, I guess |
| 23:34.00 | wunderkin | Incoming bandwidth used is: 79.63 Kbps for ulaw |
| 23:35.19 | wunderkin | it was probably using ulaw, make sure you have a disallow=all before your allow |
| 23:35.27 | hohum | *shrug* it must be asterisk's implementation of ulaw then |
| 23:35.27 | Toerkeium | yes I have it |
| 23:35.36 | hohum | because in my lab it doesn't run at 80Kbps |
| 23:35.46 | hohum | that's dumb |
| 23:35.47 | wunderkin | um no its just you arent counting all of the overhead |
| 23:35.50 | hohum | look at the spec |
| 23:36.25 | hohum | there's no way in hell that there is an additional 16K of IP overhead unless your network is braindead |
| 23:37.35 | Toerkeium | it's always in UDP 17 |
| 23:40.18 | hohum | http://www.comptechdoc.org/independent/networking/guide/netudp.html |
| 23:40.28 | hohum | Source IP address (32 bits) |
| 23:40.29 | hohum | Destination IP address (32 bits) |
| 23:40.29 | hohum | blank filler(0) (8 bits) |
| 23:40.29 | hohum | Protocol (8 bits) |
| 23:40.29 | hohum | UDP length (16 bits |
| 23:40.37 | *** join/#asterisk Maan (n=mbsat@249.142.77.83.cust.bluewin.ch) |
| 23:40.42 | Maan | hi all |
| 23:41.12 | wunderkin | RTP: 4.69 Kbps UDP: 3.13 Kbps IP: 7.81 Kbps |
| 23:41.46 | Toerkeium | ok: |
| 23:41.56 | hohum | that's 96Bits per packet, right |
| 23:42.05 | hohum | and RTP is usually done in 20ms windows |
| 23:42.08 | Toerkeium | from voipstunt to my * box I have: 18.1kbps (could be now gsm) |
| 23:42.14 | hohum | meaning every second a total of |
| 23:42.16 | hohum | ... |
| 23:42.22 | Toerkeium | and from my softphone to * 80 |
| 23:42.25 | Toerkeium | kbps |
| 23:43.13 | hohum | 50 packets? that doesn't sound right |
| 23:44.03 | Toerkeium | no, from voipstunt to * 18.1kbps, and from my softphone to * 80kbps |
| 23:44.17 | Toerkeium | so the problem is me between * |
| 23:44.25 | Toerkeium | your gsm setup did work |
| 23:44.30 | Toerkeium | I guess |
| 23:44.32 | hohum | BTW bandwidth is cheap |
| 23:44.38 | Toerkeium | not here :) |
| 23:44.44 | hohum | where is "here" |
| 23:44.48 | Toerkeium | Argentine |
| 23:44.54 | hohum | ah |
| 23:44.55 | Toerkeium | it's pretty expensive, including local bandwidth |
| 23:45.00 | hohum | yeah |
| 23:45.20 | hohum | South America is a hodgepodge of fiber, satellite, microwave and WiMax |
| 23:45.25 | Toerkeium | in fact, some part (because it's divided in 2 NAP) is more expensive than international bandwidth |
| 23:45.39 | *** part/#asterisk fugi (n=fugi@ultra.bl.org) |
| 23:47.12 | hohum | I know how expensive remote regions can be |
| 23:47.19 | hohum | we have stuff all over africa |
| 23:47.25 | *** join/#asterisk saftsack (n=saftsack@p54A7EBDD.dip.t-dialin.net) |
| 23:47.33 | hohum | and in most countries in Africa there's no infrastructure |
| 23:47.56 | hohum | a great deal of the continent is accessible by sat links only |
| 23:48.01 | Toerkeium | well.. it's not a problem of infrastructure here |
| 23:48.12 | hohum | which is the most expensive type of bandwidth you can buy |
| 23:48.13 | rajiv | whats a quick way to get asterisk to play back "three zero two" if ${EXTEN} contains 302 ? |
| 23:48.14 | Toerkeium | it's a telco "monopolio" sorry,, don't know the translation |
| 23:48.24 | hohum | monopoly |
| 23:48.27 | SarahEmm | monopoly :) |
| 23:48.30 | Toerkeium | heh :) |
| 23:48.34 | *** join/#asterisk Luke-Jr (n=luke-jr@user-0c93tin.cable.mindspring.com) |
| 23:49.04 | hohum | Toerkeium we have a similar issue in brasil |
| 23:49.11 | Toerkeium | it's abour USD 800 1Mbps indiscrimined (local (g4 and NAP)) and international |
| 23:49.22 | hohum | because Embratel has like a 95% market share |
| 23:49.26 | Toerkeium | most cheap, it's about USD 400 |
| 23:49.41 | Toerkeium | and if you have problems, "anda a cantarle a gardel" |
| 23:49.42 | hohum | but Embratel was just recently purchased by Telmex and we have a good relationship with them |
| 23:49.51 | rajiv | oh i think i want saynumber() or saydigits() |
| 23:49.54 | Toerkeium | which means, if you have problems "go to sing to gardel" |
| 23:49.55 | Toerkeium | ;) |
| 23:50.08 | *** join/#asterisk Winkie (n=urmom@cpc3-stre1-0-0-cust656.bagu.cable.ntl.com) |
| 23:50.10 | hohum | who is the local incumbent? |
| 23:50.39 | *** join/#asterisk techie (n=gus@voipops.net) |
| 23:51.02 | Toerkeium | well.. G4 is: telecom, telefonica, impsat and prima (grupo clarin), they have the 60/70% of internet users |
| 23:51.14 | Toerkeium | in the other hand all other medium ISP (NAP) |
| 23:51.20 | hohum | hehe |
| 23:51.20 | hohum | yeah |
| 23:51.24 | b4ka | oh, another argentinian |
| 23:51.26 | hohum | we know the Telefonica guys well |
| 23:51.43 | Toerkeium | yeah, that's what monopoly is basically :) |
| 23:52.02 | Toerkeium | but I guess they don't have too much time doing this |
| 23:52.06 | Toerkeium | thanks to VOIP :) |
| 23:52.11 | Toerkeium | actually, I use only voip lines |
| 23:52.25 | Toerkeium | but ok, I pay them anyway. they terminate the calls |
| 23:52.54 | Toerkeium | are you from AR b4ka ? |
| 23:52.57 | b4ka | yep |
| 23:53.01 | Toerkeium | where? |
| 23:53.06 | b4ka | capital federal |
| 23:53.21 | Toerkeium | you know, capital federal is pretty big ;) |
| 23:53.36 | b4ka | i live in villa del parque. why? |
| 23:53.43 | Toerkeium | just courious |
| 23:53.53 | b4ka | so, you use asterisk? |
| 23:53.59 | Toerkeium | starting |
| 23:54.07 | b4ka | :) |
| 23:54.09 | Toerkeium | but I will sleep today |
| 23:54.16 | b4ka | only voip? |
| 23:54.17 | Toerkeium | I finished my test configs |
| 23:54.26 | Toerkeium | mixed with my analog PBX |
| 23:54.33 | b4ka | heh |
| 23:54.50 | b4ka | where do you work, curious :D |
| 23:55.01 | Toerkeium | exten analog PBX > linksysPAP > asterisk > voipstunt |
| 23:55.12 | Toerkeium | that's what I did, and today I feel like an expert :P |
| 23:55.18 | b4ka | heh |
| 23:55.29 | Toerkeium | tomorrow will be an idiot like everyday... |
| 23:55.38 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 23:55.42 | b4ka | i use analog lines + iax provider + e1 with voice channels |
| 23:55.47 | Toerkeium | I am running a very small web hosting company |
| 23:55.50 | b4ka | and sip/iax phones |
| 23:56.23 | Toerkeium | who provides you the e1 ? |
| 23:56.29 | b4ka | telmex heh |
| 23:56.45 | Toerkeium | in V. del parque ? |
| 23:56.50 | b4ka | umm |
| 23:56.57 | b4ka | i dont have this at home :P |
| 23:57.36 | Toerkeium | it would be a nice toy |
| 23:57.46 | b4ka | edxpensive toy |
| 23:57.55 | Toerkeium | how much for the e1 ? |
| 23:58.00 | b4ka | dunno |
| 23:58.03 | b4ka | but lots |
| 23:59.25 | b4ka | you have iplan? :D |
| 23:59.32 | b4ka | where are you? |
| 23:59.38 | Toerkeium | yeah, at work |
| 23:59.43 | Toerkeium | iplan and datco |