| 00:02.25 | *** join/#asterisk SirThomas_Home (n=tomc@209-169-199-174.us.warpdriveonline.com) |
| 00:04.45 | Gouroutrash | i like functions and price of Thomson 2030, but the quality and design are not very good, and medium compatibility but in progress |
| 00:05.06 | Gouroutrash | i dream of a cisco with blf, pickup, fast dials |
| 00:05.20 | Gouroutrash | and i go to bed |
| 00:05.23 | Gouroutrash | good night all |
| 00:05.25 | Gouroutrash | ++ |
| 00:06.23 | yannj_fr | [TK]D-Fender : I found my solution for the devstate app |
| 00:07.16 | [TK]D-Fender | yannj_fr, and it was ..... ? |
| 00:07.41 | yannj_fr | Looked at the SVN asterisk/trunck and one of digium developper created func_devstate, similar use |
| 00:10.45 | yannj_fr | well... time to sleep ( 2am)... bye bye |
| 00:11.38 | yannj_fr | (FYI : the developper is russellb) |
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| 00:54.14 | tomcontr3 | hi, can anyone help me with a little doubt that I have with AGIPHP |
| 00:54.25 | tomcontr3 | to be more spesific with GET VARIABLE |
| 00:54.39 | tomcontr3 | I did this: fwrite(STDOUT,"GET VARIABLE CALLSIGN \n"); |
| 00:54.47 | tomcontr3 | then this: $result = trim(fgets(STDIN,4096)); |
| 00:55.01 | tomcontr3 | but the result only shows : 200 no the real value |
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| 00:58.52 | tomcontr3 | any idea? |
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| 01:11.48 | tomcontr3 | ??? |
| 01:12.02 | JT | tomcontr3: why all the question marks? |
| 01:14.39 | tomcontr3 | ohh sorry |
| 01:14.49 | ZaVoid | lol |
| 01:15.00 | tomcontr3 | I was wondering is someone could give me a hand, on a little doubt I have |
| 01:15.09 | tomcontr3 | abot PHPAGI, using GET VARIABLE |
| 01:15.43 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 01:15.57 | tomcontr3 | I did this: fwrite(STDOUT,"GET VARIABLE CALLSIGN \n"); |
| 01:16.01 | tomcontr3 | then this: $result = trim(fgets(STDIN,4096)); |
| 01:16.05 | tomcontr3 | but the result only shows : 200 no the real value |
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| 01:23.35 | blebleble | should hudlite be running on 5038 or 6600? |
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| 01:25.18 | JT | tomcontr3: yeah saw that already, likely that no-one is aware of the answer |
| 01:28.57 | Qwell | random question: can a phrase be copyrighted? |
| 01:29.15 | JT | no but it can be trademarked |
| 01:29.17 | voipnet-tech | can be trademarked |
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| 01:29.24 | voipnet-tech | JINX |
| 01:29.26 | Qwell | that's the one.. |
| 01:29.38 | voipnet-tech | that is an interesting question |
| 01:30.05 | voipnet-tech | literary works can be copyrighted, and are automatically copyrighted to the author upon completion of said literary work |
| 01:30.32 | voipnet-tech | the question then is, can a single phrase be considered a literary work... |
| 01:30.51 | voipnet-tech | the one line book |
| 01:30.57 | Qwell | say, the phrase: "I'll gladly pay you Tuesday, for a hamburger today." |
| 01:31.17 | blitzrage | GO SENS GO!!! |
| 01:31.18 | Qwell | If somebody were to use that in a commercial product... they could technically be sued, no? |
| 01:32.19 | Qwell | I mean...you can sue anybody, but this particular suit wouldn't even be frivilous :D |
| 01:32.21 | voipnet-tech | http://www.copyright.gov/circs/circ1.html#wnp |
| 01:32.33 | voipnet-tech | Titles, names, short phrases, and slogans; familiar symbols or designs; mere variations of typographic ornamentation, lettering, or coloring; mere listings of ingredients or contents |
| 01:32.38 | voipnet-tech | cannot be copyrighted |
| 01:32.55 | Qwell | ahh |
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| 01:33.00 | JT | Qwell: it costs a bit of money to get a trademark though |
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| 01:33.13 | ZaVoid | yep |
| 01:33.23 | Qwell | JT: right, that's why I was specifically asking about copyright - since it's automatic |
| 01:33.35 | voipnet-tech | if they used an audio clip of the guy from popeye saying that, the audio clip is copyrighted |
| 01:33.59 | Qwell | right |
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| 01:34.44 | voipnet-tech | and those episodes of popeye in their entirety is copyrighted, so using that single phrase is using a portion of copyrighted material i guess |
| 01:34.59 | acowan | hey all... I've got an outbound route extension that has a macro call set to priority 1, I want to be able to include a SETDtmfMode in certain cases with a priority of 0 (or with 1 and move down the Macro to dialout-trunk)... any suggestions? I can't find anything online |
| 01:36.27 | ZaVoid | any of you guys use woot.com |
| 01:37.40 | voipnet-tech | lol: "The Gourmet Coffee Café has a milk frother on the side for frothing milk (obviously) |
| 01:39.04 | JT | amazing |
| 01:39.05 | ZaVoid | yeah |
| 01:39.10 | ZaVoid | i put up a woot off checker |
| 01:39.12 | ZaVoid | www.woot.us |
| 01:39.20 | ZaVoid | got bored today |
| 01:39.27 | ZaVoid | figured crap i own a similar domain name |
| 01:40.42 | *** join/#asterisk tbic (n=tbic@207.148.218.162) |
| 01:41.08 | voipnet-tech | if you had an Aastra SIP phone you could make that into a custom XML application that would show the woot-off on the phone display |
| 01:42.59 | voipnet-tech | i'm doing a demo at a technology tradeshow tommorow, every year i always give away a phone |
| 01:43.00 | voipnet-tech | until now we just had little slips of paper and had a drawing |
| 01:43.00 | voipnet-tech | this year, people will actually enter the contest on the phone itself |
| 01:43.06 | ZaVoid | thats cool |
| 01:43.07 | *** join/#asterisk rickw (n=rickw@ppp-70-128-124-134.dsl.tulsok.swbell.net) |
| 01:43.08 | voipnet-tech | using the XML Screens, enter their name and phone number, and at the end of the day it's a custom drawing |
| 01:43.08 | ZaVoid | what show tomorrow? |
| 01:43.31 | voipnet-tech | it's called MTUGS |
| 01:43.47 | voipnet-tech | the annual tech show here in Maine |
| 01:44.02 | voipnet-tech | maine is pretty low-tech... they only allow 1 day a year for hi-tech stuff |
| 01:44.06 | ZaVoid | oh nice |
| 01:44.09 | ZaVoid | i wanna move to maine |
| 01:44.11 | ZaVoid | i love it there |
| 01:44.22 | ZaVoid | lived in northern NH for a while and spenty many summers in main |
| 01:44.30 | voipnet-tech | i was born here, moved around the states, but came back |
| 01:44.34 | ZaVoid | nice |
| 01:44.35 | voipnet-tech | it's just way better here |
| 01:44.40 | ZaVoid | wherebouts? |
| 01:44.50 | voipnet-tech | windham |
| 01:44.53 | voipnet-tech | 10mi north of portland |
| 01:44.57 | ZaVoid | cool |
| 01:45.04 | ZaVoid | good beer up there |
| 01:45.09 | ZaVoid | my fave is cadilac mountain stout |
| 01:45.22 | voipnet-tech | you had Shipyard? |
| 01:45.43 | ZaVoid | yeah its decent |
| 01:45.50 | ZaVoid | had a new one when i was up in febuary |
| 01:45.56 | ZaVoid | peek or peak brewing it was called |
| 01:46.00 | ZaVoid | really nice nut brown ale |
| 01:46.06 | voipnet-tech | i sold Shipyard Brewery all their VoIP, we host their IP centrex |
| 01:46.12 | ZaVoid | or new to me at least |
| 01:50.12 | blitzrage | what the hell is with my MSN the last two nights.... |
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| 02:15.54 | mjmarrio | can someone please tell me if when the command "iax2 show users" is run that it will display iax2 users defined in an iaxfriends table in the MySQL database? |
| 02:16.57 | ZaVoid | <PROTECTED> |
| 02:17.03 | ZaVoid | it won't |
| 02:17.45 | mjmarrio | tq. The reason I ask is because I need to verify if AsteriskNow will allow users to be defined in iaxfriends table. |
| 02:18.13 | mjmarrio | I have installed add-ons to make cdr work with mysql but I am unsure about iaxfrineds |
| 02:18.27 | mjmarrio | I have defined an account but does not seem to work |
| 02:18.57 | mjmarrio | normally u have to configure before compilihng asterisk to get this to work |
| 02:19.13 | mjmarrio | I am not sure if the AsteriskNow binaries have been configured to allow this |
| 02:24.17 | ZaVoid | is asterisk now based on 1.4? |
| 02:25.07 | mjmarrio | ZaVoid: Is that question directed to me? If so I do know that AsteriskNow is based on 1.4 |
| 02:25.39 | *** join/#asterisk PMantis (n=sswitzer@cpe-74-65-53-178.rochester.res.rr.com) |
| 02:25.47 | mjmarrio | no one seems to want to talk over at the #asterisknow channel |
| 02:26.04 | vAd0r | anyone here use PAP2 V2 |
| 02:26.10 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 02:26.32 | ZaVoid | that sux.. 1.4 sux for heavy iax registrations.. wehad to go back to 1.2 |
| 02:26.41 | PMantis | Is there a way to get Asterisk to write voicemails (/var/spool/asterisk/default/xxx/INBOX/*.wav) with a different umask? |
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| 02:34.44 | elg | can the pickup application be made to work with a ringing queue? |
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| 03:01.29 | obnauticus | I'm running FreeBSD 6.2 release, and trying to install Asterisk from the ports and I get the following error: http://pastebin.ca/524500 |
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| 03:02.04 | infinity1 | how can i do this in asterisk AEL? It doesnt work! PHONES2=${EXT203} |
| 03:02.22 | infinity1 | you can't assign a variable with the contents of another variable! |
| 03:02.39 | mitcheloc | how about a dbput and a dbget? |
| 03:03.02 | infinity1 | thats a ghetto workaround, but maybe. heh |
| 03:03.09 | infinity1 | it should work.... |
| 03:03.16 | infinity1 | it wored in the old language. |
| 03:03.29 | obnauticus | anyone? |
| 03:08.28 | mitcheloc | obnauticus: i don't think freebsd is commonly used to run asterisk |
| 03:09.44 | obnauticus | mitcheloc it's possible though |
| 03:09.55 | obnauticus | I'm compiling speex from source |
| 03:09.58 | obnauticus | so i'll see if this will work |
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| 03:24.05 | luckyone | hello all, I am trying to connect to my running asterisk process, but don't seem to be able to do so... |
| 03:24.10 | mjmarrio | hello PMantis you still there? |
| 03:25.09 | mjmarrio | you could try setting the default mask in the root or asterisk home directory .bashc file |
| 03:25.16 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 03:25.40 | luckyone | when I ps -ef | grep asterisk, I see /usr/sbin/safe_asterisk and /usr/sbin/asterisk both running |
| 03:26.03 | mjmarrio | what command r u using to connect? |
| 03:26.23 | luckyone | sudo asterisk -rv -U asterisk -G asterisk |
| 03:26.44 | mjmarrio | r u using asteriskNow? |
| 03:26.49 | luckyone | no |
| 03:27.02 | luckyone | I compiled and installed * 1.4.4 |
| 03:27.15 | luckyone | I have connected to the console before, I am not sure why it isn't letting me now |
| 03:27.19 | mjmarrio | can you su to the root account? |
| 03:27.32 | luckyone | that is effectively what sudo does |
| 03:27.38 | mjmarrio | did u look in /var/llog/asterisk/messages? |
| 03:27.49 | *** join/#asterisk tsurko (n=tsurko@77.70.24.142) |
| 03:27.51 | mjmarrio | yes "effectively" being the operative word |
| 03:27.59 | luckyone | lol |
| 03:28.17 | mjmarrio | so don't send the boy in to do mans work |
| 03:28.27 | shido6 | hehehe |
| 03:29.23 | luckyone | nothing since 4:52 yesterday in /var/log/asterisk/messages |
| 03:31.52 | mjmarrio | asterisk -rx "restart now" |
| 03:32.30 | mjmarrio | tail -f /var/log/asterisk/messages in another xterm |
| 03:37.40 | luckyone | mjmarrio: Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
| 03:37.54 | luckyone | mjmarrio: where would that have gone? |
| 03:41.08 | luckyone | there is no /var/run/asterisk |
| 03:46.16 | mjmarrio | huh? |
| 03:46.32 | luckyone | mjmarrio: yeah, I don't know either |
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| 03:46.53 | luckyone | mjmarrio: I googled that message I was getting and it seems that a recompile fixed it for some people, so I am going to do that |
| 03:46.59 | mjmarrio | well I would be inclined to ps -ef|grep asterisk|cut -c10-14|xargs kill -9 if I were you |
| 03:47.46 | obnauticus | Is there anything so that like via the web a person can come in and make an account on my pbx |
| 03:47.49 | obnauticus | so that I don't have to do it |
| 03:48.18 | luckyone | mjmarrio: is that crazy funky pipe string saying to kill asterisk? |
| 03:49.19 | blitzrage | luckyone: best not to run anything you don't understand |
| 03:49.26 | luckyone | yeah |
| 03:50.06 | blitzrage | <luckyone> mjmarrio: Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) <-- means asterisk is not running |
| 03:50.27 | blitzrage | shido6: you use dynamic queue members from realtime? |
| 03:51.37 | lee_is_me | where can I find sample sip.cfg, phone1.cfg, etc files? |
| 03:52.39 | luckyone | blitzrage: why did ps show it running then? |
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| 03:52.58 | blitzrage | hung process or something? |
| 03:53.03 | luckyone | blitzrage: and why did my sip phone stay connected until I killed it? |
| 03:53.23 | luckyone | blitzrage: could have been |
| 03:53.32 | blitzrage | that msg almost always means asterisk is not running |
| 03:53.41 | blitzrage | that is the only time in over 5 years I have ever seen that msg |
| 03:53.46 | luckyone | blitzrage: recompiling now, hopefully this fixes it |
| 03:53.57 | blitzrage | huh? |
| 03:54.00 | blitzrage | what version are you running? |
| 03:54.03 | luckyone | blitzrage: that is about 4.99 more years than I have been using * |
| 03:54.11 | luckyone | blitzrage: 1.4.4 |
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| 03:54.28 | mjmarrio | yep |
| 03:54.41 | blitzrage | cd /usr/src/asterisk ; ./configure ; make menuselect ; make install ; make config ; (optional) make samples |
| 03:54.44 | blitzrage | that should work fine |
| 03:54.54 | luckyone | blitzrage: yeah, that is what I am doing |
| 03:55.01 | luckyone | blitzrage: I did a make clean too |
| 03:55.08 | blitzrage | make distclean is better |
| 03:55.25 | mjmarrio | hey obnauticus: YOu should use AsterisNow it has a web admin console for that |
| 03:55.26 | blitzrage | if you're gonna bother with a make clean that is... :) |
| 03:55.44 | luckyone | ok, resintalled |
| 03:56.00 | obnauticus | k mjmarrio |
| 03:56.21 | obnauticus | mjmarrio does it have umm |
| 03:56.27 | obnauticus | i mean does FreePBX have that? |
| 03:56.34 | luckyone | blitzrage: what do you think about starting * with /etc/init.d/asterisk start? |
| 03:56.47 | blitzrage | service asterisk start |
| 03:56.55 | blitzrage | not /etc/init.d/asterisk |
| 03:57.03 | blitzrage | (although that is the file it executes) |
| 03:57.07 | JT | ~freepbx |
| 03:57.20 | jbot | from memory, freepbx is unable to be supported here. It's a complex piece of dialplan and GUI, and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 03:57.30 | luckyone | blitzrage: ok, it started |
| 03:57.52 | blitzrage | going to bed |
| 03:58.03 | blitzrage | 10am -> 12am shift is enough for me |
| 03:58.06 | luckyone | blitzrage: night, and thanks for ideas |
| 04:00.46 | luckyone | mjmarrio: kill off processes, recompiled/installed, started process, checked that it is running, still can't connect to CLI |
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| 04:17.47 | mjmarrio | did u su to root? |
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| 04:27.26 | monstertruck | hi |
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| 04:27.44 | monstertruck | does anybody know of a voip gateway, like the spa3000 that would do iax2? |
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| 04:53.02 | tomcontr3 | hey guys, |
| 04:53.55 | tomcontr3 | does anyone kows if when you set an extension to read any number like this exten => _. |
| 04:54.08 | tomcontr3 | can I set a time limit before that extention calls the function or aplication? |
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| 04:56.13 | carrar | put a wait before it? |
| 04:57.02 | carrar | or as the 1st priority of that extension |
| 04:57.54 | tomcontr3 | my problem is that, when I pres the numbers |
| 04:58.02 | carrar | thats your phone |
| 04:58.09 | tomcontr3 | it takes like 6 to 7 seconds to execute de function |
| 04:58.16 | carrar | you need to fix your digit map/dial plan on the phone |
| 04:58.29 | tomcontr3 | DigitTimeout? |
| 04:58.51 | carrar | phones are set (and changable) to assume a dial command with a givien set of digits |
| 04:59.24 | carrar | You said before entering the context and when pressing the buttons |
| 04:59.38 | carrar | or do mean in a menu like setting? |
| 04:59.44 | carrar | ^you |
| 04:59.50 | tomcontr3 | right |
| 05:00.02 | carrar | yeah look at context timeouts |
| 05:00.06 | tomcontr3 | fixed, I used the DIgitTImeout |
| 05:00.13 | tomcontr3 | thanks |
| 05:00.36 | carrar | TIMEOUT(digit) |
| 05:00.40 | carrar | TIMEOUT(response) |
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| 05:04.10 | tomcontr3 | exactly |
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| 05:34.58 | obnauticus | can anyone here help me, i'm on freebsd and trying to install zaptel...and it's not working out too well...http://pastebin.ca/524857 |
| 05:35.00 | obnauticus | tried googling |
| 05:39.21 | carrar | Try installing linux |
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| 05:40.15 | carrar | AsahI! |
| 05:40.58 | carrar | http://pics.osburn.com/photo/34931/original |
| 05:45.49 | *** join/#asterisk dijungal (n=chatzill@208.0.231.92) |
| 05:46.21 | dijungal | hello can someone give me an example of the CLI Originate command |
| 05:46.33 | dijungal | i know the syntax but i need an example to better undersand it |
| 05:47.06 | dijungal | all the things i try don't seem towork |
| 05:47.22 | dijungal | originate SIP/phone phone |
| 05:47.25 | dijungal | dont work |
| 05:50.37 | dijungal | originate SIP/pstn-line 125 |
| 05:50.43 | dijungal | why does that not work? |
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| 05:55.01 | flenders | dijungal: where did you get that syntax? |
| 05:57.02 | dijungal | I made up that line using the syntax "Usage2: originate <tech/data> extension [exten@][context]" |
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| 05:58.26 | mjmarrio | can someone tell me if 1.4 still needs to set MYSQL_FRIENDS option in the Makefile of the channels directory? |
| 05:59.07 | mjmarrio | this is to support iaxfriends via MySQL |
| 05:59.32 | JT | dijungal: were are you using this origination syntax? |
| 05:59.46 | dijungal | asterisk CLI |
| 05:59.51 | JT | err |
| 05:59.58 | carrar | Usage1: originate <tech/data> application <appname> [appdata] |
| 05:59.59 | JT | originate only works in the AMI |
| 06:00.01 | flenders | dijungal: use this "originate sip/05 extension 08@local" |
| 06:00.04 | JT | Asterisk Manager Interface |
| 06:00.09 | JT | not the cli |
| 06:00.18 | flenders | you missed the 'extension word' |
| 06:00.24 | flenders | 'extension' word |
| 06:00.30 | JT | and using the correct interface |
| 06:01.17 | mjmarrio | IAX2 / SIP peers in MySQL Database |
| 06:01.17 | mjmarrio | The SIP and IAX2 channels support definition of friends in a MySQL database. |
| 06:01.17 | mjmarrio | To enable this, you need to edit the Makefile in the channels directory of your source tree and enable MYSQL_FRIENDS. This enables database definition of both IAX2 and SIP friends. Make sure you have the MySQL development kit (libraries) installed before compilation. |
| 06:01.21 | dijungal | btut i'm trying to understand the syntax |
| 06:03.22 | dijungal | what does <tech/data> mean? |
| 06:03.30 | *** join/#asterisk friedrich| (n=friedric@e177251029.adsl.alicedsl.de) |
| 06:03.32 | flenders | originate <tech/data> extension <extension@context> |
| 06:03.43 | flenders | SIP, Zap |
| 06:03.48 | flenders | technology |
| 06:04.04 | flenders | SIP/05 is a sip account on my asterisk |
| 06:04.25 | flenders | so, there's a [05] defined on my sip.conf file |
| 06:04.41 | dijungal | ok i have a 124 so i use SIP/124 |
| 06:04.51 | flenders | yeah |
| 06:05.51 | dijungal | ok great thats all i need it worked.. :) |
| 06:05.53 | dijungal | thanks much |
| 06:05.57 | flenders | no worries |
| 06:06.06 | JT | flenders: have you actually made it work in the cli? |
| 06:06.16 | dijungal | yea that tech/data thing was throwing me off... lol |
| 06:08.10 | flenders | jt: yeah |
| 06:08.40 | flenders | even this works from a shell: ~# asterisk -rx "originate sip/05 extension 08@local" |
| 06:09.03 | JT | i see |
| 06:09.06 | dijungal | works great.. now i can get it to send a call to call my extension and then send a call to an outside line and link the two |
| 06:09.07 | JT | is this new to 1.4? |
| 06:09.23 | flenders | JT: not sure |
| 06:09.37 | dijungal | originate SIP/zyxelphone extension {number}@SPA |
| 06:09.42 | flenders | JT: I remember asking here when I was still on 1.2.x, and no one knew how it worked |
| 06:09.57 | JT | i thought it was only for AMI |
| 06:10.11 | flenders | you have much more control over it using AMI |
| 06:10.20 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 06:10.26 | dijungal | <PROTECTED> |
| 06:11.11 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 06:11.12 | flenders | I don't think you can transfer a call using CLI, for example |
| 06:11.33 | *** part/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 06:11.47 | flenders | or maybe it does work |
| 06:11.51 | flenders | command is there |
| 06:13.13 | flenders | no, it doesn't work |
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| 06:44.40 | JT | Kumbang: good work |
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| 06:51.13 | infinity1 | how can i do this in asterisk AEL? It doesnt work! PHONES2=${EXT203} |
| 06:51.20 | infinity1 | you can't assign a variable with the contents of another variable! |
| 06:52.29 | Kumbang | JT: im sorry, it was an accident :) |
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| 06:54.51 | JT | :) |
| 06:55.24 | mjmarrio | I have untar'd 1.4 but cannot find the MYSQL_FRIENDS in Makefile of channels dir. Anyone know if it is still done this way? |
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| 07:09.02 | stimpie | I need to buy a gateway system for pstn <-> voip with 4 E1's anybody knows how much cpu is required for this, voip codec will be G.728 (and others) |
| 07:10.12 | JT | asterisk does not support G.728 |
| 07:10.22 | JT | and i haven't heard of anyone using it for telephony |
| 07:10.34 | JT | there's probably good reason for this (outdated?) |
| 07:11.05 | stimpie | hmm our partner prefers G.728 |
| 07:11.20 | JT | your partner might be suffering from the effects of drugs |
| 07:11.26 | JT | are you sure you don't mean G.729? |
| 07:11.36 | stimpie | lets make a call |
| 07:12.07 | JT | i've heard a lot of different codecs that are used with VoIP, and I've never heard G.728 mentioned |
| 07:13.07 | stimpie | it should be 726 |
| 07:14.16 | JT | oh |
| 07:14.19 | JT | that's much better |
| 07:14.27 | JT | supported and pretty good audio qulity |
| 07:15.59 | JT | not quite as good as g.711, but close |
| 07:17.01 | stimpie | do you know how much cpu it uses? |
| 07:23.06 | JT | not too much |
| 07:23.16 | JT | you will still need a beefy machine for 240 concurrent calls |
| 07:23.45 | JT | xeon dual or quad core |
| 07:23.48 | JT | would be good |
| 07:24.00 | JT | but you're really going to just have to do some testing |
| 07:24.01 | JT | bbl |
| 07:24.08 | *** join/#asterisk kiscokid (n=Ron@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 07:24.49 | kiscokid | I'm trying to build zaptel. Can anyone tell me how to use the patch command? |
| 07:31.59 | *** join/#asterisk sergee (i=kvirc@195.94.224.197) |
| 07:31.59 | sergee | are there any cisco AS53XX/54XX users here? :) |
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| 07:41.28 | tzafrir | kiscokid, from inside the zaptel command, patch <pathc_file -p1 --dry-run |
| 07:41.45 | tzafrir | if you get no error from the dry-run, leave out that switch |
| 07:42.09 | tzafrir | you may need to use -p0 instead of -p1 |
| 07:42.29 | tzafrir | Patches made by 'svn diff' are typically applied with -p0 |
| 07:43.35 | *** join/#asterisk TBW (n=s@MAILOUT.THEBIGWORD.COM) |
| 07:44.09 | kiscokid | tzafir if you have a directory zaptel-1.4.2.1 and a patch file zaptel-1.4.2.1-patch are you supposed to apply the patch before building? |
| 07:46.42 | tzafrir | what patch is that exactly for? |
| 07:46.47 | tzafrir | for a specific device? |
| 07:46.58 | tzafrir | or is it from asterisk.org as well? |
| 07:47.41 | kiscokid | don't know for sure but I had problems building so I thought I was supposed to apply the patch but I don't know for sure. |
| 07:47.53 | kiscokid | It is in ftp.digium.com |
| 07:48.05 | stimpie | my asterisk process is using 100% cpu while all calls have ended, is there any way too say which thread eats the cycles? |
| 07:48.09 | *** join/#asterisk angryuser (n=aster@df01t2-212-195-206-189.d4.club-internet.fr) |
| 07:48.12 | kiscokid | zaptel directory |
| 07:48.33 | tzafrir | zaptel-1.4.2.1-patch from asterisk.org is the diff from zaptel-1.4.2 (or from 1.4.1?). It is only useful if you have one zaptel directory and don't want to keep reloading the complete tarball. If you already downloaded the complete tarball: ignore it |
| 07:49.00 | kiscokid | ok, thanks |
| 07:49.01 | *** join/#asterisk canberk (n=canberk@teknopet.com) |
| 07:49.03 | tzafrir | ps -eLf |
| 07:49.16 | tzafrir | will give you a pid of the specific thread. |
| 07:49.28 | tzafrir | try stracing it. Maybe it will give you an idea |
| 07:49.54 | tzafrir | strace -p PID |
| 07:50.35 | tzafrir | stimpie, --^ |
| 07:51.02 | stimpie | strace output: ioctl(0, SNDCTL_TMR_STOP or TCSETSW, {B38400 opost isig icanon echo ...}) = -1 EIO (Input/output error) |
| 07:51.02 | stimpie | write(1, "*CLI> ", 6) = -1 EIO (Input/output error) |
| 07:51.02 | mjmarrio | can anyone tell me if iaxfriends is upported in 1.4 |
| 07:51.36 | tzafrir | interesting. Even more so if you can reproduce it. What version of Asterisk? |
| 07:51.58 | mjmarrio | I am pulling my hair out here trying to get iax users into a database |
| 07:52.21 | tzafrir | what is the main PID of asterisk? |
| 07:52.31 | stimpie | asterisk version 1.4.2 |
| 07:52.33 | tzafrir | ls -l /proc/PID_OF_ASTERISK/fd/1 |
| 07:53.21 | tzafrir | from what terminal did you run asterisk? do you use safe_asterisk or a similar wrapper script? |
| 07:54.48 | stimpie | asterisk was started in verbose mode and that terminal is killed |
| 07:54.55 | stimpie | that might be the problem |
| 07:55.31 | stimpie | lrwx------ 1 root root 64 May 31 09:53 /proc/32403/fd/1 -> /dev/pts/7\ (deleted) |
| 07:56.20 | stimpie | thanks, I think we found the cause |
| 07:56.27 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 07:57.35 | mjmarrio | can anyone assist me? |
| 07:57.57 | *** join/#asterisk drrt (n=junior@ip242-64.baltnet.ru) |
| 07:58.16 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 07:58.33 | kiscokid | trying to build zaptel-1.4.2.1 but getting errors on line 134 of torisa.c |
| 07:59.55 | tzafrir | Do you have any ISA card? |
| 08:00.06 | tzafrir | anyway, what wrror do you get? |
| 08:00.11 | tzafrir | What OS is it? |
| 08:00.32 | tzafrir | (what distro? what kernel) |
| 08:00.57 | kiscokid | <PROTECTED> |
| 08:01.08 | kiscokid | no ISA cards |
| 08:01.40 | tzafrir | well, you can skip that module, but you'll get exactly the same error in zaptel-base.c, IIRC |
| 08:02.02 | kiscokid | yeah, so how do I build it |
| 08:02.16 | tzafrir | I was sure that this error has been solved in centos. But maybe it has been solved on the -42 kernels |
| 08:02.26 | tzafrir | ~centosbug |
| 08:02.40 | jbot | extra, extra, read all about it, centosbug is a problem with the 2.6.9-42 kernels prior to 2.6.9-42.0.1. If you can't compile zaptel, do a 'yum update', you're running an old kernel. If you HAVE to run an old kernel, the fix is "sed -i s/rw_lock/rwlock/ /usr/src/kernels/*/include/linux/spinlock.h" |
| 08:02.48 | kiscokid | what is ~centosbug? |
| 08:03.08 | tzafrir | a factoid that jbot remembers. Here's another one: |
| 08:03.11 | tzafrir | ~jbot |
| 08:03.21 | jbot | jbot is probably a hack! |
| 08:04.04 | kiscokid | so if I do a yum update it will fix the kernel? |
| 08:04.41 | tzafrir | unless you use trixbox. they have disabled kernel updates |
| 08:04.57 | kiscokid | nope, not trixbox |
| 08:05.00 | tzafrir | Also be aware that for a kernel update to actually have any effect you need to boot |
| 08:05.13 | kiscokid | no prob |
| 08:05.32 | tzafrir | yeah. An update is generally a good idea |
| 08:05.49 | kiscokid | (perhaps a silly question) does an update ever break things? |
| 08:07.02 | *** join/#asterisk andyd (n=andyd@host90-152-23-30.ipv4.regusnet.com) |
| 08:07.42 | tzafrir | I can't really say never. But it generally helps much more than breaks. The problem generally is that it may not be totally bug-to-bug compliant and hence break working systems that relied on existing bugs. Or introduce new bugs. But in your situation, where you don't really have a working system that shouldn't break |
| 08:08.05 | kiscokid | ok, thanks |
| 08:08.31 | tzafrir | And the rate of changes in stable systems such as centos is not really high, so problems in the updates get spotted fast enough |
| 08:09.12 | kiscokid | where would the centosbug be listed before it gets to jbot? |
| 08:09.34 | tzafrir | you just ask jbot |
| 08:09.51 | tzafrir | Anything you think is useful to be a factoid? |
| 08:10.40 | kiscokid | not yet |
| 08:11.04 | kiscokid | but is there a repository of centosbugs outside of irc? |
| 08:11.04 | tzafrir | Basically the syntax is: jbot, ITEM is blah blah blah |
| 08:11.19 | tzafrir | http://bugzilla.redhat.com |
| 08:11.31 | tzafrir | And I think centos has a list of their own bugs |
| 08:11.38 | tzafrir | not to mention: |
| 08:11.40 | tzafrir | ~mantis |
| 08:11.59 | jbot | mantis is, like, at http://bugs.digium.com |
| 08:11.59 | tzafrir | ~bugs |
| 08:12.34 | kiscokid | ok |
| 08:13.31 | Uatec | hi, i'm looking in to buying a load of cisco 7912s, but looking on the voip-info wiki, it says that they come configured with skinny as default |
| 08:13.52 | Uatec | yesterday i thinki heard tell that to get them to use sip you have to update the firmware |
| 08:14.21 | Uatec | is this the case? do they work with SIP out of the box, or do you have to fiddle with them? if so, what kind of things do you have to do? |
| 08:19.01 | *** join/#asterisk qdk (n=qdk@213.150.62.32) |
| 08:19.10 | tzafrir | From what I know you have to ask them specifically with SIP firmware. Or buy SIP firmware separately from your distributor |
| 08:19.31 | tzafrir | Not exactly sure |
| 08:20.12 | tzafrir | (you can use chan_skinny / chan_sccp , but I guess that it's more of a pain) |
| 08:22.31 | angryuser | good day |
| 08:23.25 | Uatec | lol |
| 08:23.30 | Uatec | i just told my boss that |
| 08:23.38 | Uatec | and he pretty much said "oookay, not getting those then" |
| 08:23.39 | Uatec | phew |
| 08:23.43 | Uatec | i cba with the work of flashing them |
| 08:25.52 | angryuser | i have a question is is possible group misdn ports/channels, for example port=1,2 =>group=1 and do Dial(misdn/g1/${EXTEN}) ?? |
| 08:26.23 | matsk | Can someone that runs SVN test patch for app_minivm in bug http://bugs.digium.com/view.php?id=9765 to confirm that it solves the bug ? |
| 08:27.38 | matsk | any SVN'er out there ? |
| 08:34.26 | Uatec | i have exten => s,n,Dial(mISDN/g:isdn/${EXTEN:1}) |
| 08:34.35 | Uatec | so i wonder what g:isdn does... |
| 08:35.40 | Uatec | ah |
| 08:35.42 | Uatec | i have: |
| 08:35.45 | Uatec | [isdn] |
| 08:35.45 | Uatec | ports=1,2,3,4 |
| 08:35.45 | Uatec | ; at the moment |
| 08:35.45 | Uatec | context=external |
| 08:35.45 | Uatec | msns=* |
| 08:36.08 | Uatec | i guess you could put another [group1] ports=1,2 and [group2] ports=3,4 |
| 08:40.57 | angryuser | <Uatec> thankyou |
| 08:44.29 | angryuser | <Uatec> what kind f card do you use for isdn? |
| 08:44.51 | angryuser | <Uatec> i have b410p, and still unable to get it working |
| 08:46.08 | *** join/#asterisk Comete (n=lefieux-@krypton.saint-lo.fr) |
| 08:46.28 | Comete | hi |
| 08:49.15 | Comete | i wonder how to present 2 different call numbers for outgoing calls ? |
| 08:50.36 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 08:50.46 | Comete | for example, i have internal-user1 and internal-user2 who have 2 different external call numbers |
| 08:51.00 | angryuser | <Comete> and? |
| 08:51.46 | angryuser | <Comete> you have 2 account for them? (sip provider, whatever) |
| 08:51.47 | Comete | if internal-user1 call an external number, i would like its number to be presented |
| 08:51.49 | Uatec | yeah, we have a b410p |
| 08:51.58 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 08:51.59 | Uatec | angryuser, what kind ISDN do you use? |
| 08:52.23 | angryuser | <Uatec> b410p bri, i have a strange port's problems |
| 08:52.33 | Comete | same thing for internal-user2 |
| 08:52.34 | angryuser | http://bugs.digium.com/view.php?id=9400 i wrote it here |
| 08:53.13 | Uatec | angryuser, which versin of asterisk are you using? |
| 08:53.14 | angryuser | <Comete> eys call by what? sip isdn zap? |
| 08:53.36 | Comete | angryuser: sip |
| 08:53.43 | angryuser | <Uatec> i have tryed 1.4.0 same thisn with 1.4.4 |
| 08:53.50 | angryuser | *thing |
| 08:53.57 | Uatec | oh bri |
| 08:53.59 | Uatec | ok |
| 08:54.05 | Uatec | well i'm using business edition |
| 08:54.11 | Uatec | and i did have some trouble getting it to work |
| 08:54.49 | angryuser | my port's go down after first call, and never go up:) |
| 08:54.54 | Uatec | with help from digium support i put together a document about how to get it working, it's actually a guide to a complete install with the b410p of business edition, but i'm sure you could understand which bits were relevant |
| 08:55.03 | angryuser | so 4 calls => no calls |
| 08:55.11 | Uatec | i don't know if it would help you, but you could compare it to what yo have done |
| 08:55.29 | Uatec | on the other hand, it might be completely irrelevant since you're using 1.4 not 1.3 |
| 08:55.32 | angryuser | yes, i would like to read it |
| 08:55.57 | angryuser | <Uatec> where is is? |
| 08:56.12 | kiscokid | tzafrir: the yum update fixed my problem |
| 08:56.16 | Uatec | it's not published yet |
| 08:56.26 | Uatec | and i don't think it will be, because it's for internal use only, really |
| 08:56.31 | angryuser | <Comete> your provider normally shoul you external number automaticly |
| 08:56.37 | Uatec | and i gave a copy of digium |
| 08:56.40 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 08:56.43 | tzafrir | kiscokid, good |
| 08:56.47 | Uatec | and they're integrating the fix in to the next version of the business edition |
| 08:56.51 | angryuser | <Uatec> can you send it to me? |
| 08:57.06 | Comete | angryuser: in fact the server and the router talk to each other in ISDN but the router use ADSL |
| 08:57.08 | kiscokid | tzafrir thanks for your help |
| 08:57.14 | *** join/#asterisk frenzy (n=frenzy@unaffiliated/frenzy) |
| 08:58.09 | Uatec | angryuser, sending it via DCC now |
| 08:58.40 | angryuser | <Comete> i am not sure if you can push your ID umber to you provider hen you call out |
| 08:59.05 | Comete | ok |
| 08:59.30 | Comete | angryuser: i can push one in misdn.conf but two... |
| 08:59.52 | Uatec | Comete, you can use the SetCIDName() application, i think |
| 08:59.55 | Comete | with callerid=XXXXXXX |
| 09:00.10 | Comete | Uatec: ah ? |
| 09:00.11 | Uatec | but depending on your phone company, it's behavior will vary |
| 09:00.30 | Uatec | for example, i can set it to any of the numbers which i do own with this provider |
| 09:00.43 | Uatec | but i canot set it to one of the numbers which i have with another provider |
| 09:01.10 | Uatec | if it still does not work, even though you know that you're doing the right thing, call your phone provider |
| 09:01.28 | Uatec | and assuming they're not NTL they'll know what you're talking about and tell you if you can or cannot do this somehow, |
| 09:01.34 | angryuser | <Uatec> i have messaged my email to you, if you could send it please |
| 09:04.11 | Comete | Uatec: ok thanks |
| 09:04.27 | *** join/#asterisk Delvar (n=Delvar@host-83-146-53-46.bulldogdsl.com) |
| 09:05.12 | *** join/#asterisk mechanix[au] (n=mechanix@mechanix.riscom.net) |
| 09:06.32 | Uatec | np, hope it helps |
| 09:06.59 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 09:11.44 | angryuser | <Uatec> can i user normal ethernet cables for isdn? |
| 09:12.33 | angryuser | i cant find any data about isdn cables:) |
| 09:13.26 | Uatec | yup |
| 09:13.40 | Uatec | 5e and greater, so i hear |
| 09:14.23 | angryuser | i will try to install * 1.2 and do trunking if b410p works with it |
| 09:14.51 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 09:19.38 | *** join/#asterisk Mahmoud (n=fake@unaffiliated/mahmoud) |
| 09:19.41 | Mahmoud | sup guys |
| 09:21.41 | *** join/#asterisk denke (n=denke@mehess.adsl.datanet.hu) |
| 09:24.12 | Comete | Uatec: can i do something like: exten => _1NXXNXXXXXX/2100,1,Set(CALLERID(1010)=Joe Smith <2095551212>) ? |
| 09:24.50 | Mahmoud | how to dial someone and keep a counter in seconds? |
| 09:25.07 | Mahmoud | i want to disconnect a phone call just right after it being answered by 5 seconds |
| 09:25.44 | denke | Mahmoud: read the Dial () documentation |
| 09:25.55 | Mahmoud | i read.. |
| 09:26.05 | Mahmoud | i came up with dial(fooo stuff | S5); |
| 09:26.09 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 09:26.36 | Mahmoud | Dial(Technology/resource[&Tech2/resource2...][|timeout][|options][|URL]): |
| 09:28.15 | denke | S(x) - Hang up the call after 'x' seconds *after* the called party has |
| 09:28.16 | denke | <PROTECTED> |
| 09:28.20 | denke | sry |
| 09:28.33 | Mahmoud | yeah.. |
| 09:28.38 | Mahmoud | exten => 999,1,dial(${pots}/#33*4646# | S(10)) |
| 09:28.45 | Mahmoud | htis is my current command, but doesn't work |
| 09:29.19 | denke | <Mahmoud> Dial(what|how long|S5) |
| 09:29.26 | denke | something like this |
| 09:30.00 | denke | is wou do not want to tell how long to ring, then Dial(what||S5) |
| 09:30.10 | Mahmoud | ahhhhhh |
| 09:30.28 | Mahmoud | I think ew can use "," instead of | right? |
| 09:31.29 | Mahmoud | s/ew/we/ |
| 09:31.29 | denke | yes |
| 09:31.29 | denke | they are equal |
| 09:31.29 | Mahmoud | but , is depreciated or something? |
| 09:31.29 | Mahmoud | i saw it in the PDF document on asterisk's website, which had lots of depreciated commands |
| 09:31.31 | denke | no, i do not know about it, they are fully equal |
| 09:31.33 | denke | like? |
| 09:31.46 | denke | asterisk TFOT? |
| 09:31.50 | Mahmoud | yeah |
| 09:32.01 | Mahmoud | some commands changed since then, but not updated there |
| 09:32.15 | Mahmoud | new ones are: core show application foo |
| 09:32.30 | Mahmoud | older ones don't have 'core' |
| 09:32.42 | Mahmoud | it works but prints a warning message "hey fag, use the new one" |
| 09:32.57 | denke | yes it's true, guys at develpoer section changet this syntax... |
| 09:33.35 | matsk | ~book |
| 09:33.49 | jbot | it has been said that book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 09:34.06 | Mahmoud | the past of telephony :P |
| 09:34.16 | denke | it is in that too :) |
| 09:34.35 | Mahmoud | asterisk is a nice app.. i liked it |
| 09:34.54 | Mahmoud | but not perfect, which is not expected either |
| 09:35.44 | matsk | I was just looking for the linkt to the TFOT, it wa not related to your discussion ;-) |
| 09:36.18 | denke | I like it too, it is not perfect, true, but I had to learn a lot about No7, PDH, SDH, and so on..... |
| 09:36.53 | denke | old telephone centers such az CrossBar, and Rotary.... |
| 09:38.33 | denke | in PBX es I have worked with Alcatel, Panasonic Lucient, Siemens .... and not one of them can be mentioned in the same paper as asterisk.... |
| 09:38.54 | denke | if it would be bugless.... it would be perfect |
| 09:42.01 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 09:42.51 | *** join/#asterisk DoDaT69 (n=dodat69@66-240-110-115.atl.fdn.com) |
| 09:43.41 | *** join/#asterisk santibiotico (n=santi@ip23498.bcn.altecom.net) |
| 09:43.43 | santibiotico | hi |
| 09:44.01 | santibiotico | i'm trying to add some extra info in the cdr logs |
| 09:44.15 | santibiotico | i've added the extra fields at cdr_custom.conf |
| 09:44.33 | santibiotico | and in the dialplan i use Set(CDR(newfield)=value) |
| 09:44.47 | santibiotico | but i don't get the info in the cdr file |
| 09:44.56 | santibiotico | do i need to do anything else? |
| 09:45.30 | *** part/#asterisk kiscokid (n=Ron@adsl-216-101-109-187.dsl.snfc21.pacbell.net) |
| 09:45.48 | tzafrir | ~thebook |
| 09:46.07 | jbot | i heard thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 09:46.38 | tzafrir | matsk, in case you didn't find that link ---^ |
| 09:46.43 | *** join/#asterisk mkl1525 (n=qwertz@pD9530A78.dip0.t-ipconnect.de) |
| 09:47.43 | mkl1525 | Hi, I've got the problem, that callers stay in queue for ever although there's a time limit (even the wiki example didn't work). So does the queue need some zaptel timing device to get the time? |
| 09:47.49 | tzafrir | denke, if it would be bugless it wouldn't be software. Unless it is mutt |
| 09:49.39 | denke | tzafrir: true, but to tell the trouth Im afraid when I have to give out a production system from my hand..... |
| 09:50.20 | denke | tzafrir That's true that safe_asterisk is exists, but even then... |
| 09:50.44 | tzafrir | Don't count on safe_asterisk. It hides some other types of problems, and it can loop without asterisk running in some cases |
| 09:51.45 | *** join/#asterisk cy_` (n=dont@office.adfinis.com) |
| 09:52.02 | cy_` | yo |
| 09:52.03 | tzafrir | denke, and what makes you think other systems don't have bugs? |
| 09:52.31 | cy_` | what do i need to get started with asterisk ? |
| 09:53.09 | denke | I do not think that, but this week i have reported 2 bugs and found a 3. which coul easyly take the whole system down... |
| 09:54.20 | tzafrir | If the bug is a show-stopper, maybe it is worth rebooting the system sometime late at night. |
| 09:54.30 | denke | tzafrir: ...good to know about that... (safe) I think then i will have to read it's source and change a few things .... or write another |
| 09:54.43 | tzafrir | If it is not a show-stopper, let it wait until a show-stopper bug comes |
| 09:55.32 | denke | one of them leads to crash, and two leads to overload the system... |
| 09:55.42 | denke | both of them is in the ARA. |
| 09:56.17 | tzafrir | Sounds like show-stoppers |
| 09:58.20 | denke | all of them is 100% reproduceable, so I do not win anything by restarting asterisk.... I have to make the system sork othervise... by not useing yhe bugous parts |
| 09:58.55 | denke | tzafrir: My big problem is, that i'm a starter in 'C', so i can not repair that bugs ... yet |
| 10:00.13 | santibiotico | is there any way to write src field in cdr's? |
| 10:00.42 | santibiotico | if i use Set(CDR(src)=value) i obviously get a read-only error |
| 10:00.55 | santibiotico | is there any way to by-pass this permission issue? |
| 10:01.33 | denke | santibiotico: I think only by hacking the source, or use the user field |
| 10:01.51 | santibiotico | and how do i use the userfield? |
| 10:02.08 | santibiotico | if i use Set(CDR(UserField)=value) i don't get any output in the cdr file |
| 10:02.38 | denke | My system works a bit else ... so I need a moment |
| 10:03.03 | threat | hey |
| 10:03.35 | *** join/#asterisk e-milio (n=emilio@adsl-9-191-156.mia.bellsouth.net) |
| 10:03.56 | e-milio | hello there |
| 10:04.49 | e-milio | is there anybody awake with that can help setup a pri ? |
| 10:04.57 | threat | sorry |
| 10:05.01 | JT | yes, maybe |
| 10:05.05 | threat | I am a nooblet |
| 10:05.38 | e-milio | I have actually 1 t1 (from telco) and a PRI that goes to a NEC pbx |
| 10:05.47 | santibiotico | any way to set the CDR(userfield) |
| 10:05.48 | santibiotico | ? |
| 10:06.05 | JT | the t1 from the telco is in pri mode? |
| 10:06.05 | denke | santibiotico: SetCalledParty, SetCallerid SetCDRUserfield |
| 10:06.17 | santibiotico | denke these are deprecated functions |
| 10:06.22 | e-milio | JT: no, clear channel T1 22 channles, 23,24 are data |
| 10:06.41 | JT | CAS? |
| 10:06.50 | e-milio | and the telco provides clock |
| 10:06.56 | JT | yes |
| 10:07.18 | JT | are those 22 channels in CAS/RBS mode? |
| 10:07.32 | denke | santibiotico I use the latest snapshot, and i can see theese options... |
| 10:08.09 | e-milio | mmm not sure. I just conf it with e&m and it is working now |
| 10:08.21 | e-milio | the questions is actually on the PRI that goes to a NEC pbx |
| 10:08.33 | JT | maybe it's meant to be e&m mode then :) |
| 10:08.50 | e-milio | hopefully... |
| 10:08.59 | santibiotico | denke: anyway, if i check the cdr files there is no userfield data |
| 10:09.21 | e-milio | my question is if on the First T1 the clock is handled by the telco |
| 10:09.49 | e-milio | and the nec PBX has to be slave, do I have to be the clock for this ? |
| 10:10.00 | JT | yes' |
| 10:10.03 | denke | | userfield | varchar(255) | YES | | NULL | | |
| 10:10.08 | JT | clock must come from somewhere |
| 10:10.09 | e-milio | or by beeing slave on the first T1 it is just ok. ? |
| 10:10.12 | denke | grr, sry |
| 10:10.28 | JT | no, it must provide clock to the nec if the nec expects to be slave |
| 10:10.46 | denke | santibiotico I work with a database, and i do have a field named userfield... |
| 10:11.02 | denke | santibiotico I do not know if it helps you |
| 10:11.15 | e-milio | JT: even if you havce other span that you is already providing you with clock? |
| 10:11.49 | JT | yes, you accept clock as slave on that span, and give it on the span to the nec |
| 10:11.57 | JT | i fail to see how that isn't simple to understand :) |
| 10:12.06 | JT | no clock = no work |
| 10:12.58 | e-milio | no clock =no work (on each span or on each system: this was my confusion) |
| 10:13.24 | JT | each port is a digital serial connection |
| 10:13.30 | JT | consider them independantly |
| 10:13.38 | JT | they must have one side providing clock on each |
| 10:14.09 | denke | Anybody knows how to contact the current developers of the chan_sccp driver? |
| 10:14.39 | e-milio | JT: thanks a lot, it is clear now... Not working but clear |
| 10:14.44 | JT | :) |
| 10:15.23 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
| 10:18.09 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 10:18.46 | *** join/#asterisk alin` (n=user@193.226.173.50) |
| 10:19.43 | alin` | can somebody tell me how strong a PC must be in order that asterisk work with 30 calls? |
| 10:19.59 | alin` | 30 calls in the same time |
| 10:20.57 | JT | doing what? |
| 10:21.00 | JT | transcoding? |
| 10:21.42 | denke | and not to mention, what codec? :) |
| 10:23.55 | alin` | doing calls. |
| 10:24.13 | alin` | yes, calls require transcoding |
| 10:24.14 | JT | alin`: no, you will need to give much more details or we cannot answer your question |
| 10:24.22 | *** join/#asterisk mjmarrio (n=mike@210.19.201.38) |
| 10:24.23 | JT | alin`: to and from what codecs |
| 10:24.24 | JT | ? |
| 10:24.52 | mjmarrio | can anyone help me with MySQL problem? |
| 10:25.04 | alin` | JT: doing calls between 30 agents with 30 outside callers. |
| 10:25.06 | denke | <mjmarrio> maybe |
| 10:25.15 | denke | <mjmarrio> How can I help? |
| 10:25.35 | alin` | JT: the codecs arbitrary |
| 10:25.40 | mjmarrio | I am trying to get iaxfriends table working with Asterisk 1.4 |
| 10:25.44 | *** join/#asterisk manopulus (n=manopulu@213.197.181.154) |
| 10:25.47 | JT | alin`: it is not, please specify |
| 10:26.06 | JT | if you don't know, just say so, instead of being evasive |
| 10:27.00 | alin` | I want to know the minimum power of a system that keeps 30 calls with transcodecs between ARBITRARY codecs. |
| 10:27.14 | mjmarrio | denke: how much do you know about mysql and asterisk v1.4? |
| 10:27.32 | alin` | somebody asks me this question and he insists to have an answer... |
| 10:27.39 | denke | I'm working like this |
| 10:27.41 | alin` | I leave a moment |
| 10:27.45 | mjmarrio | I would appreciate any thoughts at all |
| 10:28.07 | JT | alin`: let's assume the codec is the most cpu hungry then, G.729 or ilbc |
| 10:28.27 | JT | alin`: a P4 or equivalent generation xeon would by my guess at a minimum |
| 10:28.30 | denke | mjmarrio with compileing, or using? cdr, or res_conf? |
| 10:28.42 | manopulus | tzafrir: hello, have a free minute? |
| 10:28.58 | mjmarrio | denke have you defined iax users in your database? |
| 10:29.35 | mjmarrio | denke: originally we used AsteriskNow but as I understand it is based on 1.4 |
| 10:29.54 | alin` | JT: yes, we can suppose that all 30 calls must translate from the most cpu hungry |
| 10:29.56 | *** join/#asterisk porche (n=porche@81.215.215.76) |
| 10:30.01 | porche | hi all |
| 10:30.06 | alin` | than what computer do I need? |
| 10:30.32 | porche | got a question about analog lines/tdm2400/hang up detection |
| 10:30.47 | denke | mjmarrio I created the database, but I did not use |
| 10:30.51 | porche | can any1 help? |
| 10:30.58 | mjmarrio | I got CDR to work ok but doesn't seem to recognise config in iax.conf |
| 10:31.00 | alin` | JT: I do not think that a P4 keeps all 30 calls |
| 10:31.01 | JT | alin`: read up, i've already answered |
| 10:31.18 | JT | alin`: have you tried? |
| 10:31.29 | denke | mjmarrio static, or dynamic? |
| 10:31.31 | alin` | JT: maybe with very much memory... |
| 10:31.35 | mjmarrio | dynamic |
| 10:31.44 | alin` | I did not tried, because I cannot |
| 10:31.53 | denke | did you map it in extconfig.conf? |
| 10:31.54 | alin` | that is why I ask... |
| 10:32.07 | JT | alin`: the operation is cpu dependant, not memory |
| 10:32.15 | JT | at least with standard modern quantities of ram |
| 10:32.16 | mjmarrio | I know that in earlier version I had to enable MYSQL_FRIENDS in Makefile but I can't seem to find information in 1.4 |
| 10:32.33 | alin` | JT: ok, thanks a lot |
| 10:32.58 | porche | JT, may I also ask? |
| 10:33.03 | denke | mjmarrio not needed, just map the dynamic configuration at extconfig.conf |
| 10:33.09 | mjmarrio | denke: also I am confused about ARA and whether it is relevant to iaxfriends or not |
| 10:33.11 | JT | porche: depends on the question |
| 10:33.26 | alin` | JT: you say that the memory is not important for this? |
| 10:33.30 | alin` | just the CPU? |
| 10:33.42 | JT | alin`: not if you have 512 or more |
| 10:33.54 | JT | unless you're doing something very complicated, software wise |
| 10:33.54 | mjmarrio | denke: ok I will have a look tq |
| 10:33.55 | porche | JT, question is about analog lines connected to a TDM2400 digium card, call comes in, asterisk answers, but cannot detect hang up |
| 10:34.04 | JT | more ram would also help if you are recording all 30 channels |
| 10:34.07 | alin` | JT: ok, thanks |
| 10:34.13 | alin` | I have one more question |
| 10:34.18 | JT | porche: that's normal |
| 10:34.19 | denke | mjmarrio ARA dose the whole thing ;) |
| 10:34.26 | JT | porche: analogue POTS has terrible signalling |
| 10:34.37 | JT | very difficult for a computer to determine what a hang up is |
| 10:34.51 | alin` | but for digium , there are others producers of boards PCI that work with asterisk? |
| 10:34.52 | porche | JT, i see, i can see the busy signals on the phone line after hang up |
| 10:34.55 | JT | porche: you need to ask your telco to setup Polarity Reversal on Hangup |
| 10:35.12 | JT | alin`: sangoma is the main one |
| 10:35.24 | JT | there are a few others too, but much less used |
| 10:35.30 | mjmarrio | denke: So in the case of AsteriskNow I installed the addons for 1.4 is that the right thing to do? |
| 10:35.42 | porche | i see, busydetect is of no use then |
| 10:35.50 | JT | porche: indeed |
| 10:35.53 | manopulus | hello, for xorcom isdn gw, which connected to asterisk via usb, what drivers shall I use? bristuff? clean asterisk? some patches? |
| 10:36.04 | porche | got it JT, thanks alot for the info |
| 10:36.09 | denke | mjmarrio yes, and map the configs at extconfig.conf |
| 10:36.10 | JT | np |
| 10:36.32 | denke | mjmarrio what about a private chat? it is hard to follow like this |
| 10:36.33 | alin` | JT: thanks |
| 10:36.52 | mjmarrio | Oh that would be great ! |
| 10:36.58 | mjmarrio | how do I go there? |
| 10:38.18 | tzafrir | manopulus, bristuff. If you're using xpp older than r3814 (that is: zaptel 1.2.17.1, or zaptel 1.4.2.1, as opposed to our latest tarballs, or latest svn) you also need a little patch to bristuff's zaptel |
| 10:38.32 | tzafrir | what distro do you use? |
| 10:38.40 | manopulus | tzafrir: ubuntu |
| 10:38.56 | tzafrir | which ubuntu? |
| 10:39.59 | e-milio | JT: that this sound right : http://www.pastebin.ca/525385 for span1=t1 from telco and span2 master to a pbx ? |
| 10:40.05 | *** join/#asterisk eeos (n=eeos@86.53.50.16) |
| 10:40.09 | eeos | hi there |
| 10:40.24 | manopulus | tzafrir: 6.06 |
| 10:40.49 | tzafrir | hmm... the version of asterisk there is probably a bit old |
| 10:40.54 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 10:41.37 | tzafrir | eeos, e&m=1-22 ? why not 1-24? |
| 10:41.49 | tzafrir | Though I really don't know such systems |
| 10:42.00 | manopulus | tzafrir: thank you! |
| 10:42.01 | JT | e-milio: at a glance, yes |
| 10:42.09 | JT | tzafrir: he has data on 23 and 24 |
| 10:42.28 | e-milio | tzafrir: 23,24 data |
| 10:43.11 | tzafrir | manopulus, basically get the latest bristuff 0.3.0 |
| 10:43.24 | tzafrir | run just the ./download.sh |
| 10:43.55 | manopulus | I have asterisk here, bristuffed, latest... but not with xorcom board |
| 10:44.02 | e-milio | JT: if it says primary dchan down is a pbxissue on the other side? |
| 10:44.03 | tzafrir | then get http://updates.xorcom.com/astribank/src/zaptel-1.2.17.1.xpp.r3965.tar.gz |
| 10:44.08 | *** join/#asterisk Vide800 (n=vide@71.Red-80-34-171.staticIP.rima-tde.net) |
| 10:44.14 | Vide800 | hi |
| 10:45.02 | Vide800 | question: I have a sip channels that is "on hold", the client doesn't see any call but she cannot receive any other call |
| 10:45.04 | tzafrir | replace the zaptel symlink with a link to the directory extracted from there. And then rebuild. Or just rebuild and reinstall zaptel |
| 10:45.11 | JT | e-milio: possibly |
| 10:45.12 | Vide800 | sip show channels: |
| 10:45.13 | Vide800 | 192.168.1.227 701 2e1adfc3406 00102/00005 ulaw Yes Rx: BYE |
| 10:45.22 | Vide800 | how can I kill this channel? |
| 10:45.26 | Vide800 | soft hangup doesn't work |
| 10:45.35 | JT | e-milio: i assume you are of course connecting your T1 card to the NEC PBX with a T1 crossover cable, correct? |
| 10:45.46 | tzafrir | (the only changes should be in the xpp/ directory, basically, and minor changes in zaptel-base.c that we're pushing upstream) |
| 10:46.04 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 10:46.44 | Vide800 | any idea? |
| 10:46.53 | e-milio | JT: nope, regular cable :( |
| 10:46.55 | tzafrir | Vide800, soft hangup <tab> <tab> |
| 10:47.04 | tzafrir | ah, ok |
| 10:47.19 | Vide800 | tzafrir: in soft hangup <tab> that channel (SIP/701) is not shown |
| 10:47.21 | e-milio | JT: Is it imperative ? |
| 10:47.30 | JT | yes |
| 10:47.37 | JT | it will not work without one |
| 10:47.45 | *** join/#asterisk br4k3r (n=email@bas9-ottawa23-1242450783.dsl.bell.ca) |
| 10:47.45 | tzafrir | It should be SIP/701-something , not SIP/701 , right? |
| 10:47.48 | JT | physically impossible signal won't get through |
| 10:48.08 | Vide800 | tzafrir: yeah, it should be, but no SIP/701-* at all |
| 10:48.17 | tzafrir | show channels |
| 10:48.29 | tzafrir | maybe something else is connected to it? |
| 10:48.29 | e-milio | JT: thakns again, will do, unfortunatly later because no access to location... |
| 10:48.42 | Vide800 | it doesn't appear in show channels |
| 10:48.44 | *** join/#asterisk angryuser (n=aster@df01t2-212-195-206-189.d4.club-internet.fr) |
| 10:48.48 | Vide800 | only in in "sip show channels" |
| 10:49.01 | tzafrir | pardon my ignorance, but is the fact that 23 and 24 are data implicit in e&m signalling? |
| 10:49.29 | JT | e-milio: there are instructions on voip-info.org on making a T1 crossover cable |
| 10:49.51 | JT | tzafrir: not afaik, his telco just delivers him data unrelated to telephony over the same t1 |
| 10:50.05 | JT | e-milio: otherwise use a pre-terminated t1 crossover cable if you can find one |
| 10:50.32 | voipnet-tech | morning |
| 10:50.46 | voipnet-tech | brb |
| 10:54.19 | e-milio | JT: preterminated = buy one right ? |
| 10:55.19 | JT | e-milio: yes |
| 10:55.24 | JT | as opposed to making your own |
| 10:55.37 | e-milio | tzafrir: I really dont know. It is an XO t1 where they deliver 128k of internet access over the same T1 where they provide 22 channels of voice. |
| 10:55.43 | JT | but they're not that easy to find premade |
| 10:55.51 | JT | e-milio: so you may need to make your own |
| 10:55.59 | e-milio | JT: will do |
| 10:58.11 | JT | e-milio: in addition to voip-info, ref http://www.arcelect.com/RJ48C_and_RJ48S_8_position_jack_.htm "RJ-48C" |
| 10:58.22 | JT | that diagram is a little confusing though |
| 10:59.51 | e-milio | JT: thanks again. That helps |
| 11:00.24 | JT | tx goes to rx on the other wise, + to +, - to - |
| 11:00.29 | JT | s/wise/side/ |
| 11:10.00 | *** join/#asterisk stoffell (n=stoffell@fw.catsanddogs.com) |
| 11:11.54 | Winkie | hey guys, i'm having issues trying to detect channel status, ChanAvail seems to return available no matter what the state of the device |
| 11:12.17 | Winkie | i need to be able to check if a device is ringing / has a call on it already / is in DND before i actually dial |
| 11:12.21 | Winkie | is there a decent way to do this? |
| 11:12.41 | *** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl) |
| 11:16.48 | manopulus | Winkie: in general you may check SIP and IAX peer status |
| 11:16.50 | manopulus | with quality |
| 11:17.35 | manopulus | I do it just to check is remote host online, my dial macro checks is online, if not, wait for second or two, checks again, if not online, try (if exists) the fallback route (again online, wait, online and so on) |
| 11:17.54 | manopulus | ChanAvail is for ZAP |
| 11:18.43 | Winkie | manopulus: it supposedly works for SIP but i can't seem to make it work, detecting things like DND seems impossible though |
| 11:19.01 | Winkie | you would hope that there was a way to send a command to the phone to report it's status :( |
| 11:19.11 | manopulus | Winkie: partially, possible, by reading sip headers |
| 11:20.15 | manopulus | Winkie: you can keep last status, reported by last dial command, you can change rules to set DND or forward on phone with those features and so on... look by your situation |
| 11:20.19 | *** part/#asterisk porche (n=porche@81.215.215.76) |
| 11:22.26 | Winkie | hmm, chanisavail seems to detect just fine if they're online or not |
| 11:22.31 | Winkie | just not whether they're on a call or not |
| 11:22.47 | Winkie | i can deal with that using groups, but i doubt there's a specific return value for Dial that indicates DND? |
| 11:22.51 | Zeeek | isn't the result for DND the same as BUSY? |
| 11:23.00 | Winkie | bah i feared as much |
| 11:23.14 | Winkie | we have a need for some complex queuing |
| 11:23.36 | Winkie | it's hard to explain, but i think i have a decent idea of how to do this |
| 11:24.44 | *** join/#asterisk kclaussen (n=kclausse@204.13.224.242) |
| 11:25.54 | quitte | is there a mirror for asterisk somewhere? digium doesn'T work for me |
| 11:27.08 | puzzled | quitte: http://www.laimbock.com/asterisk/ has 1.2 sources at the end of the page |
| 11:27.24 | quitte | thanks |
| 11:27.27 | *** join/#asterisk saftsack (n=saftsack@pd9e055aa.dip.t-dialin.net) |
| 11:30.16 | *** join/#asterisk MrWup (i=root@80-46-38-124.static.dsl.as9105.com) |
| 11:30.18 | MrWup | hi |
| 11:30.32 | MrWup | with asterisk dialplan, how do i do an if statement? |
| 11:31.32 | MrWup | i dont want gotoif |
| 11:31.32 | MrWup | cause that makes things really difficult |
| 11:31.32 | tzafrir | bristuff uses some mirror |
| 11:31.32 | quitte | hmm. doesn't work either. maybe I have some vry weird problem wth my firewall |
| 11:31.32 | MrWup | i simply need a step which is like if something, do command |
| 11:31.32 | MrWup | and then it goes onto the next priority |
| 11:31.41 | tzafrir | http://www.asterisk-support.de/mirror/ |
| 11:32.48 | quitte | tzafrir: thanks. that works |
| 11:35.11 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 11:38.23 | eeos | Zeeek: http://phpfi.com/238355, getting on with your exercise. what do you think of t? |
| 11:41.22 | eeos | Zeeek: still, cannot understand how to set up our asterisk to connect to the sip line from our provider. |
| 11:42.40 | manopulus | hello, customer have snom 360. suddenly, his keys bacame blocked, snom works, receiving calls.... but they cannof place any call. question - are it is not sno "feature", maybe customer blocked keys someway? |
| 11:44.27 | Zeeek | eeos I'll take a look in a moment (actual work is calling) |
| 11:45.08 | *** join/#asterisk Tebi (n=rantis@gw.aller.fi) |
| 11:45.13 | Zeeek | eeos so can the users call each other? |
| 11:47.51 | *** join/#asterisk msetim (n=msetim@200.195.161.164) |
| 11:47.59 | *** join/#asterisk kippi (n=none@untrust-gct.equinoxit.net) |
| 11:48.00 | kippi | hey |
| 11:48.32 | kippi | using a linksys SPA942 and when I hangup, asterisk is keeping the calls open on the switch, anyideas? |
| 11:48.50 | Zeeek | manopulus one of the features mentioned in snom doc : " Keyboard lock" |
| 11:49.24 | Zeeek | kippi NAT or direct on same LAN as asterisk ? |
| 11:50.24 | eeos | Zeeek: at the moment thy can' because the test server is dow :( |
| 11:50.36 | eeos | s/dow/down |
| 11:50.44 | Zeeek | how unfortuante. But the dialplan works? |
| 11:51.14 | eeos | I will tell you i a few minutes |
| 11:51.19 | kippi | direct on same LAN ? |
| 11:51.23 | eeos | (as soon as it is back up) |
| 11:51.41 | Zeeek | kippi same subnet? |
| 11:52.24 | kippi | the phone is on a 10.54.1.x address and the call server is on 10.54.2.x |
| 11:52.44 | Zeeek | kippi is there a NAT router between them |
| 11:52.55 | kippi | yep |
| 11:53.20 | Zeeek | I have a 941 connected tot he asterisk box here and don't have that problem |
| 11:53.33 | Zeeek | does audio work both ways? |
| 11:54.00 | kippi | yeah, everything works fine, asterisk just seems to be keeping the call open |
| 11:54.33 | Zeeek | have you watched debug or at least teh CLI messages during the call? |
| 11:54.59 | *** join/#asterisk grEvenX (n=even@ti500720a080-7918.bb.online.no) |
| 11:55.21 | kippi | its not seeing my hangup |
| 11:57.10 | Zeeek | kippi that doesn't answer my last two questions |
| 11:58.54 | kippi | audio: yes both ways, how can I get the last message sent? the last one I see at the mo is Zap/1-1 answered SIP/1161-b470d640 |
| 12:02.41 | *** join/#asterisk porche (n=porche@81.215.215.76) |
| 12:02.47 | porche | hi there |
| 12:02.52 | porche | JT, still there? |
| 12:04.53 | JT | a bit |
| 12:04.56 | porche | :) |
| 12:05.00 | porche | talked to my telco |
| 12:05.10 | porche | they say no polarity reversal, talked about |
| 12:05.40 | porche | some CPB, PLO |
| 12:05.47 | porche | states/codes |
| 12:05.49 | porche | sent |
| 12:05.56 | porche | about call hang up |
| 12:06.59 | porche | the contact said, it's something about the timing of the detection |
| 12:07.06 | porche | i am really clueless |
| 12:10.56 | msetim | Hi |
| 12:19.47 | *** join/#asterisk isamar (n=isamar@202.95.221.162) |
| 12:19.52 | isamar | hi folks.... |
| 12:21.08 | isamar | anybody using chan_ooh323 .? |
| 12:21.18 | isamar | I am facing issues dialing from * to an ATA |
| 12:21.24 | isamar | http://bugs.digium.com/view.php?id=7794 |
| 12:21.37 | isamar | anybody has any workaround for that? |
| 12:23.23 | *** join/#asterisk Cyber-Dogg (i=Cyber-Do@24-178-240-97.dhcp.stls.mo.charter.com) |
| 12:23.34 | *** join/#asterisk mjmarrio (n=mike@210.19.201.38) |
| 12:23.51 | Zeeek | ZAP |
| 12:24.48 | msetim | what mean the _ in the begining extension dialplan |
| 12:25.05 | Zeeek | that it contains wildcard |
| 12:26.24 | msetim | thanks guys, I found the answer, is a match E.R |
| 12:26.59 | Zeeek | since you asked that question, you might want to read this: http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html |
| 12:27.37 | Zeeek | the text is a beginner level explanation of asterisk that answers about 80% of the questions asked here daily |
| 12:28.15 | Zeeek | ~barebones http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html |
| 12:31.31 | *** join/#asterisk WeBRainstorm (i=nobody@89-96-52-24.ip10.fastwebnet.it) |
| 12:34.52 | *** join/#asterisk mcab (n=mb@mostly-harmless.ca) |
| 12:35.53 | Uatec | ahhh |
| 12:35.59 | Uatec | i love big long pointeless interviews |
| 12:36.08 | WeBRainstorm | hi guys |
| 12:36.20 | WeBRainstorm | I have a problem with asterisk and sipsak |
| 12:36.32 | Uatec | 4 hours dedicated to upgrading one users phone from a single line to a multiline handset |
| 12:36.43 | WeBRainstorm | SIP packets generated |
| 12:36.52 | WeBRainstorm | have a wrong UDP checksum |
| 12:37.00 | WeBRainstorm | (so tells tethereal) |
| 12:37.11 | WeBRainstorm | so they're discarder from the router |
| 12:37.39 | angryuser | hm, i am searching ip hone with keyboard, i saw one, but dont remember it's name or brand, it had some futuristic design;) anyone? |
| 12:38.05 | angryuser | i do remember a keyboard |
| 12:38.30 | Zeeek | iPhone |
| 12:38.47 | isamar | <PROTECTED> |
| 12:38.52 | porche | http://www.voipsupply.com/index.php?cPath=95 |
| 12:38.57 | isamar | anybody using ooh323 ? |
| 12:39.06 | porche | angryuser, you are after an ip phone |
| 12:39.11 | porche | rather than adaptor |
| 12:39.42 | angryuser | yes, ip phone |
| 12:40.18 | porche | check out the link |
| 12:40.23 | porche | http://www.voipsupply.com/index.php?cPath=95 |
| 12:42.34 | angryuser | it was written full name call compatibility |
| 12:42.34 | porche | you can find other providers as well, try google |
| 12:42.34 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:54.14 | *** join/#asterisk coppice (n=chatzill@10.198.17.210.dyn.pacific.net.hk) |
| 12:54.35 | isamar | coppice... |
| 12:54.55 | coppice | hi |
| 12:54.57 | isamar | how is going? |
| 12:55.23 | *** join/#asterisk elg (n=fugalh@216.31.27.110) |
| 12:55.24 | isamar | I am friend of benjk of Tokyo.. |
| 12:55.48 | Zeeek | I ate there a few times |
| 12:56.13 | isamar | :-) |
| 12:56.30 | isamar | I need a hand with ooh323 |
| 12:56.43 | isamar | do you know if there is solution for this already? |
| 12:56.45 | isamar | http://bugs.digium.com/view.php?id=7794 |
| 12:56.55 | isamar | I couldn't find anywhere... :-( |
| 12:57.41 | tzafrir | isamar, that bug seems to be suspended |
| 12:57.54 | tzafrir | However what version of ooh323c do you use? |
| 12:58.04 | isamar | yeah.. I know.. but still there.. :-( |
| 12:58.08 | WeBRainstorm | guys, problem with Asterisk 1.2.x and UDP checksum incorrect |
| 12:58.16 | WeBRainstorm | problem with sipsak on the same hosts... |
| 12:58.28 | isamar | tzafrir: I am using asterisk-addons-1.4.1 |
| 12:58.30 | WeBRainstorm | caputured trafic with tethereal tells e checksum incorrect.... |
| 12:58.35 | isamar | with asterisk 1.4.1 |
| 13:00.09 | WeBRainstorm | any clue ? |
| 13:02.11 | isamar | why h323 in asterisk is ever unstable? |
| 13:03.17 | isamar | fucking hell! |
| 13:03.42 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 13:03.56 | *** join/#asterisk galeras (n=root@200.31.204.42) |
| 13:06.20 | *** join/#asterisk Skarmeth (n=Skarmeth@201009092213.user.veloxzone.com.br) |
| 13:06.49 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 13:07.21 | coppice | I guess some people just don't care |
| 13:08.00 | tzafrir | which h323 exactly? |
| 13:08.37 | JT | ircrly: because very few use it |
| 13:08.37 | tzafrir | Technically the "h323 in asterisk" is not the one he meant... |
| 13:09.11 | *** join/#asterisk bmg505 (n=leon@196.209.178.152) |
| 13:10.50 | coppice | JT: that is a self fulfilling argument |
| 13:11.15 | JT | ? |
| 13:11.33 | coppice | if it doesn't work, few people use it |
| 13:11.42 | JT | indeed |
| 13:11.45 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 13:11.48 | JT | and if digium doesn't care about it |
| 13:12.04 | JT | very little development will occur |
| 13:12.36 | [TK]D-Fender | My chicken ate your egg... |
| 13:12.58 | tzanger | [TK]D-Fender: those chickens tend to be the first in the pot |
| 13:12.59 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 13:12.59 | *** mode/#asterisk [+o mog] by ChanServ |
| 13:13.47 | [TK]D-Fender | tzanger: Screen is mounted, projector mount & 2nd screen (free extra @ 80") pending arrival. |
| 13:14.19 | tzanger | eh? |
| 13:14.32 | *** join/#asterisk Katty (n=Katty@hera.copi-rite.com) |
| 13:14.34 | Katty | mew. |
| 13:14.39 | creativx | hmm I wonder if I should write a module for asterisk that can interpret http://www.lolcode.com/ |
| 13:14.57 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:15.04 | [TK]D-Fender | tzanger: Was sure I told you.. I'm ditching my 52" rear projection TV for a projector & 120" screen :) |
| 13:15.57 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 13:16.52 | coppice | Is it a cricket ball? |
| 13:16.54 | coppice | Is it a tennis ball? |
| 13:16.55 | coppice | Is it a baseball ball ball ball? |
| 13:16.57 | coppice | No. Its a pixel! |
| 13:17.30 | tzanger | no you didn't tell me |
| 13:17.32 | porche | :) |
| 13:17.37 | tzanger | sounds good though |
| 13:17.37 | porche | i have hang up problems :) |
| 13:18.17 | quitte | i still can't fetch asterisk with svn. is that a problem on my end? |
| 13:18.56 | tzanger | creativx: heh |
| 13:18.58 | tzanger | I HAS VAR |
| 13:19.13 | tzanger | IM IN UR LOOP |
| 13:19.14 | tzanger | hahahahaha |
| 13:20.05 | Katty | [TK]D-Fender: sorry bout yesterday, i've been running around like a madwoman. |
| 13:20.18 | tzanger | BIPM flash driver (size->0x2000000 mem->0xFC000000) |
| 13:20.19 | tzanger | <PROTECTED> |
| 13:20.27 | tzanger | razzafrazzasonofadammitalltohell... |
| 13:24.23 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 13:24.28 | porche | what's the diff between 1.2 and 1.4 versions? |
| 13:24.51 | Uatec | hey, does anybody know where i can find asterisk business edition isos? to see if there is a newer ISO to download than the one i currently have? |
| 13:24.55 | *** join/#asterisk zdrulio (n=krlozano@82.119.72.130) |
| 13:25.00 | zdrulio | hello all |
| 13:25.07 | zdrulio | i have little problem |
| 13:25.19 | tzanger | Uatec: I would imagine you contact your Digium rep |
| 13:25.42 | Uatec | there's no place online? |
| 13:25.42 | Uatec | lame |
| 13:26.19 | bkruse | :P |
| 13:27.32 | *** join/#asterisk BouYYY (n=BouYYY@81-86-77-70.dsl.pipex.com) |
| 13:27.53 | creativx | manual <3 |
| 13:29.18 | zdrulio | i have 2 asterisk. A1 and A2. connected by IAX. A2 register to A1. i have call rule -> Begins with 9 and followed by 3 digits -> to trunk. but when i call to IAX account in A1 call is OK, but when i call SIP acc -> Unable to create channel of type 'IAX2' (cause 3 - No route to destination) |
| 13:29.23 | zdrulio | can i fix that ? |
| 13:30.40 | [TK]D-Fender | zdrulio: Now why do you think it is that you are getting an error talking about **IAX2** when calling a **SIP** account? I think you are confused on the tech of your call... |
| 13:31.28 | zdrulio | hm. when i call to iax account is OK |
| 13:32.11 | [TK]D-Fender | zdrulio: If you even SEE IAX being referred to when trying to call a SIP account then your Dial is using the wrong account. |
| 13:35.30 | *** join/#asterisk grndslm (n=grndslm@24-116-87-97.cpe.cableone.net) |
| 13:37.19 | zdrulio | uh |
| 13:37.23 | zdrulio | i don`t understand |
| 13:38.04 | zdrulio | what i must do ? |
| 13:38.32 | [TK]D-Fender | zdrulio: Fix your DIAL statement. its clearly not using a SIP account. |
| 13:39.20 | *** join/#asterisk thansen|laptop (n=thansen@151.155.248.199) |
| 13:39.34 | *** join/#asterisk champster (n=asterisk@AH.tescogroup.com) |
| 13:40.03 | *** join/#asterisk Mad|Cow (n=madcow@74.92.109.205) |
| 13:40.45 | zdrulio | and if i want to keep SIP accounts (my co-workers, work with x-lite and it`s not support iax) what you reccomend me ? |
| 13:42.08 | *** join/#asterisk saftsack (n=saftsack@pD9E055AA.dip.t-dialin.net) |
| 13:44.39 | thansen|laptop | what's the best way to marry php and asterisk? I saw 2 diff. projects but maybe there are others.. |
| 13:45.55 | WeBRainstorm | AGI |
| 13:45.55 | *** join/#asterisk frigidzephyr (i=frigidze@nat/digium/x-00a1d32ecac71f02) |
| 13:46.12 | tzafrir | thansen|laptop, for AGI? For manager? |
| 13:46.31 | angryuser | hi |
| 13:46.46 | thansen|laptop | tzafrir: AGI (afaik that's all the scripting/controlling stuff) |
| 13:47.13 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 13:47.14 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:49.26 | *** join/#asterisk pifiu (n=someone@216.5.79.1) |
| 13:49.26 | *** join/#asterisk kclaussen (n=kclausse@204.13.224.242) |
| 13:51.18 | *** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek) |
| 13:53.26 | *** join/#asterisk JraNil64 (n=jranil@85.133.132.5) |
| 13:53.49 | JraNil64 | hello everybody |
| 13:54.03 | JraNil64 | <PROTECTED> |
| 13:54.09 | JraNil64 | <PROTECTED> |
| 13:54.19 | Zeeek | about 279 people |
| 13:54.30 | JraNil64 | yeeah, i see Zeeek |
| 13:54.45 | *** join/#asterisk af_ (n=getsmart@81-174-46-93.f5.ngi.it) |
| 13:55.09 | JraNil64 | is asterisk capable to serve 1000 internal line? |
| 13:56.56 | creativx | how big is your ram and cpu |
| 13:57.03 | creativx | planning to run it on a p233? |
| 13:57.16 | tzafrir | JraNil64, yes. Usually not all the lines are in use. In a typical use you can think about some 20% capacity. Now if you're talking about a call center... |
| 13:57.35 | tzanger | hmm, anyone here use scalix before? |
| 13:57.37 | Zeeek | Dell call center or pr0n call center |
| 13:57.45 | tzanger | wondering about it's ability to tie in to things like asterisk |
| 13:57.47 | JraNil64 | yeah, something like that |
| 13:57.58 | creativx | a pr0n call center.. wait what? you can get porn over the phone??????? |
| 13:58.22 | *** join/#asterisk AlienPenguin (n=Miranda@ip-145-151.sn2.eutelia.it) |
| 13:58.33 | JraNil64 | there is a proposal to use Mitel MN 3300 ICP, and i am thinking if it is possible to use ow big is your ram and cpu |
| 13:58.33 | JraNil64 | <creativx> planning to run it on a p233? |
| 13:58.33 | JraNil64 | <tzafrir> JraNil64, yes. Usually not all the lines are in use. In a typical use you can think about some 20% capacity. Now if you're talking about a call center... |
| 13:58.34 | tzafrir | Something like 200 concurrent calls will take a bit of careful planning. But this is not the main part. |
| 13:59.01 | *** part/#asterisk porche (n=porche@81.215.215.76) |
| 13:59.02 | tzafrir | Making sure that the system is reliable enough is a bit trickier. |
| 13:59.38 | tzafrir | Here just throwing CPU at the problem won't help... |
| 13:59.46 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 14:00.01 | AlienPenguin | Hi, i have aproblem i want asterisk to asnwer to a specific port of a SIP UA (evntually the one present in the contact header) not the port from which the incoming SIP message was received |
| 14:00.05 | AlienPenguin | is it possible? |
| 14:01.09 | *** join/#asterisk eeos (n=eeos@86.53.50.16) |
| 14:01.11 | JraNil64 | tzafrir, what is the main problem, if there is no concern for cpu/ram and redundancy ? |
| 14:01.32 | eeos | Zeeek: yes, it works really well!!!! |
| 14:01.37 | thansen|laptop | tzafrir: is there a php manager? |
| 14:01.49 | Zeeek | eeos oh? What does? :) |
| 14:01.52 | [TK]D-Fender | zdrulio: pastebin the CLI output of your complete call attempt at verbose 10 |
| 14:01.58 | tzafrir | thansen|laptop, ask others. I was just asking some clarification questions |
| 14:02.05 | thansen|laptop | oh :) |
| 14:02.22 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 14:02.26 | eeos | Zeeek: the exercise you gave me. well, you asked me at 12:50, but the pc we use as test server was down. |
| 14:02.39 | eeos | Zeeek: and then lunch break, you know |
| 14:02.43 | tzafrir | JraNil64, redundancy. |
| 14:02.55 | eeos | Zeeek: so now the two softphoes call each other rather well. |
| 14:02.59 | Zeeek | eeos so the lusers can call each other ? |
| 14:03.06 | Zeeek | great! Champagne! |
| 14:03.07 | eeos | Zeeek: yes! |
| 14:03.13 | Zeeek | <PROTECTED> |
| 14:03.35 | eeos | Zeeek: that would be agood idea :) |
| 14:04.12 | JraNil64 | tzafrir, so you mean if I make a complete redundant servers, there would be no lackage in the system? |
| 14:04.25 | Zeeek | eeos now, excercise #2 : talking to your SIP provider |
| 14:04.35 | Zeeek | let me know when it's done |
| 14:04.45 | eeos | Zeeek: I have 2 questions for you: 1) we can only call from one pc to another using the full sip address extension@192.168.x.y. is ther a way to make them call each other using only the extension withou putting the ip address of the server? |
| 14:05.02 | eeos | Zeeek: hey! for exercise 2, I need some bloomig help! |
| 14:05.06 | Zeeek | eeos should work without the adrs |
| 14:05.19 | eeos | Zeeek: it does not! |
| 14:05.19 | Zeeek | ~barebones |
| 14:06.13 | *** join/#asterisk irule (n=irule@189.164.43.19) |
| 14:06.28 | eeos | Zeeek: yes, it works. it did not work because we left the registrations to the external sip provder ON in the softphoes :( |
| 14:07.00 | eeos | Zeeek: so second question: how do I configure the external sip provider so that our asterisk can talk to it? |
| 14:07.18 | eeos | Zeeek: I imagine that we should configure the sip providr as a user. is that right? |
| 14:07.20 | Zeeek | what does and what does not work? |
| 14:07.54 | eeos | Zeeek: calling fom one softphone to another on the local network through asterisk using only the extensions without ip address |
| 14:07.55 | Zeeek | . |
| 14:07.56 | Zeeek | .. |
| 14:08.19 | Zeeek | eeos yes, it works |
| 14:08.55 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 14:08.59 | *** join/#asterisk KriS83 (n=kmarcrof@212.202.141.92) |
| 14:08.59 | Zeeek | have you read this: http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html |
| 14:09.05 | Zeeek | it would help you a lot |
| 14:09.12 | eeos | Zeeek: reading |
| 14:09.12 | KriS83 | Hi |
| 14:09.48 | [TK]D-Fender | Zeeek: Dear God that article is ancient |
| 14:09.58 | Nugget | ha ha "lamp" |
| 14:10.11 | Zeeek | or Laaamp as we now say |
| 14:10.11 | Nugget | "a" is the only letter in there that's worth a shit. |
| 14:10.28 | eeos | Zeeek: looks very interesting, but a bit out of date, does it still work? |
| 14:10.42 | eeos | actually m works very well and so do l and p :P |
| 14:10.42 | Zeeek | yes still works |
| 14:10.54 | *** join/#asterisk NoCarrier (n=NoCarrie@unaffiliated/badpacket) |
| 14:11.26 | Nugget | mysql is a festering piece of maldesigned shite. |
| 14:11.37 | Nugget | it's a horrible, horrible database. |
| 14:11.43 | [TK]D-Fender | Nugget: Don't hold back... tell us how you REALLY feel! |
| 14:11.44 | tzafrir | JraNil64, honestly, I'm not really qualified to tell you how to do that. Some people on this channel might help you for a fee, as they value their time. Alternatively, do your own research... |
| 14:12.31 | eeos | Nugget: what about oracle and postgresql, just to keep you going |
| 14:12.46 | Zeeek | falt files |
| 14:12.48 | JT | they're quite good in comparison to mysql |
| 14:12.50 | JraNil64 | tzafrir, thank you for ur time. sure there would be a fee, for a good ansewr |
| 14:12.51 | Nugget | sometimes you just gotta use oracle, which is fine, but you're foolish if you use it when you don't gotta. |
| 14:12.56 | Nugget | postgresql is pretty dang nice. |
| 14:13.04 | eeos | Zeeek: thanks for the link |
| 14:13.22 | Zeeek | happt to be of service towards your goal of excercise #2 |
| 14:13.57 | Nugget | asterisk is a bit flaky, but it's incredible what you can do with it. Really transforms the phone into a modern tool, imho. |
| 14:14.15 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 14:14.30 | eeos | Nugget: access? sqlserver? firebird? |
| 14:14.47 | eeos | Nugget: I thought we would have slaughtered all database servers first |
| 14:14.59 | Nugget | nobody's come close to what the msaccess front end can do, but the msaccess back end is comically bad. |
| 14:15.34 | [TK]D-Fender | Nugget: No I don't think it turns it into a "modern tool" so much as gives a rather large amount of control with relatively easy means of programming what you want to do. |
| 14:15.36 | Nugget | mssql is great, especially if you're too cheap to pay for a proper dba, but it's mostly because the nerds at sybase knew what they were doing. |
| 14:15.50 | carrar | Postgres |
| 14:15.52 | carrar | !! |
| 14:15.56 | Nugget | I've never used firebird for anything real so I have no comment. |
| 14:16.33 | [TK]D-Fender | Nugget: And yeah... Postrgess is respectable. They are quite concerned with diong the job "right" and if it means they aren't the performance king, big deal.... |
| 14:16.55 | Nugget | I love postgresql. it's what I would generally prefer to use for a project that needed a database. |
| 14:18.08 | Nugget | and really, postgresql performance has been quite respectable for many years now. it's way faster than mysql as soon as you graduate beyond simple selects and start doing real things with the data. |
| 14:19.06 | [TK]D-Fender | Nugget: For really small stuff I could be happy with SQLite + ODBC. |
| 14:19.19 | Nugget | yeah, that's a good solution |
| 14:19.59 | [TK]D-Fender | Nugget: People often think too big. The above combo give network access to a tiny platform that'll do all the basics. |
| 14:20.14 | Nugget | sqlite is a breeze to embed, too. |
| 14:20.52 | [TK]D-Fender | Nugget: I'm a little peaved that * hasn't gone with this approach a bit more natively... |
| 14:22.09 | Nugget | I'm not closely in tune with asterisk development but from my vantage it really seems like digium is getting their act together and approaching the development cycle in a more robust manner (over the past year or so). |
| 14:22.42 | *** join/#asterisk dlynes_laptop (n=dlynes@d207-216-161-56.bchsia.telus.net) |
| 14:22.46 | Nugget | I find that quite encouraging |
| 14:22.55 | *** join/#asterisk matsk (n=mk@194.68.102.175) |
| 14:23.16 | Nugget | I don't feel like I need to closely monitor the lists and this channel just to be able to run a stable asterisk install like I did two years ago when I started. |
| 14:23.36 | carrar | it helps |
| 14:23.46 | carrar | if it's your biz |
| 14:24.46 | [TK]D-Fender | Nugget: Well more than anything else I'm of the belief that we seriously need to crack down on the most "real" aspects that affect real-world use. SIP needs that complete rewrite (all THREE or more of them in the works), app_queue... 'nuff said ... , MFC/R2 native (no external libraries) , ISDN BRI support so that BRIStuffed stuff can be forgotten about. |
| 14:25.12 | Nugget | but it's more fun to futz with dundi and chan_mobile! :) |
| 14:25.53 | *** join/#asterisk kclaussen (n=kclausse@204.13.224.242) |
| 14:26.29 | *** join/#asterisk ESCulapio__ (n=The@66.44.88.200.l.sta.codetel.net.do) |
| 14:26.47 | [TK]D-Fender | Nugget: Yeah... 2 things the world at large SERIOUSLY doesn't care about... |
| 14:28.28 | *** join/#asterisk boxrock (n=chatzill@198.23.33.65.cfl.res.rr.com) |
| 14:28.38 | Corydon76-home | Wouldn't the world not caring about it also mean that the world at large will not interfere in their proper functioning? |
| 14:29.27 | Mercestes | Corydon76-home, assuming it functions properly to begin with. |
| 14:33.22 | *** join/#asterisk ctaloi (n=ctaloi@nat-66-218-1-202.usadatanet.com) |
| 14:34.00 | *** join/#asterisk mazpe (n=lesterm@c-71-206-91-61.hsd1.fl.comcast.net) |
| 14:35.02 | [TK]D-Fender | Nugget: I'd clump AEL 1/2 into that category too.... but then again that was done by a non-digium member of the community more or less alone... |
| 14:35.07 | [TK]D-Fender | IIRC |
| 14:35.10 | *** join/#asterisk RobH (n=RobH@rrcs-24-73-86-239.se.biz.rr.com) |
| 14:35.28 | RobH | Morning folks. Can someone tell me where Asterisk stores its sound files that I record for mailbox names? |
| 14:36.17 | Mercestes | /var/spool/asterisk/voicemail/<context>/<mailbox>/greet.{wav,gsm,mp3} |
| 14:36.19 | [TK]D-Fender | RobH: typically /var/spool/asterisk/voicemail |
| 14:36.28 | Mercestes | ha, beat ya to it. |
| 14:36.49 | Nugget | look in /etc/asterisk/asterisk.conf for the astspooldir location to be sure. |
| 14:37.02 | RobH | That is where I thought, except I put in a new vmail context (called public) and it did not create a public folder. Is this something I have to do manually? |
| 14:37.25 | Corydon76-home | [TK]D-Fender: uh, how was AEL done again? |
| 14:38.07 | Corydon76-home | [TK]D-Fender: AEL 1 was written by Mark. AEL 2 was written by a guy who was subsequently hired by Digium. |
| 14:38.25 | RobH | Nugget: astspooldir => /var/spool/asterisk |
| 14:39.41 | *** join/#asterisk aydiosmio (i=aydiosmi@judecca.aculei.net) |
| 14:40.17 | JT | [TK]D-Fender: 2 things to add to that list: T.38, H.323 |
| 14:40.20 | aydiosmio | anyone ever use the upper registration feature of Brekeke PBX? |
| 14:40.40 | JT | H.323 needs to be implemented properly and reliably in ONE channel driver |
| 14:40.48 | JT | T.38 endpoint needs to happen |
| 14:41.31 | Mercestes | RobH: Try leaving them a voicemaiol |
| 14:41.48 | RobH | Playing with it now, hoping it will create the files that way =] |
| 14:42.47 | *** join/#asterisk luckyone_ (n=hidden@CPE-65-28-7-102.kc.res.rr.com) |
| 14:42.55 | luckyone_ | good day everyone |
| 14:44.42 | KriS83 | Is it possible, if when calling from or to one SIP Account/number, that a second SIP Account/number is shown busy, untill I hang up again? |
| 14:45.07 | *** part/#asterisk frigidzephyr (i=frigidze@nat/digium/x-00a1d32ecac71f02) |
| 14:45.18 | *** join/#asterisk ctaloi (n=ctaloi@nat-66-218-1-202.usadatanet.com) |
| 14:46.20 | [TK]D-Fender | Corydon76-home: Ok, I'm more familiar with #2, and didn't know he was hired. And the subsequent part well... I'll claim a poshumous victory on that one ;) |
| 14:46.25 | Mercestes | KriS83, You'd have to patch the code to manipulate app_state |
| 14:46.28 | [TK]D-Fender | Corydon76-home: Peace :) |
| 14:46.47 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 14:46.49 | KriS83 | Mercestes, hmm.. not so nice :) |
| 14:47.02 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.219.nw.nuvox.net) |
| 14:47.07 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 14:47.19 | Mercestes | KriS83, What are you trying to do? |
| 14:47.36 | [TK]D-Fender | JT : Agreed. I would personally figure getting H.323 solid should be a higher priority than T.38. |
| 14:47.49 | Mercestes | I vote T.38 over H.323 |
| 14:48.03 | KriS83 | Mercestes, well I have 2 Companies, and each has it's own phone numbers... so every employee has 2 incoming and two outgoing lines |
| 14:48.14 | Mercestes | ok |
| 14:48.23 | KriS83 | We use Grandstream GXP2000 Phones and use the BLF Keys |
| 14:48.47 | [TK]D-Fender | Mercestes: I agree that there may be more fax users looking forward, but defer to the fact that lacking a solid base voice protocol is more fundamental to *. |
| 14:48.52 | KriS83 | But now I would have to configure 2 Line-keys for each employee |
| 14:48.54 | Mercestes | ,*-* |
| 14:49.08 | KriS83 | To see if he/she is one the phone |
| 14:49.35 | Mercestes | KriS83, why? |
| 14:49.38 | [TK]D-Fender | KriS83: You have thoroughly misunderstood the concept of "lines" if thats the case... |
| 14:49.53 | [TK]D-Fender | KriS83: "show application chanisavail" |
| 14:50.19 | Mercestes | chanisavail never worked for me, btw. :P |
| 14:50.26 | [TK]D-Fender | KriS83: Users should typically only have 1 registration (line) which may support multiple simultaneous calls. |
| 14:51.03 | [TK]D-Fender | KriS83: For which if any channel is in use would only require 1 BLF entry |
| 14:51.07 | mazpe | anyone seen the error "facility not subscribed" when trying to use idefisk? |
| 14:51.20 | Mercestes | He sleeps with his mercestes effigy like a gothic teddy bear |
| 14:51.33 | KriS83 | [TK]D-Fender, How would you then choose which number to send for outgoing calls? I will not be able to tell everyone to dial a prefix to select a line |
| 14:51.50 | Mercestes | KriS83, Set all your lines into a group. |
| 14:51.53 | Mercestes | and then dial the group. |
| 14:53.10 | [TK]D-Fender | KriS83: can you clarify what you mean by "selecting a line" please... |
| 14:53.38 | [TK]D-Fender | KriS83: What kind of channel is being used and when you say "2", what is the difference between them. |
| 14:53.59 | KriS83 | The number that is sent out and the Zap Group that is used |
| 14:54.44 | *** join/#asterisk __chris (n=chris@unaffiliated/redlined) |
| 14:54.45 | KriS83 | Line 1 = Zap/7 & outgoing Number 1234 and Line 2 = Zap/10 & outgoing Number 4321 |
| 14:55.25 | KriS83 | We only use sip internal, and terminate to PMX/E1 |
| 14:55.42 | [TK]D-Fender | KriS83: Ok so you want them to be able to pick an exact identity to use for outgoing calls basically, but can't get them to jsut change how they dial? |
| 14:55.53 | *** join/#asterisk ber111 (i=brad@neu.cow.org) |
| 14:55.57 | KriS83 | jepp |
| 14:56.10 | KriS83 | They are used to just Press a button to select a line |
| 14:56.11 | Mercestes | KriS83, why does it have to be that way? |
| 14:56.13 | KriS83 | and dial |
| 14:56.20 | ber111 | I noticed when i make international voip calls the volume is low, is there any way to crank the gain up on them |
| 14:56.21 | [TK]D-Fender | KriS83: Ok, well I guess you're stuck then. so "oh well" |
| 14:56.23 | Mercestes | Yea, Asterisk is not a turn-key system. This is *new* technology. |
| 14:56.35 | ber111 | i call thru a softphone |
| 14:56.39 | Mercestes | So you don't have line 1 going to "line 1" on phone 1, 2, 3, 4, whatever. |
| 14:56.56 | Mercestes | You have five lines, *ALL* the phones use *ALL* five lines through dynamic hunting\ |
| 14:57.11 | Mercestes | you only need to use *ONE* line key on the phone and that key will handle over 100 calls. |
| 14:57.12 | [TK]D-Fender | ber111: yeah... increase the recording gain on your mics |
| 14:57.45 | Mercestes | There is no press hold, and tell someone to "pick up line one" Asterisk doesn't work that way. In asterisk, you *transfer* the call to another phone. |
| 14:57.56 | Mercestes | KriS83, It's time to go read again. |
| 14:57.59 | Mercestes | ~book |
| 14:58.11 | jbot | i heard book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 14:58.11 | [TK]D-Fender | Mercestes: He wasn't to push a single line-key to change the line (and CID along with) for the call without mucking with dialplan. Users would reject that so he's caved to the only other option. |
| 14:58.11 | Mercestes | ~wikis |
| 14:58.13 | jbot | rumour has it, wikis is http://www.voip-info.org |
| 14:58.18 | [TK]D-Fender | Mercestes: its the fact he can't have them use a different prefix or anything |
| 14:58.42 | Mercestes | [TK]D-Fender, Your english confuses me. |
| 14:58.43 | [TK]D-Fender | s/have/MAKE/ |
| 14:59.02 | Mercestes | why would he have to use a different prefix? |
| 14:59.32 | Zeeek | can a bug be pre-fixed? |
| 14:59.53 | Mercestes | Zeeek, Only if it's a sub-bug of another parent bug in an ordered list. |
| 15:00.39 | [TK]D-Fender | Mercestes: Each user has 2 outound identities (as dialed through his Zap channels). His users would reject using a single reg to PICK which to use via a dialplan pattern (ie using a different prefix ) and want to press a single line button that KNOWS which to use. |
| 15:00.47 | *** join/#asterisk phillipk (n=pkey@216.248.143.87) |
| 15:01.09 | Mercestes | .... |
| 15:01.15 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 15:01.27 | Mercestes | By outbound identities.....he has employees who pretned they work for two different companies? |
| 15:02.50 | Mercestes | Well that's easy two then. You just setup two "lines" with two sip usernames and route out via g1 or g2 by context or CID. |
| 15:03.03 | *** join/#asterisk zeeesh (i=zeeesh@14-237-154-202.wol.net.pk) |
| 15:03.05 | zeeesh | hi |
| 15:03.06 | Mercestes | s/two/too/ |
| 15:03.11 | Mercestes | You rock, Jbot. |
| 15:03.53 | Zeeek | hi zeeesh |
| 15:05.20 | KriS83 | Mercestes, thats what I have |
| 15:05.37 | [TK]D-Fender | Mercestes: The problem with using 2 SIP users is the fact he needs 2 BLF account on a phone to track that single person. THAT was the point from the beginning |
| 15:05.50 | KriS83 | But now if I dial out with one line, the other is still visable as being available |
| 15:05.52 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 15:06.07 | KriS83 | [TK]D-Fender, thanks.. thats exaclly what I mean |
| 15:06.09 | KriS83 | :) |
| 15:07.01 | [TK]D-Fender | KriS83: You are in a shitty position. Maybe someone will make a mod to the presence code so we can create a hint like : exten => 100,hint,SIP/100a@SIP100b |
| 15:07.09 | [TK]D-Fender | KriS83: That would do it. |
| 15:07.30 | zeeesh | how can i send fake caller id ... or .. receiver will receive a "private" or "fake caller id" ?? bcoz right now ... at my server .. |
| 15:07.36 | file | you can. |
| 15:07.42 | file | [TK]D-Fender: ^^^^ |
| 15:07.48 | [TK]D-Fender | file: z0mg |
| 15:07.56 | [TK]D-Fender | KriS83: there you have it! |
| 15:08.34 | *** join/#asterisk bbryant (i=brett@nat/digium/x-ff0c44953aef6a43) |
| 15:08.59 | KriS83 | [TK]D-Fender, I don't understand? Where do I have what? :) |
| 15:09.02 | *** join/#asterisk SwK (n=SwK@user-69-73-37-99.knology.net) |
| 15:09.12 | [TK]D-Fender | [TK]D-Fender>KriS83: You are in a shitty position. Maybe someone will make a mod to the presence code so we can create a hint like : exten => 100,hint,SIP/100a@SIP100b |
| 15:09.20 | KriS83 | k |
| 15:09.30 | [TK]D-Fender | KriS83: Apparently you CAN. |
| 15:10.18 | KriS83 | I guess not... but thanks for your help |
| 15:10.43 | [TK]D-Fender | KriS83: Guess not what? |
| 15:11.09 | KriS83 | <[TK]D-Fender> KriS83: Apparently you CAN. <- I don't think I can, as I'm not really a coder |
| 15:11.19 | KriS83 | So I guess I can't |
| 15:11.59 | [TK]D-Fender | KriS83: We're talking about setting up a single HINT to replace the 2 you are using now for BLF. How is it you don't think you can do this since you've done twice as much already? |
| 15:12.45 | KriS83 | [TK]D-Fender, sorry I missunderstood you I think |
| 15:13.05 | KriS83 | I was thinking it requires a patch to asterisk code? |
| 15:13.20 | *** join/#asterisk eeos (n=eeos@86.53.50.16) |
| 15:13.26 | eeos | back |
| 15:14.25 | Zeeek | eeos about that excercise "2? Done? |
| 15:14.40 | zeeesh | <[TK]D-Fender>: how to send fake caller id to receiver ? |
| 15:14.42 | __chris | <PROTECTED> |
| 15:15.09 | *** join/#asterisk btsteve (n=btsteve@204.10.20.30) |
| 15:15.18 | [TK]D-Fender | KriS83: .. just make a better HINT in your dialplan as I gae you the example for |
| 15:15.39 | KriS83 | [TK]D-Fender, yepp I'll try that. |
| 15:16.03 | [TK]D-Fender | zeeesh: Use the CALLERID function with a PRI line |
| 15:16.30 | zeeesh | or how to send ... actual callied ? |
| 15:17.03 | *** join/#asterisk nowork (n=jfu2808@216.254.141.97) |
| 15:17.30 | [TK]D-Fender | zeeesh: "exten => _9x.,1,Set(CALLERID(num)=123456789) |
| 15:17.39 | zeeesh | ok |
| 15:17.48 | zeeesh | thnk |
| 15:18.01 | nowork | hi, my sip provider asked me to treat him as gateway. .i tried with type=user or type=peer all failed.he didn't see my call.. any one has any idea what should i change ? |
| 15:18.36 | nowork | 1.2.4 |
| 15:19.16 | JT | nowork: froend |
| 15:19.18 | JT | friend |
| 15:19.20 | JT | and 1.2.4 wtf |
| 15:19.25 | JT | that's incredibly old |
| 15:19.27 | JT | upgrade now |
| 15:21.07 | [TK]D-Fender | nowork: And considering you gave no real details, not debug info, no config samples, there is only about a MILLION things you could ahve done wrong before we even get around to asking about NETWORKING isuues you might have. |
| 15:21.43 | [TK]D-Fender | "Hi I bought a SIP line but it doesn't work, WHY!@!!@!!@ PLEASE!!@" |
| 15:21.52 | Zeeek | I know why |
| 15:22.30 | nowork | hm, let me check debug ifno |
| 15:22.33 | nowork | info |
| 15:23.12 | nowork | asterisk1*CLI> sip show peers and I see his ip is UNREACHABLE |
| 15:23.30 | Zeeek | that doesn't mean much |
| 15:24.30 | btsteve | Does anyone have anyidea what might be happening: we have a soft switch that is sending calls to our asterisk voicemail server. the call comes in and we hear the prompts "blah blah hit # for more options and then the beep" we try leaving a message and we always get a 3 or 4 sec message and no audio the asterisk is hanging up the call right after it geive the beep. |
| 15:24.30 | *** join/#asterisk hfb (n=hfb@pool-72-87-254-188.lsanca.dsl-w.verizon.net) |
| 15:24.42 | monstertruck | does anybody know of a gateway, like the spa3000 that can do iax2? |
| 15:25.15 | nowork | checked debug: I see this : |
| 15:25.16 | nowork | May 31 10:21:02 NOTICE[12890] app_dial.c: Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
| 15:25.16 | nowork | May 31 10:21:02 VERBOSE[12890] logger.c: == Everyone is busy/congested at this time (1:0/0/1) |
| 15:25.16 | nowork | May 31 10:21:02 DEBUG[12890] app_dial.c: Exiting with DIALSTATUS=CHANUNAVAIL. |
| 15:25.49 | Zeeek | nowork what is needed is a pastebin sample of your sip.conf entry |
| 15:26.27 | Mercestes | btsteve, Try calling an "Answer()" before you Voicemail(u${exten}) it. |
| 15:26.37 | [TK]D-Fender | What is also needed is to see what was ACTUALLY DIALED. |
| 15:26.49 | Zeeek | no, that would make it TOO EASY |
| 15:26.59 | [TK]D-Fender | Who cares about the error message when we can't see if you jsut mangled the hell out of your DIAL STATEMENT. |
| 15:27.15 | Zeeek | must be a Polycom issue |
| 15:27.43 | [TK]D-Fender | Why does nobody get that? Don't whine that you don't like the answer when we can't see the QUESTION. |
| 15:27.54 | *** join/#asterisk unixlike (n=dsadas@31.67.modemcable.oricom.ca) |
| 15:28.03 | [TK]D-Fender | Zeeek: You are clearly too wise for us :) |
| 15:28.07 | Zeeek | in all fairness, one doesn't know the message is that ambiguous |
| 15:28.33 | *** part/#asterisk thojo (n=ttr@0x5733db9d.bynxx19.adsl-dhcp.tele.dk) |
| 15:28.50 | Zeeek | the no path to for example - after yiu get used to the fact that just just means "I can't reach" and add this to the fact that the peer was "UNREACHABLE" |
| 15:29.06 | unixlike | Do anyone could tell me the best settings for payload/jitter buffer i could use for a Polycom 601 or it really depends on my infrastructure ? |
| 15:29.16 | Zeeek | so asterisk says "I can't reach the UNREACHABLE sip peer" |
| 15:29.34 | iCEBrkr | I'm going back to Tampa, Tampa, Tampa, Tampa. I'm going back to Tampa, Tampa, Tampa, Hrrrm, I don't think so. |
| 15:29.38 | Zeeek | Polycom? [TK]D-Fender |
| 15:29.48 | [TK]D-Fender | Zeeek: Technically we don't even know that he's CALLING through his SIP peer. |
| 15:30.20 | Zeeek | well, we know he's supposed to be |
| 15:30.32 | [TK]D-Fender | Zeeek: And thats about it :) |
| 15:30.36 | Zeeek | yup |
| 15:30.37 | nowork | sip.conf should be fine..it's working to other sip supplier. |
| 15:30.56 | Zeeek | then I guess you can take 'er from here, eh? :) |
| 15:30.57 | btsteve | Mercestes: do i just insert Answer() infront of Voicemail() |
| 15:31.06 | [TK]D-Fender | nowork: Should use your DIAL attempt and your SIP peer entry in a PASTEBIn masking ONLY passwords |
| 15:31.15 | Mercestes | btsteve, Yes. and be sure to renumber your priorities too |
| 15:31.28 | Zeeek | Mercestes or use 'n' priorites |
| 15:31.44 | btsteve | Mercestes:so it should be a seperate line ok |
| 15:31.57 | Mercestes | Not entirely true. I tend to put Answer() before anything else if I use it at all, and you cannot start with priority "n" |
| 15:32.16 | *** part/#asterisk zdrulio (n=krlozano@82.119.72.130) |
| 15:32.16 | Zeeek | I know, which is why I alwasy start with a noop |
| 15:32.31 | btsteve | Mercestes:any other parameters i need in the () after it |
| 15:32.35 | luckyone_ | why am I not able to attach to * with asterisk -rvv -U asterisk -G asterisk ? |
| 15:32.50 | luckyone_ | my machine shows a running * |
| 15:33.09 | Mercestes | luckyone_: uhh....did you try just asterisk -r ?? |
| 15:33.16 | luckyone_ | Mercestes: yes |
| 15:33.28 | [TK]D-Fender | luckyone : perhaps you could pastebin your complete attempt and some proof of who its running as. |
| 15:33.30 | [TK]D-Fender | ~pb |
| 15:33.32 | jbot | from memory, pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or at paste.lisp.org |
| 15:33.34 | [TK]D-Fender | ^^^^^^^^ |
| 15:34.09 | luckyone_ | [TK]D-Fender: will do, one moment |
| 15:34.09 | tzafrir | luckyone, are you trying to start asterisk or connect to an existing asterisk? |
| 15:34.09 | Zeeek | what is pastebin? |
| 15:34.09 | luckyone_ | tzafrir: connect via cli to * |
| 15:34.09 | JT | what's with the assumptions this week |
| 15:34.15 | Mercestes | Zeeek: It's IRC-speak for "please paste all information into this channel at once." |
| 15:34.16 | JT | so many newbies assuming their configs "should be working" |
| 15:34.22 | JT | me bashes head against wall |
| 15:35.49 | Mercestes | I agree, their configs *should* be working. The question is, why they are not. |
| 15:36.23 | JT | like they all are coming from the secret order of the Classified Config Files |
| 15:36.31 | nowork | :)..let me try pastebin.ca |
| 15:36.38 | Mercestes | It's a new syntax they don't want to get out. |
| 15:36.46 | *** join/#asterisk eeos (n=eeos@86.53.50.16) |
| 15:36.51 | eeos | back again |
| 15:37.57 | luckyone_ | Mercestes: my config is working |
| 15:37.57 | Mercestes | luckyone_: prove it. |
| 15:37.58 | luckyone_ | Mercestes: my sip phone rings when I dial my number |
| 15:38.02 | [TK]D-Fender | JT : "If it was fine, you wouldn't be in here whining!" |
| 15:38.06 | Mercestes | my sip phone shits gold when i dial it. |
| 15:38.25 | luckyone_ | Mercestes: lol, you have to give me your sip.conf!!! |
| 15:38.30 | btsteve | Answer() did not do the trick we are getting the 4 sec voicemail in our e-mail box before the call is eve dicconnected |
| 15:38.35 | Mercestes | luckyone_: but it works fine. |
| 15:38.38 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 15:38.40 | [TK]D-Fender | Mercestes: Problem being that implies your phone was already full of shit... |
| 15:39.09 | JT | [TK]D-Fender: maybe they misread you as saying "masking only NON PASSWORDS" and don't trust you |
| 15:39.14 | Mercestes | [TK]D-Fender, Again, you fail to grasp basic english. Shit here is being used as a verb, not a noun. Gold being the noun in question. |
| 15:39.29 | Zeeek | btsteve the echo test works with 2 way audio? |
| 15:39.29 | Mercestes | :P |
| 15:40.01 | [TK]D-Fender | Mercestes: Shit is shit... doesn't matter what you ATE :) YOU fail to grasp english... |
| 15:40.03 | [TK]D-Fender | pwned |
| 15:40.05 | Mercestes | btsteve, Here is a silly question, are you using SIP from * to the softswitch? If yes, are you Natting or firewalling that SIP connection?? |
| 15:40.28 | kippi | how can I run a bash script from the dial plan? |
| 15:40.29 | Mercestes | [TK]D-Fender, ....? |
| 15:40.38 | [TK]D-Fender | *sigh* |
| 15:40.47 | luckyone_ | [TK]D-Fender: http://paste.ubuntu-nl.org/23451/ |
| 15:40.50 | btsteve | The softswitch is in the same location on the same network. not firewall between the too |
| 15:41.03 | Mercestes | btsteve, "no" would've sufficed. ;) |
| 15:41.04 | btsteve | How do i do the Echo test. |
| 15:41.10 | btsteve | Sorry |
| 15:41.13 | Mercestes | S'ok. |
| 15:41.34 | Mercestes | exten => 1234,1,echotest() or something like that. |
| 15:42.09 | *** join/#asterisk Cyber-Dogg (i=Cyber-Do@24-178-240-97.dhcp.stls.mo.charter.com) |
| 15:42.13 | Mercestes | Oh, I was wrong, it's just Echo |
| 15:42.20 | Mercestes | exten => 1234,1,Echo |
| 15:42.30 | JT | Answer |
| 15:42.31 | JT | first |
| 15:42.39 | JT | doesn't always work correctly otherwise |
| 15:42.43 | btsteve | ok will try |
| 15:43.14 | Mercestes | s/exten => 1234,1,Echo/exten => 1234,1,Answer() and exten => 1,2,Echo/ |
| 15:43.23 | JT | if echo fails, change eho for Playback(somefile) |
| 15:43.34 | JT | like Playback(tt-monkeys) |
| 15:43.34 | Mercestes | gah. |
| 15:43.50 | Mercestes | s/exten => 1234,1,Answer() and exten => 1,2,Echo/exten => 1234,1,Answer() and exten => 1234,2,Echo |
| 15:43.52 | Mercestes | s/exten => 1234,1,Answer() and exten => 1,2,Echo/exten => 1234,1,Answer() and exten => 1234,2,Echo/ |
| 15:43.58 | Mercestes | ...oops |
| 15:44.00 | JT | Mercestes: give yp |
| 15:44.01 | JT | up |
| 15:44.10 | Mercestes | I conceed. |
| 15:44.16 | btsteve | ok echo test did not work |
| 15:45.30 | [TK]D-Fender | luckyone : It looks like it begins to connect, you get a few messages , THEN it bombs... wierd |
| 15:46.33 | JT | A101d still sitting at USD$1k, drop already :P |
| 15:46.46 | btsteve | ok echo test did not work |
| 15:46.49 | JT | what is with people charging more for the express versions too |
| 15:47.43 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 15:50.21 | *** join/#asterisk ploieel (n=manni@Fb2d1.f.ppp-pool.de) |
| 15:51.11 | [TK]D-Fender | JT : ? |
| 15:51.16 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 15:51.40 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 15:51.57 | Mercestes | btsteve, define "not work?" |
| 15:52.52 | btsteve | We did not get an echo back |
| 15:53.41 | jkiff | What would cause reloads to take forever? (In my case, I enter "reload", and it just sits there.) |
| 15:54.07 | [TK]D-Fender | JT : US import is loking REALLY appealing as GWB has sent their dollor on a march to hell. CAD = .93 USD. I remember it being at at .60 and below in the past decade..... |
| 15:54.53 | btsteve | ok we did Playback(tt-monkeys) it worked and brought half the office to our cubicles |
| 15:54.54 | [TK]D-Fender | JT : $899 - http://www.telephonydepot.com/product_p/105-052-101d.htm |
| 15:55.00 | JT | [TK]D-Fender: yes AUD to USD exchange rate is excellent also |
| 15:55.06 | JT | hmm |
| 15:55.30 | Mercestes | and yet you still compare your dollars to ours. |
| 15:55.43 | JT | nice, everywhere else is still at $999 |
| 15:56.06 | JT | Mercestes: yours gets weaker and weaker every day :P |
| 15:56.30 | Mercestes | btsteve, Think it might be time for some very verbose debugging output. Something is definately wrong there. Seem syoru getting one way audio from yoru endpoint to asterisk (which should not happen). Are you sure you don't have some kind if built in firewall hiding somewhere? |
| 15:57.09 | Mercestes | JT: welcome to the top btw. ...or...are we at the bottom now? |
| 15:57.51 | *** part/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 15:57.57 | btsteve | I have a ton of debug info |
| 15:58.01 | *** join/#asterisk brookshire (i=mbrooks@hijacked.us) |
| 15:58.20 | Mercestes | pastebin.ca |
| 15:59.01 | JT | Mercestes: pretty much :P |
| 16:00.28 | Mad|Cow | Is anyone hosting multiple PBX's from one system? I'm looking for some pointers on how I might be able to run a system with multipe customers shareing the same hardware. |
| 16:00.44 | btsteve | http://www.pastebin.ca/525893 |
| 16:01.00 | JT | [TK]D-Fender: does sangoma work with canadian dollars of us dollars natively? |
| 16:01.34 | JT | if cad, the exchange rate thing must shit them |
| 16:02.17 | [TK]D-Fender | Mercestes: JT : http://www.ratemyeverything.net/post/987/Canada_vs_the_United_States.aspx |
| 16:02.40 | [TK]D-Fender | JT : Dunno.... |
| 16:04.47 | JT | hrm, only 50% more for an A102d instead of A101d... |
| 16:05.09 | zeeesh | <[TK]D-Fender>: or if i want to send real caller id to receiver end, (note:i m authenticating all of my users from their caller id) then where to change ???? |
| 16:05.47 | JT | Set(CALLERID(num)=911) |
| 16:09.56 | nowork | JT, TK, what should I pate on the bin..for troubleshooting? |
| 16:11.01 | *** join/#asterisk RyanW (n=cableguy@d58-106-136-86.sbr4.nsw.optusnet.com.au) |
| 16:12.25 | RyanW | Hello, i have a system with 3 TDM400p (4x FXO) and also a TE210P but the module loading order for the zaptel channels is not always the same so consequently asterisk fails to start half the time after a reboot. |
| 16:13.18 | JT | digium only recommends a maximum of 2 cards per pc btw |
| 16:13.40 | JT | RyanW: why on earth do you have 12 FXO ports anyway? |
| 16:14.02 | mog | 2400p |
| 16:14.06 | RyanW | customer has 12 PSTN lines at the moment and we're porting their numbers to the new E1 service next week |
| 16:14.06 | mog | would be way to go |
| 16:14.24 | JT | ah ok |
| 16:14.27 | JT | optus pri? |
| 16:14.28 | zeeesh | right now using this format "exten => XXXXXX,1,deadagi(astcc.agi,${CALLERIDNUM})" if i wil change it to this "set(CALLERID(num)=911)" then will it creat any disturbance at my agi file or not ?? |
| 16:14.31 | RyanW | yes. |
| 16:14.36 | JT | right |
| 16:14.42 | JT | the way to go :) |
| 16:14.44 | *** join/#asterisk Defraz (n=t0tal@fw.fuzecore.com) |
| 16:14.54 | nowork | JT? |
| 16:15.31 | JT | zeeesh: i have no idea, and ${CALLERIDNUM} is deprecated |
| 16:15.39 | RyanW | any ideas why sometimes the TDM400p beats the TE210P to load ? |
| 16:16.13 | JT | RyanW: why not disable the te210p and then later remove the tdms when the numbers are ported? seems like an easier way around |
| 16:17.15 | RyanW | I'd like both to avoid downtime during the porting |
| 16:17.35 | JT | you might be able to get the module loading more downpat in the init scripts, i dunno |
| 16:17.47 | JT | but 4 digium cards, behaviour is pretty unpredictable |
| 16:18.30 | RyanW | 74 zap channels is pretty crazy |
| 16:18.38 | kippi | how can I run a bash script from the dial plan? |
| 16:18.44 | RyanW | System |
| 16:18.44 | JT | it's fine if they're pri channels :) |
| 16:18.58 | RyanW | TrySystem |
| 16:21.48 | *** join/#asterisk juanjoc (n=juanjoc@200.69.219.113) |
| 16:23.53 | JT | RyanW: since it's only temporary, might be easier to grap an old desktop or server and chuck the TDM400Ps in it |
| 16:24.08 | JT | and get it to take the analogue calls and send them to the main server |
| 16:24.14 | JT | s/grap/grab/ |
| 16:27.35 | *** join/#asterisk steliosk (n=Stelios@62.169.217.209) |
| 16:27.48 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 16:31.50 | RyanW | JT, cheers' i worked out how to disable the TE210P's module from loading |
| 16:33.21 | kippi | how can I run a bash script from the dial plan? |
| 16:33.47 | [TK]D-Fender | kippi: "show application system" |
| 16:33.57 | kippi | thanks |
| 16:38.31 | RyanW | JT, i had to add the zaptel modules to the modprobe blacklist so the kernel wouldn't load them before the init script |
| 16:40.34 | MrWup | with ExecIf, how do i do a && ? |
| 16:40.44 | MrWup | exten => s,n,ExecIf($[${INCOMING} = 0]|MixMonitor|${LOGGING_KEY}-${XFERCOUNT}.wav|ab) |
| 16:40.46 | MrWup | thats my line |
| 16:41.03 | MrWup | but i want to check that incoming = 0 and xfercount < 3 |
| 16:41.44 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-72-65-27-9.bflony.east.verizon.net) |
| 16:41.51 | SuPrSluG | hello all |
| 16:42.27 | SuPrSluG | any difference between tdm400p and tdm400b? |
| 16:42.46 | *** join/#asterisk vader-- (n=me@c-71-226-197-0.hsd1.nj.comcast.net) |
| 16:42.48 | vader-- | hola |
| 16:43.08 | vader-- | any of you guys know if there is a pay phone(analog or digital) that works with asterisk? |
| 16:43.11 | Qwell[] | SuPrSluG: b indicates that it was purchased as a bundle |
| 16:43.49 | Qwell[] | the actual base board is a tdm400p.. tdm400b technically doesn't exist, because that means no modules, which makes it not a bundle...which makes my head hurt |
| 16:44.15 | Qwell[] | file: with ec? |
| 16:44.16 | SuPrSluG | one's w/ mods one's w/out gotcha thanx |
| 16:44.22 | Qwell[] | SuPrSluG: well, not even that.. |
| 16:44.34 | file | Qwell[]: no, ei... echo introduction |
| 16:44.37 | Qwell[] | SuPrSluG: you'd usually say like "tdm410b", which means one module (fxo? I always forget) |
| 16:44.41 | Qwell[] | file: excellent |
| 16:44.56 | file | includes hardware jpah support |
| 16:45.10 | Qwell[] | SuPrSluG: If you say tdm400p, everybody will know what you're talking about |
| 16:45.15 | Qwell[] | file: in hardware? scary |
| 16:46.11 | SuPrSluG | looking at one on ebay that comes w/ mods cheap. just doing the homework so no suprises |
| 16:46.38 | Dr-Linux | can i use custom MOH in for different queues? |
| 16:46.59 | SuPrSluG | the 1st number rerpresenst fxs and the 2nd fxo. |
| 16:47.04 | *** join/#asterisk cr4z3d (n=cr4z3d@ip70-162-116-242.ph.ph.cox.net) |
| 16:48.52 | *** join/#asterisk ploieel (n=manni@Fb231.f.ppp-pool.de) |
| 16:48.53 | RyanW | Dr-Linux, yes you can, set the musicon hold before entering that queue |
| 16:49.30 | RyanW | Dr-Linux, use the MusicOnHold app to do so. |
| 16:49.43 | jkiff | Hi all, I have a queue on a 1.2.18 box made up of permanent members (member => SIP/201, for instance), and agents who can log in and out (member => Agent/202, for instance). If you call the queue, and a permanent member tries to do an attended transfer with *2, it works just fine. However, when an agent does it, the call disconnects when the agent presses *. I checked features.conf, and *0 is disconnect. How might I go about making attended t |
| 16:49.51 | Dr-Linux | RyanW: i'm already using Native MOH, but i need a little hints, basically i want to use different MOH files for particular clients |
| 16:50.10 | *** join/#asterisk wunderkin (i=wunderki@ip68-104-149-97.ph.ph.cox.net) |
| 16:50.37 | [TK]D-Fender | Dr-Linux: You set the music class in each queue. Go read the sample again |
| 16:50.59 | Dr-Linux | [TK]D-Fender: thanks, reading |
| 16:51.10 | RyanW | Do you want diff music per queue or callers? |
| 16:52.27 | [TK]D-Fender | RyanW: Get with the program.... Dr-Linux>can i use custom MOH in for different queues? |
| 16:52.36 | *** join/#asterisk alrs (n=lars@pozug.com) |
| 16:52.48 | Dr-Linux | RyanW: per queue. Also i'm thinking to provide a way that my customer should upload their wav/mp3 files |
| 16:53.58 | RyanW | anyway you can gonfigure the music in the queues.conf file or leave that setting blank and it will use the dafault class when joining the queue unless you change the default class for this particular caller with the command MusicOnHold in the dialplan before Queue() |
| 16:54.43 | RyanW | i recently setup a system to allow callers to select different hold music using 1-9 while on hold. the catch was they lost their place in the queue every time they changed it. |
| 16:55.39 | Dr-Linux | RyanW: can i /msg you? |
| 16:55.49 | RyanW | sure. |
| 17:02.42 | killfill | hello! |
| 17:03.09 | killfill | yellow alarms are givvimg me a headake.. :S.. all calls are getting hangup when this happend: |
| 17:03.12 | killfill | http://www.pastebin.ca/526032 |
| 17:03.30 | killfill | why would one get this alarms? |
| 17:03.32 | killfill | (its a TE110p |
| 17:03.34 | killfill | ) |
| 17:05.12 | *** join/#asterisk notoriousrab1982 (n=notoriou@76.195.14.206) |
| 17:05.16 | killfill | what could i check?.. im lost about this.. :( |
| 17:05.41 | *** join/#asterisk cayorde (n=flexable@host164-105-dynamic.16-87-r.retail.telecomitalia.it) |
| 17:06.03 | vader-- | any of you guys know if there is a pay phone(analog or digital) that works with asterisk? |
| 17:06.29 | RyanW | vader, why not buy a conventional pay phone and connect it to an ata |
| 17:08.08 | *** join/#asterisk [[blah]asfd (n=ckwall@63.149.122.93) |
| 17:08.43 | SuPrSluG | i'm having an issue w/ a user here that says she got one way audio after being transfered to a UPS supervisor. He hung up on her. Any way to tell if it's my end or a providers? |
| 17:09.30 | SuPrSluG | so one way audio after a transfer. |
| 17:10.10 | jkiff | In reference to my question a minute ago, http://pastebin.ca/526052 that's what happens on the CLI. The arrow denotes where * was pressed. |
| 17:10.41 | irule | hi there, anybody know what is the difference between hylafax and hylafax+? |
| 17:11.26 | *** join/#asterisk codazoda (n=Joel_Dar@mail.hurdmanivr.com) |
| 17:12.01 | vader-- | how will the pay part work with asterisk ryanW? |
| 17:12.09 | *** part/#asterisk codazoda (n=Joel_Dar@mail.hurdmanivr.com) |
| 17:12.49 | [[blah]asfd | is it possible to pass account code from one asterisk server to another? For example I have two servers in the office that set account code and all of that, but when they connect to the outside world the calls are set out via a third server at my data center. Rather than doing a complex dial plan on the third server to set the account code of who placed the call, can I just pass from the servers 1 and two the account code that was set there? This woul |
| 17:15.07 | *** join/#asterisk boch (n=fran@190.48.217.225) |
| 17:16.22 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 17:17.11 | *** join/#asterisk aptura (n=jondoe@S010600a0c93f6f7e.vs.shawcable.net) |
| 17:19.13 | Zeeek | . |
| 17:20.49 | k31th | hows it all going |
| 17:22.53 | *** join/#asterisk thinko (n=jdoe6alp@smaug.rackdragon.com) |
| 17:28.22 | *** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
| 17:32.39 | *** join/#asterisk lokkju_wrk (n=lokkju@unaffiliated/lokkju) |
| 17:33.57 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 17:34.24 | *** join/#asterisk nuonguy (n=john@c-24-6-175-26.hsd1.ca.comcast.net) |
| 17:34.33 | blitzrage | afternoon all -- does anyone want to elaborate on what "timeoutrestart" does in queues.conf? I've read the description in the sample file, but I still don't quite *get* it |
| 17:35.31 | *** join/#asterisk dahunter3 (n=dahunter@static-71-118-10-18.lsanca.dsl-w.verizon.net) |
| 17:35.55 | *** join/#asterisk Renacor (n=Renacor@p54b9efb8.dip.t-dialin.net) |
| 17:36.20 | Renacor | anybody familiar with zap channels not dropping after a caller hangs up on analog lines? |
| 17:36.37 | *** join/#asterisk adr3nalin3 (n=Foobar@65-123-189-218.dia.static.qwest.net) |
| 17:38.48 | unixlike | Do anyone could tell me the best settings for payload/jitter buffer i could use for a Polycom 601 or it really depends on my infrastructure ? |
| 17:39.56 | linagee | does anyone know how to use tc/red/qdisc instead of just tc/qdisc? (it's supposed to not drop packets as harshly) |
| 17:40.22 | blitzrage | unixlike: it depends how much jitter is actually on your line -- you make the buffer too small, it isn't as effective, make it too big, you introduce too much unnecessary latency |
| 17:41.03 | linagee | unixlike: why not use the adaptive jitterbuffer? |
| 17:42.46 | dahunter3 | I keep trying to add a *55 extension and my phone keeps saying call failed... I'm not seeing anything on the console... does that mean that the phone is not sending it to the server? |
| 17:43.01 | *** join/#asterisk `pariah (n=josh@unaffiliated/pariah) |
| 17:44.41 | blitzrage | linagee: I find it actually causes issues |
| 17:44.41 | [TK]D-Fender | dahunter3: With SIP debug enabled for its IP? |
| 17:44.56 | Zeeek | issues.... bad |
| 17:45.09 | blitzrage | I think it was causing segfaults or something if I remember correctly |
| 17:45.25 | blitzrage | can't remember... but I couldn't end up using it unfortunately |
| 17:45.33 | blitzrage | seemed like a good idea :) |
| 17:45.40 | blitzrage | so now I just add 20ms of jitter statically |
| 17:45.46 | blitzrage | errr... jitterbuffer :) |
| 17:45.50 | blitzrage | not jitter |
| 17:45.58 | dahunter3 | [TK]D-Fender: Good call, I didn't have it on. I'm looking again with it on. |
| 17:46.12 | unixlike | linagee: is there any adaptive jitter buffer in a polycom 601 ? |
| 17:46.13 | *** join/#asterisk ixela (i=ixela@nat/digium/x-81a783aad593b2d4) |
| 17:46.22 | Zeeek | blitzr adding jitter would improve quality, as long as the jitter is negative |
| 17:46.39 | blitzrage | Zeeek: :) |
| 17:46.41 | Uatec | hey, does anybody know any PCI SATA RAID controllers that work with poundsign linux? |
| 17:47.45 | Zeeek | dahunter3 what phone? |
| 17:47.45 | dahunter3 | Okay, I see it sending some sip header like: To: *55203 <sip:*55203@192.168.1.100:5060>;tag=as36207ca2 --- Does that mean asterisk is getting it but can't find what to do with it in the dial plan? |
| 17:47.45 | dahunter3 | Zeeek: Aastra 480i |
| 17:48.11 | Zeeek | dahunter3 is there an extension like that? |
| 17:48.17 | linagee | unixlike: oh in the phone itself? *shrug*. try messing with asterisk. |
| 17:49.03 | dahunter3 | Well, I was following this: http://www.voip-info.org/wiki/index.php?page=Asterisk+and+Aastra+Phones. And made extensions like this: |
| 17:49.04 | mcab | unixlike: I think polycoms have an adaptive jitter buffer, but I've never messed with it |
| 17:49.20 | dahunter3 | extensions_additional.conf:exten => *55.,1,Set(_ALERT_INFO=info=alert-autoanswer) |
| 17:49.21 | dahunter3 | extensions_additional.conf:exten => *55.,2,Dial(SIP/${EXTEN:3},12,Tt) |
| 17:49.43 | Zeeek | with the underbar before? |
| 17:49.47 | Zeeek | or not? |
| 17:50.04 | blitzrage | dahunter3: you forgot the _ |
| 17:50.13 | Zeeek | echo test |
| 17:50.14 | blitzrage | _ ====== pattern match! |
| 17:50.17 | dahunter3 | blitzrage: ahh! |
| 17:50.29 | Zeeek | underbar undaerbar |
| 17:50.29 | blitzrage | ALWAYS _ with pattern match :) |
| 17:50.37 | blitzrage | underscore... crazy Euros |
| 17:50.53 | Juggie | blitzrage, how about those ducks! :> |
| 17:51.07 | blitzrage | Juggie: screw you! |
| 17:51.19 | Juggie | blitzrage, you had to know the sens would choke! |
| 17:51.26 | blitzrage | Juggie: of course |
| 17:51.29 | blitzrage | they're the Sens |
| 17:51.32 | blitzrage | what do you expect? |
| 17:51.37 | blitzrage | (nothing less) |
| 17:51.42 | Zeeek | sports? |
| 17:51.42 | Juggie | blitzrage, i've been waiting for it for 3 rounds now :) |
| 17:51.43 | blitzrage | or rather... |
| 17:51.47 | blitzrage | (norhing more :)) |
| 17:51.53 | unixlike | mcab: i don't see any setting about adaptive .. in the web interface, maybe it's in the cfg files |
| 17:51.58 | Juggie | anaheim is going to take this series in 5 or 6. |
| 17:52.06 | Juggie | i kind of hope its 6 so i make the money off my game6 tickets :) |
| 17:53.21 | dahunter3 | That worked like a champ. thanks guys! |
| 17:53.49 | Zeeek | don't mention champs at the moment |
| 17:58.34 | fetcher | for a site with 8 CO lines, is there any drawback to using 2 separate TDM400 cards, instead of a TDM800 with 2 quad modules, or a higher-density card from Sangoma, etc? |
| 17:59.23 | ber111 | pci space? |
| 17:59.29 | fetcher | I thought 2*TDM400 might be less likely to lose all ports at once after a lightning hit or other hardware failure |
| 17:59.42 | Zeeek | always a little risky to use two PCI cards |
| 17:59.44 | ber111 | if lightning hits your box youa re in trouble |
| 17:59.53 | fetcher | ber111: they'll be the only two PCI cards intalled. Each needs only one interrupt, right? |
| 18:00.01 | mcab | unixlike: ugh, don't touch the web interface if you can help it... |
| 18:00.02 | fetcher | Zeeek: timing issues? Too many interrupts? |
| 18:00.20 | Zeeek | fetcher yes, depending on mobo and distro etc |
| 18:00.24 | ber111 | i've never had any issues using multiple cards from service perspective |
| 18:00.30 | ber111 | sometimes i forget which ports are #1 |
| 18:00.31 | Zeeek | I happen to have 3 digium cards and works fine |
| 18:00.49 | Zeeek | but many people have had problems |
| 18:00.58 | Zeeek | many other havce had NO problems |
| 18:01.11 | Zeeek | so if you can experiemtn, why not? |
| 18:01.25 | *** join/#asterisk spinull (n=fu69@adsl-69-153-189-214.dsl.lbcktx.swbell.net) |
| 18:01.30 | fetcher | my only experience with multiple cards so far is filling a box with 1-port X100P's, which worked ok (well, no worse than usual :) |
| 18:01.54 | spinull | i need to know about asterisk |
| 18:01.57 | Zeeek | assuming you've got the useless stuff turned off, go for it! |
| 18:02.12 | spinull | is Distibuted Universal Number a feature that lets you share one line among many people |
| 18:02.13 | Zeeek | spinull what exactly |
| 18:03.44 | *** join/#asterisk bkunyiha (i=Billk@66.113.79.5) |
| 18:03.44 | spinull | can one pstn line accept multiple incoming calls and have asterisk distribute them to several phones |
| 18:05.14 | bkunyiha | I have a question on perl agi |
| 18:05.14 | bkunyiha | anyone here to help? |
| 18:05.14 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 18:05.15 | [[blah]asfd | can anyone tell me how to pin out the 2.5 mm mini jack to an rj11 mod plug? the mini jack has blue red yellow and unshielded wires on it. What color goes to what position number on the rj11? |
| 18:05.15 | [[blah]asfd | cant find any diagrams on google |
| 18:05.15 | Zeeek | [[blah]asfd been there! Impossible to find |
| 18:05.15 | [[blah]asfd | Zeeek:do you know the answer? |
| 18:05.15 | bkunyiha | Please IM me for a question on asterisk perl agi |
| 18:05.15 | Zeeek | If I did I'd post it, sorry |
| 18:05.18 | [[blah]asfd | thanks |
| 18:05.28 | Zeeek | I finally bought the proper headsets |
| 18:05.36 | spinull | Zeeek? |
| 18:05.54 | Zeeek | spinull look at http://www.dundi.com |
| 18:06.12 | Zeeek | I don't think it's what you need |
| 18:06.52 | *** join/#asterisk TimRiker (n=timr@pdpc/supporter/bronze/TimRiker) |
| 18:07.01 | spinull | no |
| 18:07.27 | spinull | what i need is a way for people to continuously call a number and not get a busy signal |
| 18:07.49 | spinull | on one line |
| 18:08.36 | [TK]D-Fender | spinull: sure.... if that line is a T1 circuit. |
| 18:09.23 | *** join/#asterisk IPmonger (n=ipmonger@c-68-84-208-206.hsd1.pa.comcast.net) |
| 18:09.38 | *** part/#asterisk IPmonger (n=ipmonger@c-68-84-208-206.hsd1.pa.comcast.net) |
| 18:09.45 | spinull | why does it have to be a T1 |
| 18:09.57 | spinull | why not just a regular line with call waiting |
| 18:10.25 | spinull | but rather than the caller waiting, you pick it up from another phone |
| 18:10.37 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
| 18:11.46 | spinull | just like if im on the phone and someone calls and i don't pick up, it goes to the answering machine |
| 18:12.40 | SuPrSluG | should polycoms use canreinvite=no as arule of thumb |
| 18:12.44 | phillipk | Because once you've picked up an analog line, you've completed a circuit. The only phone at that point that would get the call waiting tone would be the one that is currently involved in the circuit. |
| 18:13.20 | *** join/#asterisk yannj_fr (n=yannj@vpn.intelunix.fr) |
| 18:13.51 | Zeeek | SuPrSluG depends on whether they are behind NAT |
| 18:14.41 | SuPrSluG | behind nat. so yes. had one way audio problem on a transfer |
| 18:14.44 | Katty | AHH |
| 18:15.03 | Zeeek | {{Katty}} |
| 18:15.13 | carrar | spinull, you could get a dozen analog lines and have the phone company roll them over to the next POTS line ;) |
| 18:15.15 | Zeeek | bye |
| 18:16.35 | phillipk | Well, that was rather poorly worded, but the point is, you can't do that with a single analog line. |
| 18:16.35 | spinull | carrar, thats the problem though, we have 4 lines right now and it costs alot |
| 18:16.35 | *** join/#asterisk neverblue (n=neverblu@unaffiliated/neverblue) |
| 18:16.36 | carrar | thats the cost of doing biz |
| 18:16.55 | carrar | How much are you paying for 4 POTS lines? |
| 18:17.04 | Zeeek | OTH a voip carrier could route one numpber to your asterisk |
| 18:17.25 | Zeeek | and you could route them (assuming enough channels) |
| 18:18.22 | carrar | assuming he has the bandwidth |
| 18:18.40 | Zeeek | yes |
| 18:18.51 | carrar | and some resemblance of qos |
| 18:19.30 | phillipk | Which I would think would be more expensive than 4 POTS lines. |
| 18:19.52 | Zeeek | so much of these things are dependent on the country |
| 18:19.58 | blitzrage | qos is unnecessary if you subscribe to the thought of QoS by overprovisioning :) |
| 18:20.14 | carrar | hahah |
| 18:20.18 | Zeeek | or if you don't mind shitty calls |
| 18:20.20 | `pariah | Hello, I've set up a couple of AsteriskNOW and plain asterisk servers in the past, but my expirences with AsteriskNOW! and their complex dial plan have not been very good. I'm looking for an OS to run asterisk on that would be very minimal, could anyone recommend a distro of linux or another OS if that is a possiblilty |
| 18:20.39 | Zeeek | windows vista |
| 18:20.43 | blitzrage | CentOS |
| 18:20.46 | `pariah | Zeeek: thats funny |
| 18:21.01 | carrar | CentOS 4.4 or CentOS 5 |
| 18:21.02 | [TK]D-Fender | spinull: Can you (with nice normal phones) be on 2 SEPARATE calls at once on a SINGLE line? |
| 18:21.12 | *** join/#asterisk bkw__ (n=brian@ppp-70-128-114-89.dsl.tulsok.swbell.net) |
| 18:21.30 | `pariah | So CentOS seems to be the distro of choice for a minimal *JUST ASTERISK* server? |
| 18:21.56 | carrar | pariah, depends who you ask, but it works well for me |
| 18:21.59 | [TK]D-Fender | `pariah: its easy to get everything you need for * running on it, its well documented. |
| 18:22.14 | Zeeek | commonly recommended |
| 18:22.49 | [TK]D-Fender | `pariah: You can use whatever you feel like as long as *'s reuirements are met. Slackware is great for minimal installs as well. |
| 18:23.52 | `pariah | The server is a Dual PIII 600 mHz with 1 gig ECC and SCSI drives in RAID 1 |
| 18:24.30 | `pariah | there will be about 10 VoIP phones max, with 2 FXO cards to interface with a panasonic KSU |
| 18:25.05 | `pariah | The server will run ONLY asterisk, should this beast do the job with out load issues? |
| 18:25.27 | `pariah | correction the processors are PIII 500 mhz x 2 |
| 18:25.50 | *** part/#asterisk TimRiker (n=timr@pdpc/supporter/bronze/TimRiker) |
| 18:26.20 | [TK]D-Fender | `pariah: Should be plenty without transcoding |
| 18:26.30 | yannj_fr | is it sip signalling that load the CPU? |
| 18:26.36 | [TK]D-Fender | `pariah: Make sure the PCI spec is right |
| 18:26.48 | [TK]D-Fender | yannj_fr: No, SIP = 0 load |
| 18:27.06 | `pariah | [TK]D-Fender: i have all of the IRQ issues resolved and already have the interfaces installed, each on their own IRQ |
| 18:27.18 | [TK]D-Fender | `pariah: Should be good to go then. |
| 18:27.29 | `pariah | [TK]D-Fender: what is this transcoding? |
| 18:27.33 | yannj_fr | so if you only have call between phone using the same codec |
| 18:27.39 | yannj_fr | the load is 0? |
| 18:27.46 | [TK]D-Fender | `pariah: Translating from one codec to another. |
| 18:27.59 | vader-- | hmmm i have a friend who's work is having voip issues they have a main location with 2 x T1 doing voip through the internet (PATECH), and they have 8 remote locations that are on DSL with 1-2 phones that are VOIP at them |
| 18:28.23 | [TK]D-Fender | yannj_fr: Or close enough. But then again that is RTP, not SIP. I was a "loaded question", and I answered it as such :) |
| 18:28.35 | `pariah | [TK]D-Fender: do you recommend CentOS 4.4 or 5? |
| 18:29.14 | [TK]D-Fender | `pariah: CentOS 4.4 is well established, so I'd go with that for now. |
| 18:30.10 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 18:31.01 | vader-- | any good tools for diagnosing the connections and see if there is any issues ? |
| 18:31.14 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 18:31.17 | `pariah | i can only find iso's for 4.5 |
| 18:31.20 | yannj_fr | I am build a big network with asterisk, 2 big sites, call between sites are in g729 from end to end, and in a site in g711, I provided mailbox and musiconhold, each site register 200 phones and the server will be Xeon dual core 2Ghz with 4GB ram, what do you think of that? |
| 18:31.38 | nowork | hi, can we chage the ringtone into music?? |
| 18:32.17 | *** join/#asterisk pipwerk (i=pip@ringbreak.dnd.utwente.nl) |
| 18:32.33 | *** join/#asterisk dahunter3 (n=dahunter@static-71-189-121-177.lsanca.dsl-w.verizon.net) |
| 18:32.39 | [TK]D-Fender | vader--: You haven't described the actual problem, don't go asking for solutions... |
| 18:33.01 | [TK]D-Fender | nowork: Depends on the phone. |
| 18:33.23 | nowork | TK: any detail info? |
| 18:33.26 | vader-- | tkd im not i was wondering if there is a suite of tools that i could use for doing speed tests at their locations |
| 18:33.57 | [TK]D-Fender | vader--: www.dslreports.com has a bunch if you want to test bandwidth, etc |
| 18:37.44 | vader-- | im wondering if their main location's internet connection is too weak to handle all those branch offices |
| 18:39.26 | [TK]D-Fender | nowork: maybe YOU should be giving some details as to exactly what you have and what you want to do with it. You are being incredibly vague. |
| 18:40.53 | *** join/#asterisk PioneerVM2 (n=IceChat7@ool-45779466.dyn.optonline.net) |
| 18:41.11 | De_Mon | AddQueueMember(elephant-${EXTEN:1},${CUT(CHANNEL,-,-1)},,j) |
| 18:41.26 | De_Mon | Shouldn't this line jump to n+101 if the member exists? |
| 18:41.33 | *** join/#asterisk andyd (n=andyd@213-228-240-161.dsl.prodigynet.co.uk) |
| 18:41.59 | *** join/#asterisk elg (n=fugalh@azerial.fastwave.biz) |
| 18:42.55 | PioneerVM2 | hi -- im having a problem setting caller ID with AGNVoip.com -- they say: "You need to change your Contact info: It is now Contact: <sip:123456789@192.168.1.1> but it needs to be YOURDID@IP, not AGN_SID@IP. SIDE is to be used only in case of Digest AUth and SIP REgistration. You are now using IP auth -- this is why our system is overwriting your ANI." Does this make sense to anyone? |
| 18:43.08 | [TK]D-Fender | vader--: Consider if a remote site uses an ITSP call, thats 2 connections through head office. Also consider that most DSL providers are cheaper with their upstream than they have to be. |
| 18:43.35 | vader-- | ya |
| 18:45.10 | *** join/#asterisk corpcomp (n=corpcomp@125-238-151-144.broadband-telecom.global-gateway.net.nz) |
| 18:45.32 | PioneerVM2 | nevermind, found the problem "fromuser=" was set and it had to be removed |
| 18:45.52 | *** join/#asterisk c6vette (n=khagan@wsip-70-184-112-47.ph.ph.cox.net) |
| 18:49.01 | [TK]D-Fender | De_Mon: Looks about right |
| 18:51.53 | corpcomp | Been trying to get Asterisk going for months now. Totally feed up. Would someone be so kind as to point me to a tutorial that works, or at least something a newby can understand. The issue is that I can't even get Asterisk installed in any way shape or form without substantial errors. I don't care what OS, it is going onto a Blade Server. |
| 18:52.59 | [TK]D-Fender | corpcomp: CentOS 4.4 : |
| 18:53.01 | [TK]D-Fender | ~book |
| 18:53.25 | jbot | somebody said book was a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 18:53.26 | *** join/#asterisk voipnet-tech (n=voipnet-@rrcs-24-97-250-50.nys.biz.rr.com) |
| 18:53.31 | voipnet-tech | morning |
| 18:53.38 | voipnet-tech | that's not a bad book |
| 18:54.15 | blitzrage | voipnet-tech: thx! |
| 18:54.27 | corpcomp | [TK]D-Fender Tried that. |
| 18:54.40 | blitzrage | corpcomp: current version is for 1.2, not 1.4 |
| 18:54.47 | [TK]D-Fender | corpcomp: Perhaps you could provide some details about your problems... |
| 18:54.49 | blitzrage | 2nd edition will be out in a month or two which will cover 1.4 |
| 18:58.01 | *** join/#asterisk BSD_Tech (n=BSDTech@adsl-69-230-174-37.dsl.irvnca.pacbell.net) |
| 18:58.27 | BSD_Tech | ok any good coders with php/perl in her eto convert 2 agis from php to perl |
| 18:58.52 | *** join/#asterisk gardo (n=gardo@121.97.197.103) |
| 18:59.45 | corpcomp | I have tried the following OS's: FC, CentOS, Deb, Unbuntu, Slackware. My biggest problem is that I obviously don't have enough knowledge on any distro of Linux to desipher errors. Hence I just get frustrated. I am attempting ver 1.2 If I can just find a howto that actually works that would be great. I had been using TB up until now. |
| 19:00.15 | BSD_Tech | corp whats your problem |
| 19:00.45 | BSD_Tech | try asterisknow |
| 19:00.54 | [TK]D-Fender | BSD_Tech: This is for a blade server |
| 19:01.06 | BSD_Tech | ahh |
| 19:01.09 | BSD_Tech | the joys |
| 19:01.16 | [TK]D-Fender | corpcomp: What what step do things start going bad during? |
| 19:01.32 | hypa7ia | corpcomp: is it installing the distro, or compiling asterisk? |
| 19:03.29 | corpcomp | BSD_Tech: zaptel won't compile without errors is the first mission, been working on that one for 3 weeks now. I have looked @ Anow and absolutely hated it and I need root access to the server. |
| 19:03.57 | corpcomp | hypa7ia: I can install the distro, it is asterisk that is my mission |
| 19:04.20 | BSD_Tech | you have the kernel src installed in /usr/src right |
| 19:04.36 | corpcomp | hypa7ia: Yes |
| 19:04.37 | [TK]D-Fender | corpcomp: since you can't give real details we won't be able to help unless you give someone SSH access to it to try |
| 19:05.11 | hypa7ia | corpcomp: have you tried just using the packages for your distro? |
| 19:05.23 | *** join/#asterisk antonyo14 (n=oper@206.135.21.162) |
| 19:05.45 | *** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl) |
| 19:05.59 | BSD_Tech | atrpms are crap |
| 19:06.04 | corpcomp | I will go away and setup a bare Centos box and start from the beginning again and come back as soon as I hit the first error. |
| 19:06.45 | hypa7ia | good plan corpcomp |
| 19:06.57 | hypa7ia | BSD_Tech: good to know :) |
| 19:07.14 | corpcomp | TY for the comments everyone |
| 19:07.16 | *** part/#asterisk antonyo14 (n=oper@206.135.21.162) |
| 19:07.40 | BSD_Tech | bbiab DR apt |
| 19:08.26 | *** join/#asterisk antonyo14 (n=oper@206.135.21.162) |
| 19:09.32 | antonyo14 | i know there is a channel for asterisk-gui but no one is alive, but I've got a relatively small question i was wondering if anyone could answer? |
| 19:09.40 | *** join/#asterisk n00dle (n=ccraft@hillel.springsips.com) |
| 19:10.20 | c6vette | antonyo14, we wont know until you ask... :) |
| 19:10.25 | antonyo14 | ok :) |
| 19:10.33 | antonyo14 | i configured the gui and it works great |
| 19:10.40 | antonyo14 | but it will not see my already created dialplan |
| 19:10.49 | robin_sz | coo. I had a kernel panic on my Debian stable box today .. first kernel panic in .. well, quite some years |
| 19:10.55 | antonyo14 | is there a config file i can edit to point it to look at my dialplan? |
| 19:11.08 | *** part/#asterisk c6vette (n=khagan@wsip-70-184-112-47.ph.ph.cox.net) |
| 19:13.56 | n00dle | Question: Besides /var/log/asterisk/messages and a level20 CLI verbosity, what else can I do to try to debug my SLA issues? |
| 19:15.07 | MrWup | with asterisk, is there any way to tell when a call has been transfered? either blind or attended? |
| 19:15.36 | MrWup | and carry out a command only when there has been a successful transfer? |
| 19:15.42 | [TK]D-Fender | MrWup: Blid yes, there is a variable set for that. Attended I'm not sure... |
| 19:16.19 | MrWup | asterisk is severely limited for call centres if you cant do something when a successful transfer has happened |
| 19:17.02 | [TK]D-Fender | MrWup: I presume you are referring to having Queue members considered busy even though they have transferred off the call they were on. Correct? |
| 19:17.43 | antonyo14 | would I have to create a new dialplan with the * GUI or is it possible to use my already configured one? |
| 19:17.47 | MrWup | well, my problem is that i have my own logging system for calls |
| 19:18.01 | MrWup | and i need to be able to tell when one person has transfered to another |
| 19:18.32 | MrWup | say for example, PSTN -> * -> SIP1 -> SIP2 -> SIP3 |
| 19:18.47 | MrWup | at the moment all i can do is find the time that the call was first answered |
| 19:18.54 | MrWup | and the time that SIP3 finally hung up |
| 19:19.12 | [TK]D-Fender | MrWup: Ok, thats CDR and tracking this sort of thing isn't what * was designed for... |
| 19:19.24 | MrWup | what was it designed for? |
| 19:19.25 | [TK]D-Fender | MrWup: You're SOL |
| 19:22.43 | *** join/#asterisk nitrogen (n=nitrogen@nitrogen.dsl.xmission.com) |
| 19:23.47 | nitrogen | Hi, does anybody know if Linux-supported Skype USB adapters such as the Yealink B2K and B3G can be used with Asterisk? |
| 19:25.16 | [TK]D-Fender | nitrogen: You already own this? |
| 19:26.28 | nitrogen | No, I'm considering purchasing one or two |
| 19:26.28 | *** join/#asterisk kclaussen (n=kclausse@204.13.224.242) |
| 19:26.40 | nitrogen | since I don't need 8 lines, and the Digium USB adapter is almost 10x the price |
| 19:26.41 | [TK]D-Fender | nitrogen: Good. DON'T. |
| 19:26.46 | denke | nitrogen: I think theesea are a "soundcard" .... some sort of.... Linux may be treet it as... or not |
| 19:27.01 | denke | I too do not recommend it. |
| 19:27.02 | [TK]D-Fender | nitrogen: Digium discontinued their USB adapter AGES ago. |
| 19:27.20 | [TK]D-Fender | nitrogen: Buy a proper ATA like the Linksys SPA-2102 for about $70 for 2 ports. |
| 19:27.35 | nitrogen | well, I have a system with no PCI slots (a laptop), and was hoping for a USB solution |
| 19:27.38 | [TK]D-Fender | nitrogen: And forget about USB crap. |
| 19:27.46 | nitrogen | [TK]D-Fender, ok |
| 19:28.04 | nitrogen | why? |
| 19:28.11 | [TK]D-Fender | nitrogen: http://www.telephonydepot.com/product_p/105-054-212.htm |
| 19:28.57 | [TK]D-Fender | nitrogen: Because there IS nothing useful and affordable via USB. You have also clearly missed the poitn of * and VoIP if you always thing you have to plug your phone directly into your server. |
| 19:29.32 | [TK]D-Fender | think* |
| 19:29.32 | nitrogen | no, I'm aware of the SIP-enabled ATAs |
| 19:29.33 | [TK]D-Fender | nitrogen: Good. That is what you should be aiming for. |
| 19:29.57 | [TK]D-Fender | nitrogen: Cheap, effective, and versatile. |
| 19:29.59 | nitrogen | I was looking at the USB adapters because I have no need to separate the phone connection from the computer for this particular case |
| 19:30.21 | [TK]D-Fender | nitrogen: No, there is no Zaptel driver to support those things. |
| 19:30.40 | nitrogen | Okay, thanks for the advice. |
| 19:30.57 | denke | [TK]D-Fender: I need an advice. Can you help me? |
| 19:31.04 | denke | the point is: |
| 19:31.09 | [TK]D-Fender | nitrogen: There IS 1 USB solution out there, but its a pricey multi-port moster. |
| 19:31.25 | [TK]D-Fender | monster |
| 19:32.15 | denke | I need to create a context which gives you a secondary dialtone, but if i dial the first digit, it stops playing the tone. |
| 19:32.16 | nitrogen | yeah, this just needs to be a portable demo with support for two analog phones, one PSTN line, and voip |
| 19:32.16 | phillipk | nitrogen: If it's a laptop that you carry around with you, why not use a softphone? One less piece of equipment to haul around. |
| 19:32.18 | [TK]D-Fender | nitrogen: If you're feeling masichistic : http://www.xorcom.com/products/astribank_8 |
| 19:32.38 | nitrogen | [TK]D-Fender, ah, that's probably what I was looking at, not the Digium |
| 19:32.48 | [TK]D-Fender | nitrogen: Then bring a small switch, an SPA-3102 and SPA-2102 |
| 19:33.08 | [TK]D-Fender | nitrogen: Digium's S101 has been disco'd for years now. |
| 19:33.16 | nitrogen | phillipk, The laptop is acting as a portable PBX, not just for personal calls, but thanks |
| 19:33.44 | [TK]D-Fender | nitrogen: And While others may vouch for it, I'd never even demo off an Astribank let alone USE (or rather BUY) one . |
| 19:34.23 | nitrogen | [TK]D-Fender, averse to USB in a production environment? |
| 19:34.35 | nitrogen | [TK]D-Fender, I'll agree that Ethernet is more trustworthy |
| 19:35.26 | [TK]D-Fender | nitrogen: There is defiantely taht point, but my SERIOUS problem is that its 100% tied to the host system which ahs to be your * server an is only useful with *. I like agnostic hardware that works with whatever software/hardware I tell it to. |
| 19:35.37 | MrWup | whats the best PBX software for things like CDR? |
| 19:35.40 | Qwell[] | and now it's time for my random question of the day... |
| 19:35.43 | [TK]D-Fender | nitrogen: I refuse to buy gear that thinks it'll OWN ME. |
| 19:35.47 | Qwell[] | What is the different between (R) and (TM) |
| 19:36.32 | [TK]D-Fender | Qwell : "F" ;) |
| 19:36.39 | n00dle | nitrogen: I love my sipura and hate my S101U |
| 19:37.06 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 19:37.08 | [TK]D-Fender | I'm interested in seeing this when it ships : http://www.digiumcards.com/linksys_spa8000 |
| 19:37.10 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 19:37.43 | denke | [TK]D-Fender: Can you give me some advice? I need to create a context which gives you a secondary dialtone, but if i dial the first digit, it stops playing the tone. |
| 19:38.14 | nitrogen | [TK]D-Fender, Would you prefer a Linksys over one of the cheaper ATAs then? I'd just like to keep my budget as low as possible. |
| 19:38.44 | MrWup | denke, i need the same thing... currently my dialtone carries on even after pressing the first digit |
| 19:39.10 | [TK]D-Fender | denke: 2 options : Disa, but I don't think it'll respect "ignorepat", or wase 1 FXO & 1 FXS on a TDM card and loop out the FXO into the FXS as a Zaptel phone. |
| 19:39.10 | denke | MrWup that's my problem too |
| 19:39.36 | denke | pf! :) |
| 19:39.41 | [TK]D-Fender | denke: But seriously qhy do you want to keep playing tone after digits are received? |
| 19:39.57 | denke | i do not :) |
| 19:40.11 | denke | the problem is: |
| 19:40.16 | denke | if I use disa |
| 19:40.18 | MrWup | is digium's Q2 any good at CDR? |
| 19:40.32 | denke | it inmediatley starts the CDR counter |
| 19:40.54 | denke | but If I use a context and s,1,Playtones.... |
| 19:41.09 | denke | than I can not stop plazing it after the first digit |
| 19:41.31 | denke | y |
| 19:42.07 | MrWup | one solution would be... |
| 19:42.16 | MrWup | record some dialtone in a gsm file or a .al file or .wav file |
| 19:42.21 | MrWup | then use background to play it back |
| 19:42.26 | MrWup | and it will stop after first digit |
| 19:42.32 | MrWup | bit of a dirty way though |
| 19:43.07 | denke | I thout about that too, but it is a really bad sovation |
| 19:43.10 | [TK]D-Fender | denke: Did I even MENTION playtones? No. |
| 19:43.12 | MrWup | yeah |
| 19:43.23 | [TK]D-Fender | denke: I just gave you your 2 solutions. |
| 19:43.25 | denke | true! |
| 19:43.49 | [TK]D-Fender | denke: If you think you've discovered a third that will work, more power to you... |
| 19:44.14 | denke | The first solution will win somehow... (DISA) |
| 19:45.03 | denke | the external loop is .... BLJAC! :S |
| 19:45.09 | denke | thanx for the help :) |
| 19:47.24 | *** join/#asterisk elg (n=fugalh@216.31.27.110) |
| 19:47.24 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 19:49.18 | *** join/#asterisk boster (n=boster@38.98.147.68) |
| 19:50.44 | boster | is there a way to get a sip peer status in extensiosn.conf and take actions based on the status of the peer? Short of writing an agi script that calls asterisk -rx "show sip peer foo", that is |
| 19:50.46 | *** join/#asterisk dalfry (n=dalfry@70.89.177.109) |
| 19:51.47 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 19:53.49 | *** join/#asterisk DaveCanoe (n=Dave@66.96.30.6) |
| 19:55.15 | dasuberdavid | boster: you can use app_system |
| 19:57.55 | [TK]D-Fender | boster: Nope, thats about it. |
| 19:59.18 | boster | thanks |
| 20:02.57 | neverblue | do you have to use a USB headset for twinkle? |
| 20:03.06 | neverblue | i am not getting anything from the mic |
| 20:07.25 | *** part/#asterisk nitrogen (n=nitrogen@nitrogen.dsl.xmission.com) |
| 20:08.10 | *** join/#asterisk WeBRainstorm (n=ask@81-174-25-14.f5.ngi.it) |
| 20:08.12 | WeBRainstorm | hi all |
| 20:10.00 | *** join/#asterisk hfb (n=hfb@pool-72-87-254-188.lsanca.dsl-w.verizon.net) |
| 20:12.17 | MrWup | can anyone recommend some good PBX software which handles things like tracking across transfers and proper CDR better than asterisk? |
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| 20:17.35 | *** part/#asterisk BSD_Tech[laptop] (n=bsdtech@adsl-69-230-174-37.dsl.irvnca.pacbell.net) |
| 20:25.55 | ber111 | in asterisk extensions.conf does it always match from most specific match to least specific |
| 20:26.04 | ber111 | (within the same context) |
| 20:31.11 | *** join/#asterisk sasch (n=sasch@83.225.133.201) |
| 20:36.01 | *** join/#asterisk hotjava (n=wmirc_us@pool-71-107-107-231.lsanca.dsl-w.verizon.net) |
| 20:42.27 | *** join/#asterisk Shaun2222 (n=shaun@ip68-4-212-221.oc.oc.cox.net) |
| 20:42.36 | Shaun2222 | whats the usb port for on the polycom 650's |
| 20:42.54 | *** join/#asterisk asymptote (n=weldon@phobos.asee.org) |
| 20:42.55 | nowork | hi, how can I know my OpenH323 version ? |
| 20:43.03 | nowork | pwlib |
| 20:44.57 | *** join/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net) |
| 20:45.23 | *** join/#asterisk EricL (n=eric@74.9.83.194) |
| 20:46.06 | EricL | Does anyone have any idea why Cisco 7961 phones don't send back the SIP ACK and if there is anything I can do in Asterisk to compensate for it? |
| 20:47.30 | De_Mon | http://pastebin.ca/526505 |
| 20:47.50 | De_Mon | my AddQueueMember isn't jumpping |
| 20:48.36 | De_Mon | ohh... I bet I need to do n+100 to get last n+101 |
| 20:48.56 | Corydon76-home | It's not supposed to jump |
| 20:49.13 | Corydon76-home | Jumping is deprecated behavior |
| 20:49.24 | De_Mon | O_o |
| 20:49.39 | SwK | hah |
| 20:49.41 | De_Mon | so the j option is going byebye eh |
| 20:49.50 | De_Mon | jumping in general is depriciated? |
| 20:49.53 | Corydon76-home | Correct |
| 20:50.11 | *** join/#asterisk antonyo14 (n=oper@206.135.21.162) |
| 20:50.22 | De_Mon | hrm.. so I should check the status of addqueuemember and respond accordingly then |
| 20:50.31 | Corydon76-home | We use status variables now, for better control and more possible return values |
| 20:51.04 | SwK | De_Mon, check the AQMSTATUS variable... and use show application AddQueueMember to see what the values are... |
| 20:51.45 | De_Mon | I saw that, and decided using jump was easier ;) |
| 20:51.59 | Corydon76-home | I don't see where you ever got a j option for AddQueueMember anyway. It was never there. |
| 20:52.03 | [hC] | So, there seems to be an issue with calling a macro from Dial() using M(macroname) - You need to return MACRO_RESULT in order to continue or terminate the call. I'm doing a call screening type thing where your options are push 1 to accept, or hangup to reject. The problem is if I press "1", or hangup requiring sending the call to 'h,' it is not interpreted in the macro. It exits the macro and tries those exten's in the originating context. (Which of |
| 20:52.03 | [hC] | course is too late now to set MACRO_RESULT properly) |
| 20:52.03 | voipnet-tech | EricL, 7961's should send ACK after recieving 200-3 |
| 20:52.20 | [hC] | Is that a bug thats been fixed? I'm using 1.2.15 |
| 20:53.21 | voipnet-tech | EricL, perhaps the phone isn't recognizing the 200 message from *, so it's not ACK'ing |
| 20:53.23 | Corydon76-home | [hC]: not that I'm aware of |
| 20:53.48 | [hC] | Corydon76-home: As I see it, that makes the use of MACRO_RESULT as it was intended impossible. That would be a bug, no? |
| 20:54.40 | Corydon76-home | [hC]: <shrug> file a bug, if you like |
| 20:54.40 | *** join/#asterisk codestr0m (n=asura@207.135.120.85) |
| 20:55.29 | voipnet-tech | EricL, may need to tell the phone to look for a different 200 message, something like this: Vendor-Specific-9-1 = "sip-status-code=200" |
| 20:55.56 | *** join/#asterisk ltd (i=foobar@nox.amused.net) |
| 20:56.02 | codestr0m | Instead of doing.. exten => XXXXXXXXX,1,Answer() is there another way to handle this so that you can send 183 early media and a message like "please wait while I connect your party" or ringback? |
| 20:56.35 | Corydon76-home | codestr0m: Ringing() |
| 20:57.08 | *** join/#asterisk iCEBrkr (i=icebrkr@cyberdyne.org) |
| 20:57.33 | codestr0m | Corydon76-home: that'll provide early media.. and is it possible to set what hte media is? (going to look.) Thanks.. |
| 20:58.22 | Corydon76-home | I think that sends a 180, actually |
| 20:58.45 | codestr0m | I don't know sip so well ot know the difference.. 180 vs 183.. hmm.. |
| 20:59.16 | *** join/#asterisk codazoda (n=Joel_Dar@mail.hurdmanivr.com) |
| 20:59.17 | codestr0m | one is probably just ringback and the other early media I assume. there being some small difference |
| 21:00.31 | *** join/#asterisk scurb (n=scurb@c-25aae355.14-16-64736c13.cust.bredbandsbolaget.se) |
| 21:00.56 | codazoda | ViaTalk provides 4 trunks for my 4 lines of service. But, I'd prefer to have them all in a single trunk (single username and password) so that I can use the GUI's I've tried (which all support only a single outgoing trunk). Any suggestions for a VOIP provider that provides a single account for their multiple lines? |
| 21:01.12 | *** join/#asterisk kiscokid (n=ron@208.106.33.66) |
| 21:02.23 | blitzrage | codazoda: Mix Networks (disclaimer: I run the backend at Mix) |
| 21:03.06 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 21:03.09 | kiscokid | Trying to get Meetme to work but menuselect says app_meetme depends on zaptel(E) but I built and installed zaptel |
| 21:03.24 | blitzrage | single registration, single web interface for all trunks/users |
| 21:06.26 | [TK]D-Fender | kiscokid, Got a PCI timing source? Otherwise you need to make sure that ZTDUMMY was compiled and active before starting *, and you need to build * AFTER Zaptel to have it integrate |
| 21:06.29 | Katty | ahhhhhhhh |
| 21:06.29 | Katty | so |
| 21:06.30 | Katty | tell me |
| 21:06.34 | Katty | who's incharge of cluecon stuffs |
| 21:06.37 | Katty | and by stuffs i mean billing |
| 21:06.42 | Katty | or registering |
| 21:06.44 | Katty | or whatever |
| 21:06.47 | *** part/#asterisk codestr0m (n=asura@207.135.120.85) |
| 21:07.32 | kiscokid | Fender: I have a 2.6 kernel so supposedly I don't need a timing device |
| 21:08.59 | kiscokid | Fender: I built * before zaptel because I didn't have any cards to support |
| 21:09.39 | kiscokid | can I just rebuild * now? |
| 21:09.54 | Corydon76-home | kiscokid: yes |
| 21:10.08 | kiscokid | and do I need to do a make followed by a make install? |
| 21:10.14 | [TK]D-Fender | kiscokid, If you DID build Zaptel w/ ZTDUMMY support, then yes, you simply need to recompile & install * |
| 21:10.33 | [TK]D-Fender | kiscokid, Star w/ make clean then menuselect jsut in case |
| 21:10.33 | EricL | voipnet-tech:How would I go about doing that? |
| 21:10.34 | *** join/#asterisk BSD_Tech[laptop] (n=bsdtech@adsl-69-230-174-37.dsl.irvnca.pacbell.net) |
| 21:10.49 | [TK]D-Fender | kiscokid, But you can do it in-place with exist source extraction. |
| 21:10.52 | kiscokid | how can I tell if I have ZTDUMMY support? |
| 21:11.01 | [TK]D-Fender | kiscokid, its part of menuconfig |
| 21:11.18 | Corydon76-home | menuselect |
| 21:11.44 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 21:12.07 | blitzrage | I think menuconfig actually works too though :) |
| 21:12.24 | Corydon76-home | Does it? |
| 21:12.36 | blitzrage | I think Russell put it in as a synonym because someone complained |
| 21:12.45 | Corydon76-home | So it does... |
| 21:12.50 | [TK]D-Fender | blitzrage, I'm multi-tasking, dyslexia is to be expected ;) |
| 21:12.55 | blitzrage | +1 blitzrage :) |
| 21:12.59 | kiscokid | menuselect shows ztdummy with an [*], I assume that means it built? |
| 21:13.09 | Corydon76-home | lysdexia? |
| 21:13.12 | blitzrage | that means to build it -- not that it did |
| 21:13.30 | [TK]D-Fender | Corydon76-home, Dyslexics of the world... untie! |
| 21:13.30 | blitzrage | I put the sexy in dyslexia |
| 21:13.30 | *** join/#asterisk Greek-Boy (n=g@196.45.144.42) |
| 21:13.32 | Corydon76-home | The order keeps wronging out come? |
| 21:13.47 | [TK]D-Fender | kiscokid, More like it WILL be built |
| 21:14.01 | kiscokid | ok, so if there's an [*] in menuselect and I do a make it is built? |
| 21:14.12 | Corydon76-home | Correct |
| 21:14.12 | [TK]D-Fender | blitzrage, words are superfluous ;) |
| 21:14.17 | kiscokid | cool |
| 21:14.32 | blitzrage | [TK]D-Fender: indeed |
| 21:14.46 | [TK]D-Fender | blitzrage, I DON'T WANT TO TALK TO YOU |
| 21:14.57 | kiscokid | ok, gonna try to make and make install for * |
| 21:14.58 | blitzrage | I JUST WANT... |
| 21:14.59 | [TK]D-Fender | blitzrage, ! ! ! |
| 21:15.06 | kiscokid | should I do a make clean first? |
| 21:15.16 | [TK]D-Fender | kiscokid, advisable |
| 21:15.17 | Corydon76-home | You can, if you like |
| 21:15.28 | *** part/#asterisk codazoda (n=Joel_Dar@mail.hurdmanivr.com) |
| 21:15.34 | Corydon76-home | menuselect SHOULD remove files based on what changed, though |
| 21:15.51 | *** part/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net) |
| 21:16.20 | *** part/#asterisk BSD_Tech[laptop] (n=bsdtech@adsl-69-230-174-37.dsl.irvnca.pacbell.net) |
| 21:18.02 | blitzrage | ummm... what app (in linux) will let me continue a file from where it left off over HTTP ? |
| 21:18.13 | blitzrage | firefox isn't that smart apparently |
| 21:19.51 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 21:20.00 | Corydon76-home | wget |
| 21:20.11 | Corydon76-home | with --continue option, I think |
| 21:20.45 | blitzrage | yep -- I *just* foudn that :) |
| 21:20.49 | blitzrage | thx! |
| 21:22.49 | MindTheGap | what does DEBUG "must remove any reg pbx_config" means? should i just ignore it? |
| 21:24.49 | MindTheGap | It is followed by "check ctx" and "delete ctx" for all my contexts... |
| 21:26.38 | EricL | voipnet-tech: How can I tell it to look for a different ACK? |
| 21:27.21 | kiscokid | Ok, I built and started zaptel but app_meetme still says XXX |
| 21:27.24 | voipnet-tech | EricL,, is the phone sending a different ACK ? |
| 21:27.25 | EricL | voipnet-tech: Also, I did a tcpdump and came up with the phone not responding. |
| 21:27.33 | kiscokid | I also rebuilt * |
| 21:27.54 | voipnet-tech | EricL,, if the phone doesn't respond to the 200 with an ACK then the phone isn't understanding asterisk's 200 as a 200 |
| 21:28.05 | EricL | voipnet-tech: I don't even see it sending an ACK. |
| 21:28.18 | voipnet-tech | is asterisk sending a 200 ? |
| 21:28.22 | EricL | voipnet-tech: Sometimes it does ACK and sometimes it doesn't. |
| 21:28.29 | EricL | voipnet-tech: Yes, it does over and over. |
| 21:28.37 | EricL | Most of the time it doesn't ACK. |
| 21:29.51 | voipnet-tech | is the problem only happening with a single phone? |
| 21:29.51 | EricL | Nope, its with all the Cisco 7961Gs, but its not happening at all with my Polycomms. |
| 21:30.08 | kiscokid | Fender: are you still here? |
| 21:30.40 | voipnet-tech | my guess is that the cisco's aren't understanding the 200 message Asterisk is sending it. why...? I don't really know, likely a configuration item on the phone. I doubt it's anything wrong with asterisk |
| 21:32.30 | EricL | I doubt its something wrong with Asterisk, but I want to know if I can fix it with Asterisk. |
| 21:34.25 | voipnet-tech | have you been over all the info here: http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+79xx |
| 21:34.37 | voipnet-tech | looks like alot of info, i haven't read it all |
| 21:34.44 | irule | onyone know of a good doc on hylafax/iaxmodem/spandsp? |
| 21:35.11 | syzygyBSD | ~fax |
| 21:35.31 | jbot | Well, apperantly the fax was concieved of by Napoleon Bonaparte. He commissioned a system of devices that could transmit a traced image electrically over telegraph lines to a remote device that would redraw the image identically. |
| 21:35.32 | syzygyBSD | !fax |
| 21:35.32 | syzygyBSD | hmmm |
| 21:36.16 | *** join/#asterisk elg (n=fugalh@216.31.27.110) |
| 21:36.17 | syzygyBSD | not what I was looking for... |
| 21:36.39 | EricL | voipnet-tech: Over and over and over and over |
| 21:36.49 | *** join/#asterisk brussel (n=brussel@ppp-71-128-195-254.dsl.sndg02.pacbell.net) |
| 21:37.57 | voipnet-tech | EricL, i don't think i could tell you anthing that's not already listed on that page |
| 21:39.48 | EricL | Hmmm, I would hate to return 5 Cisco phones...that just seems wrong. |
| 21:41.29 | SwK | anyone in here have a SIP t38 hardware gateway already online w/ atleast 100 channels of spare capacity? if yes PM me |
| 21:41.30 | *** join/#asterisk tsurko (n=tsurko@77.70.24.142) |
| 21:41.30 | *** join/#asterisk pifiu-laptop (n=someone@216.5.79.1) |
| 21:41.33 | *** join/#asterisk boch (n=fran@190.48.217.225) |
| 21:42.07 | EricL | Does anyone have any idea how upgrade the 79X1 phones because the firmware that I have keeps giving me an "Auth Fail" |
| 21:42.17 | [TK]D-Fender | EricL, how many do you own? |
| 21:42.38 | EricL | [TK]D-Fender: 5. |
| 21:43.05 | boch | gentleman, do you know a call generator that allows to gen calls, when answer play some digits or stream files like a dialplan? |
| 21:43.07 | EricL | I have upgraded them to basic SIP compatibility, but I can't upgrade them to anything remotely new. |
| 21:43.17 | [TK]D-Fender | EricL, Ok, if you had more.... THEN it would be "wrong" to return only 5 ;) |
| 21:44.31 | EricL | We have 15 more on order (that they ordered assuming I would be able to get everything to work). |
| 21:46.51 | EricL | There seems to be lots of information on upgrading and getting a hold of firmware for the Cisco 79X0 series but very little on the 79X1 series. |
| 21:47.15 | *** join/#asterisk NirS (i=Nir@87.68.0.40.cable.012.net.il) |
| 21:49.37 | MrWup | with the asterisk manager interface... when it sends an event to the client which is connected to it... how do i know when the event has been sent? |
| 21:49.58 | MrWup | for example... you'd expect that it would be Event: Link ..some lines here.. EndLine |
| 21:50.05 | *** join/#asterisk ToyMan (n=Stuart@72.168.167.241) |
| 21:50.09 | MrWup | and then i can look for EndLine and know when ive received the entire event |
| 21:50.37 | MrWup | any ideas? |
| 21:51.00 | MrWup | (there is no standard EndLine for every event by the way... they all seem to be of different sizes and lengths) |
| 21:52.54 | voipnet-tech | EricL, the updates you are seeking are likely not at all like the firmware for the 7960 because the 7961's are Java-based instead |
| 21:53.15 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 21:53.19 | kiscokid | still having a problem getting app_meetme to build. |
| 21:53.45 | kiscokid | I built and started zaptel and it says no hardware so loaded ztdummy |
| 21:53.49 | EricL | voipnet-tech: I wish I knew what updates I was looking for. |
| 21:54.00 | EricL | voipnet-tech: They would be a lot easier to find that way. |
| 21:54.46 | kiscokid | voipnet-tech: do you have a cisco contract? |
| 21:55.57 | tzafrir_laptop | NirS, so did you get a chance to look at say.conf? |
| 21:56.12 | NirS | oh yes |
| 21:56.34 | NirS | I have a working verions of all the numbers in Hebrew in say.c for version 1.4.4 |
| 21:57.02 | NirS | I tried working on the dates and shit, but that got me really confused, as I don't yet have all the recordings |
| 21:57.04 | voipnet-tech | kiscokid, nope |
| 21:57.17 | NirS | however, I should be getting them soon, so I'll start working faster |
| 21:57.34 | *** join/#asterisk funnymanva (n=funnyman@c-75-75-99-240.hsd1.va.comcast.net) |
| 21:57.38 | NirS | now that I work only a single day per week for Atelis, will give me more time to get back to coding and FOSS |
| 21:57.57 | kiscokid | voipnet: maybe your supplier could get the updates from the cisco website if he has a contract |
| 21:58.35 | NirS | tzafrir ? |
| 21:58.39 | funnymanva | Anyone know if there is a compatibility issue with Asterisk Business Edition and the TDM800p cards? It shows in lspci, but it's not recognizing the quad FXO daughterboard. Could be a bad daughterboard, I'm going to swap it out tomorrow, but just curious if it's a known problem. |
| 21:59.11 | NirS | could be that the zaptel module you are using is too old ? |
| 21:59.26 | voipnet-tech | EricL, kiscokid is right, your best bet is to buy a smartnet contract from cisco, for a single phone, that way you can get the most up to date software from them |
| 21:59.42 | voipnet-tech | and you could <optionally> load the same software on all the phones |
| 21:59.59 | NirS | the TDM800 has a specific kernel module, so if you are using stuff older than zaptel 1.2.3, I think it won't work |
| 22:00.20 | NirS | sorry |
| 22:00.21 | NirS | 1.2.16 |
| 22:00.55 | voipnet-tech | if i remembered my tech-data login i could probably give you a price for a smartnet. my company is actually listed as a cisco reseller but we so rarely sell cisco, and when we do it's for routers not phones |
| 22:00.56 | funnymanva | i'll have to check the version, but I'm using what came with the ABE cd's. This is my first attempt at setting up an ABE system and not just building everything from SVN. |
| 22:01.49 | dalfry | hello, I need help with chan_sip's channel limit |
| 22:02.34 | dalfry | I am hitting an asterisk box with simultaneous sip calls using sipp and asterisk starts declining calls after the 99th channel has been used up |
| 22:02.42 | dalfry | how do I increase this limit? |
| 22:02.52 | NirS | funny, ABE may be older than the one supporting TDM800 |
| 22:03.23 | NirS | dalfry, that may not be a SIP limitation, but maybe a hardware limitation |
| 22:03.29 | NirS | what hardware are you using ? |
| 22:03.52 | dalfry | its a dual opteron with no zap / special voip hardware. its supposed to be a plain sip / agi box |
| 22:04.07 | NirS | hmmm... that's weird |
| 22:04.13 | NirS | are you activating an AGI in there |
| 22:04.14 | NirS | ? |
| 22:04.18 | dalfry | and it seems to be a config limit coz the limit is *always* 99 calls |
| 22:04.35 | dalfry | yes I am but all the agi does is to play a sound file and wait for it to finish |
| 22:04.43 | NirS | if you can give me access to the box, I can take a look for a minute |
| 22:04.56 | dalfry | I can't its on our private lan |
| 22:05.11 | NirS | is the playback performed from the dialplan or from the AGI ? |
| 22:05.16 | dalfry | from the agi |
| 22:05.17 | carrar | ssh tunnel! |
| 22:05.21 | EricL | voipnet-tech: I think I might just get a smartnet contract. |
| 22:05.27 | NirS | what is your AGI language ? |
| 22:05.31 | dalfry | phpagi |
| 22:05.45 | NirS | ok, then 99 is a possible limit for your environment |
| 22:05.46 | EricL | I have the firmware for 8.0.4SR1 loaded on all the phones. |
| 22:05.56 | dalfry | why is that? |
| 22:06.06 | NirS | you see, according to what you say, you will have 99 active PHP scripts running on your system |
| 22:06.14 | dalfry | yes, so? |
| 22:06.17 | EricL | I don't actually know what the newest version is though. |
| 22:06.27 | NirS | that means that Asterisk had been forked 99 times in order to perform the AGI and Playback |
| 22:06.40 | dalfry | why isn't that limit 95 or 101 and why specifically 99? |
| 22:06.42 | NirS | I think you are hitting a top limit on the machine environment |
| 22:06.52 | NirS | well, that I don't know |
| 22:07.04 | NirS | but I can suggest a test to see if I'm right |
| 22:07.07 | dalfry | exactly :) thats what prompts me to think that its a config limit |
| 22:07.11 | dalfry | tell me pls |
| 22:07.18 | *** part/#asterisk Defraz (n=t0tal@fw.fuzecore.com) |
| 22:07.28 | NirS | try to remove your AGI from the dialplan and simply dial a static file, then run the test again |
| 22:07.34 | tzafrir_laptop | NirS, here |
| 22:07.36 | dalfry | checking |
| 22:07.50 | NirS | I've run a similar test on a P4 machine about a year ago, and I got 220 active channels on g711 easy |
| 22:07.55 | NirS | what codec are you using ? |
| 22:08.04 | NirS | tzafrir, did you see my response ? |
| 22:08.21 | *** part/#asterisk elg (n=fugalh@216.31.27.110) |
| 22:08.38 | tzafrir_laptop | yes. But I was really hoping to get say.conf working. So non-C coders could have their say |
| 22:09.43 | NirS | say.conf ? |
| 22:09.43 | NirS | hold on |
| 22:09.52 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 22:10.03 | tzafrir_laptop | See also http://lists.digium.com/pipermail/asterisk-dev/2007-May/027857.html |
| 22:10.17 | NirS | hmmm... I just noticed that |
| 22:10.23 | NirS | that can be really cool to work with |
| 22:10.29 | NirS | I'll have to take a look into that one |
| 22:10.33 | Ryushin | Is there anything special I need to know in order to get twinkle to work with asterisk? |
| 22:10.56 | tzafrir_laptop | Ryushin, use "asterisk" as the realm? |
| 22:11.00 | NirS | dalfry, are you doing the test now ? |
| 22:11.11 | dalfry | yes |
| 22:11.57 | Ryushin | tzafrir_laptop: I have a Asterisk server at work, I've VPN'd into the network and I'm trying to set up a softphone to connect the asterisk server. |
| 22:12.08 | Ryushin | The polycom phones connect fine. |
| 22:12.11 | kiscokid | seems like make distclean solved my problem |
| 22:12.27 | NirS | tzafrir, you and I need to get together for a day or two and see what we can create on this hebrew stuff |
| 22:12.32 | tzafrir_laptop | the problem is that the syntax of say.conf seems to offer no support for the "options" field, which is essentially used for explicit gender |
| 22:12.40 | NirS | Noam should have all the recordings ready for me soon |
| 22:13.06 | Ryushin | I can get twinkle to register with the server. But it always times out when making a outgoing call and I don't see anything in asterisks log that I'm even making the call. |
| 22:13.08 | tzafrir_laptop | That's great. Anyway, until say.conf is ready, say.c should be used |
| 22:13.18 | NirS | well, that's what I used |
| 22:13.29 | NirS | I have all the numbers working nicely |
| 22:13.32 | NirS | you wanna test it ? |
| 22:13.46 | tzafrir_laptop | yes, please |
| 22:15.17 | NirS | hold on |
| 22:17.19 | nowork | hi, my sip provider is using Mera, he want me to treat him as a gateway..any context examples you guys hv? thanks |
| 22:19.12 | NirS | just a minute |
| 22:19.15 | NirS | need to setup here |
| 22:19.37 | *** part/#asterisk kiscokid (n=ron@208.106.33.66) |
| 22:20.33 | Greek-Boy | i'm surprised to see that a lot of people/providers are still using SIP for trunking |
| 22:21.00 | dalfry | NirS: it seems to be working now but why is there a problem with AGI? |
| 22:21.06 | nowork | Greek-Boy, so, what is in use now? |
| 22:21.11 | boch | do you know why my * says 16 active channels, 29 active calls, and lists only 12 channelss ? |
| 22:21.12 | NirS | simply |
| 22:21.17 | dalfry | and why specifically 99? |
| 22:21.28 | NirS | put, the number of AGI scripts and forks of asterisk you have running on your box is too high |
| 22:21.47 | NirS | it may an internal limitation to protect asterisk from crashing, or just a limitation of your memory |
| 22:22.14 | NirS | suggestion, choose the file to playback via the AGI, then return it to the dialplan via a variable |
| 22:22.15 | dalfry | well I am watching the memory / cpu usage and cpu is barely being scratched. memory usage is not enormous too |
| 22:22.25 | Greek-Boy | nowork: IAX is better for turnking |
| 22:22.26 | NirS | then simply playback the file using that variable |
| 22:22.38 | dalfry | NirS: Can't do that. the php script has complex business logic. can't fork back and forth |
| 22:22.40 | Ryushin | Is the asterisk server a proxy for my softphone to call outbound lines? |
| 22:22.46 | NirS | then I suggest that you dig up app_agi and see what's in ther |
| 22:22.58 | NirS | dalfry, nothing is impossible |
| 22:23.05 | NirS | it's just a matter of balancing it |
| 22:23.06 | dalfry | hmmm true |
| 22:23.18 | NirS | trust me, I've been there, and I always found a way to do it |
| 22:23.31 | dalfry | NirS: thanks a ton |
| 22:23.37 | NirS | I've written hundreds of AGI systems with PHPAGI, none of them had this problem |
| 22:23.40 | NirS | you're welcome |
| 22:23.58 | dalfry | what I still can't understand is the exact 99 number limit :) |
| 22:24.05 | NirS | if you want consulting or for me analyse your system, I'm available for consulting, you can contact via e-mail at nirs@greenfieldtech.net |
| 22:24.09 | dalfry | if it was resource related, it would be different always |
| 22:24.18 | dalfry | will do. thanks |
| 22:24.23 | NirS | unless it's hard coded somewhere |
| 22:24.37 | dalfry | hmmm |
| 22:26.10 | NirS | god damn, Set(LANGUAGE=he) doesn't want to work correctly for me |
| 22:28.41 | NirS | ok, got it |
| 22:28.43 | NirS | tzafrir |
| 22:29.12 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
| 22:30.41 | *** join/#asterisk thoughtpolice (n=austin@c75-111-136-171.plaicmtc01.tx.dh.suddenlink.net) |
| 22:31.16 | NirS | oh, I see mister AsterJet has joined us |
| 22:32.49 | Siya | is there a simple way of setting moh per user? |
| 22:33.08 | NirS | per user moh ? |
| 22:33.47 | NirS | apart from using some form of AGI/SetMusicOnHold, I don't see a normal way of doing it |
| 22:34.01 | NirS | well, maybe something using DBget and DBput |
| 22:34.12 | NirS | but other than that, I'm lost of solutions |
| 22:34.19 | *** part/#asterisk BSD_Tech (n=BSDTech@adsl-69-230-174-37.dsl.irvnca.pacbell.net) |
| 22:38.30 | Siya | NirS: thanks, am browsing the wiki but can't find something scalable or simple enough... |
| 22:41.46 | *** join/#asterisk Un1x_Laptop (i=Sean@72.53.146.162) |
| 22:42.33 | *** join/#asterisk poppo (n=adas@S0106004063d8e527.ed.shawcable.net) |
| 22:42.51 | *** join/#asterisk drrt (n=junior@ppp-static2-140.tis-dialog.ru) |
| 22:42.55 | drrt | hi |
| 22:42.57 | NirS | Siya, well, using an AGI/DB combo would be scaleable enough, I think |
| 22:43.20 | hansin321 | Qwell[]: from earlier, I think [TM] is just a trade mark and can be used if you have done your due dilligence and have determined no one else has used it. [R] is a registered trade mark and means you actually registered it; much more official. At least that is how I understand it. |
| 22:43.48 | hansin321 | I'm sure Wikipedia knows ;) |
| 22:43.48 | poppo | I am trying to do the following a user signs up on a form and enters there phone number , asterisk calls and goes to an extension( this is done) what i need to know is how to promt the user to press one and as soon as they press one it updates databes |
| 22:45.12 | NirS | poppo, simple, perform an Answer, then do a Read function to a variable, then hangup the call. Then run a deadagi on the hangup, with the information obtained from the channel and update the database. |
| 22:45.17 | NirS | fairly straight forward |
| 22:45.29 | poppo | thanks |
| 22:48.05 | Siya | NirS: I'm a simple soul, AGI/DB sounds complex to me ... ;) |
| 22:48.27 | boch | can i dial Local channels with call files ? |
| 22:49.20 | NirS | Siya, well, you can always contract me to do the work for you |
| 22:49.21 | NirS | ;-) |
| 22:49.31 | NirS | boch - in general the answer is yes |
| 22:49.53 | NirS | just beware of chan_local, it's the devil in sheeps clothing |
| 22:50.08 | Siya | NirS: for home use on my *now, highly unlikely ;) but thanks for the offer anyway |
| 22:50.28 | Siya | gnight ppl |
| 22:50.43 | NirS | oh, I see |
| 22:50.48 | NirS | well, in that case |
| 22:51.01 | NirS | you should have said you are using AsteriskNOW |
| 22:51.39 | NirS | unfortunatly to say, as much as I like AsteriskNOW, I don't yet believe it has the facilities to enable a normal user to do stuff like that |
| 22:51.54 | NirS | what kind of scaleability were you thinking of ? |
| 22:52.11 | *** join/#asterisk acidchild (i=ash@unaffiliated/acidchild) |
| 22:52.20 | acidchild | anyone know a good SMS gateway provider in Canada? |
| 22:52.28 | NirS | sms.ac ? |
| 22:52.40 | acidchild | they provide many API's? |
| 22:52.56 | NirS | don't really know |
| 22:53.05 | NirS | used to work with them in the past (way back) |
| 22:53.51 | acidchild | i went to there website, and half of it is echoed java script errors... |
| 22:53.52 | acidchild | :/ |
| 22:53.53 | acidchild | brb. |
| 22:54.48 | NirS | try going to mBlox |
| 22:54.51 | NirS | they are also good |
| 22:55.03 | NirS | well, time for bed now |
| 22:59.26 | JT | Greek-Boy: don't believe all the hype you hear from digium about IAX2 being superior to SIP |
| 22:59.43 | JT | SIP will probably always be more used than IAX |
| 22:59.59 | n00dle | ~seen russelb |
| 23:01.21 | jbot | i haven't seen 'russelb', n00dle |
| 23:01.22 | n00dle | ~seen russellb |
| 23:02.02 | jbot | russellb is currently on #asterisk (5d 15m 5s). Has said a total of 27 messages. Is idling for 1d 1h 48m 9s, last said: 'not unless you want 4 ports instead of two ;)'. |
| 23:02.03 | acidchild | n00dle: /seenserv |
| 23:02.08 | JT | <PROTECTED> |
| 23:02.59 | jbot | i haven't seen 'should work, acidchild', JT |
| 23:02.59 | n00dle | The bot seems slow today. |
| 23:02.59 | acidchild | eye irritations.. :P |
| 23:02.59 | JT | the bot is always slow |
| 23:02.59 | acidchild | anyone else useing a SMSC/gateway to send alerts? |
| 23:02.59 | JT | it must be on a 2400bit/s modem link |
| 23:03.45 | n00dle | I'm having a weird problem with it, apparently. |
| 23:03.52 | JT | something the bugtracker can't help with? |
| 23:04.44 | *** join/#asterisk ircrly (i=astrutt@punk.valuetel.net) |
| 23:04.59 | *** join/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net) |
| 23:06.34 | n00dle | Not that I've been able to search out. |
| 23:07.00 | *** join/#asterisk sid (n=unstable@tor/regular/sid) |
| 23:07.24 | sid | Why does the bayonne project exist.. why doesn't the author of that hack on asterisk instead? |
| 23:08.33 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 23:11.27 | Corydon76-home | Bayonne predates Asterisk |
| 23:12.52 | Corydon76-home | IIRC, Bayonne was started back in the days when computer CPUs were not considered powerful enough to run PBXes, as they are now |
| 23:14.54 | *** join/#asterisk bluelinq (n=bluelinq@dsl-7-36.cofs.net) |
| 23:14.57 | bluelinq | anyone with experiece using asttapi? |
| 23:17.30 | acidchild | Corydon76-home: so how did they run them? |
| 23:17.42 | acidchild | on realy fast box's or all hardware/ |
| 23:17.50 | *** join/#asterisk elg (n=fugalh@216.31.27.110) |
| 23:17.59 | sid | Corydon76-home: Did you guys use code from Bayonne? |
| 23:18.07 | Corydon76-home | acidchild: back in the day, people coded directly to the hardware |
| 23:18.10 | *** join/#asterisk anderiv (n=anderiv@207-67-87-34.static.twtelecom.net) |
| 23:18.14 | acidchild | i see... |
| 23:18.20 | Corydon76-home | sid: no, Asterisk is coded from scratch |
| 23:18.26 | elg | so I see the strftime function, but is there a way to get better resolution than one second? milliseconds or nanoseconds perhaps? |
| 23:18.33 | sid | Corydon76-home: Does Bayonne use Asterisk code? |
| 23:18.37 | Corydon76-home | sid: no |
| 23:18.45 | Corydon76-home | elg: Nope |
| 23:19.14 | sid | Don't both the projects have the same goal? and they're in the same language? |
| 23:19.35 | Corydon76-home | sid: Bayonne is in C++. Asterisk is in C. |
| 23:19.56 | sid | ahh, Bayonee is in C++ |
| 23:20.34 | n00dle | Guess I'll have to actually open a new bug myself... |
| 23:20.57 | Corydon76-home | n00dle: on? |
| 23:21.24 | n00dle | SLA and not properly answering the line. |
| 23:21.30 | Corydon76-home | That'd probably be because very few people use SLA |
| 23:21.55 | n00dle | My boss requires the functionality that SLA is supposed to make easy. |
| 23:21.58 | Corydon76-home | I haven't seen anything beyond what Russell wrote |
| 23:22.22 | Corydon76-home | Well, file a bug, then |
| 23:22.37 | n00dle | Yeah... working on gathering the infos necessary. |
| 23:24.06 | elg | does mixmonitor replace %d like record does? |
| 23:24.22 | boch | how is the channel variable containing the chann caller id ? |
| 23:24.23 | *** join/#asterisk ptiggerdine_ (n=ptiggerd@203-219-14-182.static.tpgi.com.au) |
| 23:24.43 | Corydon76-home | boch: it's now a function called CALLERID() |
| 23:24.52 | Corydon76-home | boch: see 'show function CALLERID' |
| 23:25.13 | russellb | nooooo, not bugs! :-p |
| 23:25.16 | boch | Corydon76-home, i know, but i cant call a function via manager, right ? |
| 23:25.24 | russellb | n00dle: yeah, if it's not working, just file a bug ... i'll look when i can |
| 23:25.42 | Corydon76-home | boch: sure you can |
| 23:26.01 | Corydon76-home | boch: same way you get a variable |
| 23:26.41 | n00dle | russellb: Typing, cutting and pasting like mad even as we speak. :) |
| 23:26.48 | russellb | heh |
| 23:26.58 | boch | Corydon76-home, Variable: CALLERID(num) ? |
| 23:27.17 | Corydon76-home | boch: correct |
| 23:28.17 | Corydon76-home | boch: and you can set functions the same way |
| 23:29.53 | boch | Response: Success |
| 23:29.53 | boch | Variable: CALLERID(num) |
| 23:29.53 | boch | Value: (null) |
| 23:29.53 | Corydon76-home | Did you specify a valid active channel? |
| 23:29.53 | blitzrage | can someone explain what 'timeoutrestart' does in queue.conf? The description is a bit vague |
| 23:29.56 | boch | yep, if not returns Message: No such channel |
| 23:30.54 | Corydon76-home | boch: 1.2 or 1.4? |
| 23:31.00 | boch | 1.4 |
| 23:32.21 | *** join/#asterisk Cyon (n=cyon@216.179.31.170) |
| 23:32.21 | Corydon76-home | Are you sure? It won't work in 1.2, but it should work fine in 1.4 |
| 23:33.12 | *** join/#asterisk toerkeium (i=oo@dcc-hq-host-200-59-45-53.dnsba.com) |
| 23:33.49 | boch | 1.4.1 |
| 23:35.24 | Corydon76-home | boch: yeah, that was fixed in a more recent version of 1.4 |
| 23:35.39 | Corydon76-home | Fixed 6 weeks ago |
| 23:35.55 | boch | ops :( |
| 23:36.18 | *** join/#asterisk threat (i=threat@60-240-43-214.static.tpgi.com.au) |
| 23:36.29 | boch | i could use Status to get the chann cid right ? |
| 23:37.14 | [hC] | Corydon76-sorry for the mis-assignment of that app_macro bug. first time bugtracker user. |
| 23:37.22 | boch | yes, work great with Status |
| 23:37.22 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 23:37.37 | Corydon76-home | [hC]: do you see what the issue really is? |
| 23:38.02 | [hC] | Corydon76-home: I thought I did? (as described in my description) did i not get it? |
| 23:38.19 | Corydon76-home | Macro doesn't wait for a new extension when it runs out of priorities. |
| 23:38.29 | Corydon76-home | Macro returns when it runs out of priorities. |
| 23:38.30 | n00dle | what is "Disclaimer on File?" for? |
| 23:38.36 | Corydon76-home | You should use Read() instead. |
| 23:38.43 | Corydon76-home | n00dle: if you're submitting a patch |
| 23:39.16 | [hC] | Corydon76-home: Maybe i should amend my description. I had used WaitExten as well before, and it acts the same if i press the digit during the audio files playing. |
| 23:39.41 | [hC] | Corydon76-home: as well, Read() works fine, except i need to be able to trap a hangup. |
| 23:39.41 | Corydon76-home | [hC]: please try using Read() instead |
| 23:39.58 | n00dle | Ah... finally found it on my own. (Too many distractions at the office, it's an f-ing miracle I get anything done!) |
| 23:40.14 | Corydon76-home | [hC]: so set the variable first, then do a read, if you get a hangup, it's whatever the status you preset |
| 23:40.58 | [hC] | Corydon76-home: Yeah. '1' to confirm needs to unset the variable. Let me try that. J'sec. |
| 23:42.29 | [hC] | Corydon76-home: Ah, I remember why I had avoided using Read(), was to be able to hit a digit while the audio files were being played, instead of having to listen to the whole thing before hitting read. I guess in that case there's no way around it. |
| 23:42.55 | Corydon76-home | [hC]: uh, you can do that with Read(), too |
| 23:43.05 | Corydon76-home | I do that all the time |
| 23:43.49 | [hC] | Corydon76-home: I dont just play an audio file though, i play one, use saydigits, then play another. I guess I can do without saydigits. |
| 23:45.13 | toerkeium | guys, I have connected * to an analog pbx, which has connected a traditional analog fax machine. Every time I pass a call to the fax machine from calls which incoming ITSP is my VoIP provider, the fax dont print. If the call comes from my analog line, it works without problem. Do i Need enable something in * or any idea what's going on? |
| 23:45.41 | Corydon76-home | [hC]: or set a 1 second timeout on the first Read |
| 23:45.43 | *** join/#asterisk BSD_Tech (n=BSDTech@adsl-69-230-174-37.dsl.irvnca.pacbell.net) |
| 23:47.10 | *** part/#asterisk BSD_Tech (n=BSDTech@adsl-69-230-174-37.dsl.irvnca.pacbell.net) |
| 23:52.48 | JT | toerkeium: yes, you do not do faxing over VoIP |
| 23:52.52 | JT | toerkeium: doesn't work. |
| 23:53.03 | JT | you need to use T.38 which is FoIP |
| 23:53.11 | *** join/#asterisk dotSlashW (n=HTP@200.80.197.5) |
| 23:53.12 | JT | or not use IP |
| 23:54.05 | dotSlashW | so, I'm still trying to get asterisk talk to a shortel system |
| 23:54.12 | dotSlashW | how do I debug a SIP trunk ? |
| 23:54.16 | Ryushin | Anyone have a few minutes to help me figure out why my twinkle softphone won't work? I've been able to set up KIAX, but now I want to set up a sip based softphone. My polycom phones work fine. |
| 23:54.32 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 23:54.52 | JT | dotSlashW: sip debug |
| 23:55.00 | Ryushin | This should be really easy, but I don't know why it's being a pain. |
| 23:55.05 | JT | sip isn't a trunk btw |
| 23:55.57 | n00dle | Whew! Finally finished gathering info and reporting... http://bugs.digium.com/view.php?id=9851 |