| 00:00.26 | *** join/#asterisk slePP (~slepp@S01060040f48412ad.ed.shawcable.net) |
| 00:00.48 | VirTERM | astcc? |
| 00:01.38 | RomanTorres | emrah: do you mean astcc? |
| 00:01.43 | *** join/#asterisk AmaDEE0_ (~foo@209.52.90.78) |
| 00:01.56 | *** join/#asterisk coolacid-p (~jason@216.99.98.39) |
| 00:02.26 | *** join/#asterisk webman (~adamg@52.87.233.220.exetel.com.au) |
| 00:02.29 | *** join/#asterisk Majestik (~MajestiK@S01060800208687ec.ed.shawcable.net) |
| 00:02.31 | AmaDEE0_ | I see the person who wrote the asterisk book said they got 5000 concurent connections on Asterisk. Other people get 318 on a dual Zeon. What's up with that? |
| 00:02.57 | webman | AmaDEE)_: perhaps they should use Opteron :) |
| 00:03.35 | emrah | Excuse-me, but I'm experiencing a strange problem with ASTCC. When I call the AGI application extension, it asks for a 12 digits nmber... I have only a card with 8 digits... Do you have an idea? |
| 00:03.41 | webman | AmaDEE0_: or, more seriously, perhaps one is where only the control is passed through asterisk (ie sip re-invite) and the other is with traffic + rtp + perhaps transcoding |
| 00:04.07 | *** join/#asterisk ploch (tkk@expired.cluepon.org) |
| 00:04.14 | VirTERM | this is only the voice prompt |
| 00:04.23 | VirTERM | just enter your 8 digit number...:) |
| 00:04.25 | webman | AmaDEE0_: numbers like that are extremely confusing when you don't look at the whole picture.... |
| 00:05.12 | VirTERM | ctooley: are you going to post it here? sorry to bug you... |
| 00:06.11 | emrah | Thanks a lot |
| 00:07.00 | VirTERM | emrah: did it work? |
| 00:07.27 | VirTERM | anyone else runs realtime with iax peers? |
| 00:08.33 | emrah | VirTERM: Yes, but when I place a call, it says just that it will place the call but it hungs up... Do you know where I can get a bit documentation? I didn't completly understand the peer fiel in the form... |
| 00:08.57 | VirTERM | did you create your trunks and a routing table? |
| 00:09.03 | emrah | yes. |
| 00:09.11 | VirTERM | you have to specify which trunk to use for specific pattern |
| 00:09.12 | rvhi | on my * console, some servers show color coded messages, some don't, anyone knows why? |
| 00:09.15 | emrah | But I don't understand how work the trunk section |
| 00:09.49 | VirTERM | hold on, I will look at mine |
| 00:10.02 | emrah | Thanks :) thanks a lot |
| 00:12.01 | VirTERM | TOR IAX2 user:password@216.58.12.101 |
| 00:13.05 | VirTERM | then in trunks:1416 Canada TOR 0 0 280 |
| 00:13.29 | VirTERM | to specify that you want to use trunk TOR to dial 1416xxxxxxx |
| 00:13.31 | Wazb | i tried to register Asterisk to another SIP server , in return i am getting "1.1.1.1 does not implement 'REGISTER'" |
| 00:14.28 | VirTERM | if you have your iax trunks specified in iax.conf then you can use just a name of a trunk instead of ip address for FQDN |
| 00:15.06 | emrah | Thanks a lot |
| 00:15.23 | emrah | Trunk Name |
| 00:15.23 | emrah | Technology |
| 00:15.24 | emrah | Peer/Trunk |
| 00:15.34 | PoWeRKiLL | someone use forkcdr ? |
| 00:15.41 | emrah | so, in peer/trunk I write the name of the peer, and in the trunk name too? |
| 00:15.58 | Wazb | i tried to register Asterisk to another SIP server , in return i am getting "1.1.1.1 does not implement 'REGISTER'" error , Please help!!! |
| 00:16.35 | VirTERM | trunk name = alias to be used in routes |
| 00:16.57 | emrah | ok thanks a lot .) |
| 00:16.58 | VirTERM | peer/trunk actual user/password@trunk ip or fqdn |
| 00:17.06 | VirTERM | user:password |
| 00:17.21 | emrah | Thanks a lot :) |
| 00:17.26 | VirTERM | or name of this trunk from iax.conf |
| 00:17.34 | VirTERM | :) |
| 00:18.02 | VirTERM | realtime...ANYONE? |
| 00:18.04 | VirTERM | :) |
| 00:19.32 | dmccollum | Has anyone gotten the wakeup scripts to work right? |
| 00:21.38 | Moc | /msg bkw_ exten => _91NXXNXXXXXX,n,Dial(SIP/asterlink/${EXTEN:1}) |
| 00:23.22 | *** join/#asterisk cbachman (~chatzilla@victory.ece.northwestern.edu) |
| 00:27.10 | *** join/#asterisk shmooz (~nobody@host6411912762.biz.tor.fcibroadband.com) |
| 00:28.04 | *** join/#asterisk AsteriskNoob (AsteriskNo@207-114-232-10.gen.twtelecom.net) |
| 00:28.13 | AsteriskNoob | anyone here running a TDM400P? |
| 00:30.16 | AsteriskNoob | anyone? |
| 00:30.20 | AsteriskNoob | wow is this channel dead or what |
| 00:30.38 | shmooz | hi |
| 00:30.45 | dmccollum | sorry I'm just running a x100p |
| 00:30.59 | AsteriskNoob | ah yeah i've got 2 of those in my old box, they work great |
| 00:31.17 | AsteriskNoob | just got a tdm400p with 4 fxs modules |
| 00:31.22 | AsteriskNoob | cant get it to recognize the card |
| 00:31.31 | dmccollum | Is there a low cost single port FXS card? |
| 00:32.12 | fugitivo | dmccollum: linksys pap2-na, 2 ports, $60 |
| 00:32.33 | fugitivo | works great |
| 00:32.55 | dmccollum | thanks |
| 00:33.06 | Darwin35 | ok what ver are we upto ? |
| 00:33.55 | dmccollum | Do both ports appear as one mac address or are there one for each port? |
| 00:34.32 | fugitivo | dmccollum: it's one rj45, and 2 (rj11?) for regular phones |
| 00:34.44 | mesi | Ok, fwdOUT sucks BIG TIME for there is so few traffic :-( |
| 00:34.45 | fugitivo | one mac address |
| 00:35.04 | AsteriskNoob | anyone here running a TDM400P? |
| 00:37.56 | riquisim0 | brb |
| 00:39.37 | machinehd | AsteriskNoob, are you saying it doesn't detect the hardware on bootup? |
| 00:40.55 | machinehd | if so, you tried switching to another pci slot? running kudzu? |
| 00:41.41 | AsteriskNoob | it works now |
| 00:41.52 | AsteriskNoob | i must not have had it seated all the way |
| 00:42.02 | AsteriskNoob | i noticed when i took the screw out of the board it just fell to the bottom of the case... |
| 00:42.05 | AsteriskNoob | :) |
| 00:43.10 | AsteriskNoob | now here's a dumb question |
| 00:43.20 | AsteriskNoob | how do i find out what the channel numbers are for the cards? |
| 00:44.43 | Darwin35 | if you have 1 card its zap 1 - 4 |
| 00:44.55 | AsteriskNoob | i also have a PRI card in there |
| 00:45.12 | AsteriskNoob | so i have a wcte110 and a tdm400 |
| 00:45.31 | Darwin35 | zap show channels |
| 00:45.41 | *** join/#asterisk |Vulture| (~Vulture@152.238.204.68.cfl.res.rr.com) |
| 00:45.43 | AsteriskNoob | dont i have to configure the zaptel.conf first? |
| 00:45.49 | Darwin35 | yes |
| 00:46.23 | Darwin35 | not done that on this side with a pri in the box |
| 00:46.28 | Darwin35 | only a tdm |
| 00:46.57 | AsteriskNoob | ah |
| 00:47.17 | AsteriskNoob | well asterisk didnt even install this time, gotta go review the prereq's again |
| 00:47.19 | AsteriskNoob | thanks |
| 00:47.46 | *** part/#asterisk bjohnson (~bjohnson@66.11.165.161) |
| 00:49.02 | *** part/#asterisk eKo1 (~bernd@63.245.57.70) |
| 00:52.16 | NewSole | those tdm400's what speed cpu is needed |
| 00:56.21 | *** join/#asterisk wt (~wt@242679hfc16.tampabay.res.rr.com) |
| 00:56.41 | wt | anyone here know of a good phone that can handle the elements? |
| 00:56.54 | wt | like a wall mounted thing |
| 00:57.27 | wt | kinda like what you see at an apartment complex to open a gate? |
| 00:58.21 | hardwire | are you rich? |
| 00:58.25 | hardwire | can I hit you up for money? |
| 00:58.32 | NewSole | lol |
| 00:59.02 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@l01m-20-133.d4.club-internet.fr) |
| 00:59.23 | *** join/#asterisk cybvip (~cybvip@64-235-222-179.access.ripnet.com) |
| 00:59.48 | cybvip | Hello? |
| 01:01.03 | cybvip | Can anyone here help me with an asterisk@home 0.7 problem? |
| 01:01.39 | NewSole | lol |
| 01:01.55 | cybvip | hi |
| 01:02.09 | NewSole | hi |
| 01:02.24 | cybvip | Are you a veteran asterisk user? |
| 01:02.32 | NewSole | everyone left it seems.... |
| 01:02.38 | cybvip | I see |
| 01:02.57 | cybvip | Do you have a system up and running? |
| 01:03.19 | NewSole | I do... but not an @ home system |
| 01:03.48 | cybvip | perhaps you can help me with a quick issue I have |
| 01:03.57 | NewSole | I can try... |
| 01:04.29 | VirTERM | cybvip: shoot |
| 01:04.44 | NewSole | right now having problems of my own..... TDM400 card giving headacks |
| 01:04.45 | cybvip | I just bought a digium TDM400P card and install it in a dell optiplex 110 system (PIII 733 /w 256M Ram) |
| 01:05.14 | cybvip | I installed the asterisk at home version 7.0 which compiled ok |
| 01:05.40 | cybvip | and I have dial tone to a set I connected to one of the fxs port |
| 01:06.03 | cybvip | but everytime I try to dial anything I get a reorder tone |
| 01:06.09 | VirTERM | did you configure the card? installed drivers etc |
| 01:06.17 | VirTERM | can you see the card ? |
| 01:06.36 | VirTERM | do you have a dialplan configured? |
| 01:06.51 | cybvip | I ran the getzaptelconf -s -d and as soon as I did I received dialtone on the set |
| 01:07.19 | VirTERM | ok but you need simply dialplan to be able to route your call properly |
| 01:07.36 | cybvip | I am using amp |
| 01:07.42 | cybvip | where do i set that up |
| 01:07.48 | VirTERM | never used amp... |
| 01:07.54 | VirTERM | command line only :) |
| 01:08.02 | VirTERM | but it's the same concept |
| 01:08.10 | VirTERM | what do you see on the console |
| 01:08.17 | VirTERM | when you try to dial |
| 01:08.36 | cybvip | The system is in the next room. I am accessing it via the web port amp program |
| 01:08.45 | NewSole | VirTERM.... any idea on my problem |
| 01:08.58 | VirTERM | you need a dialplan specifying how to route your call plus the fxs port has to be in the right context to have rights to us this specific portion of your dialplan |
| 01:09.20 | VirTERM | us=>use |
| 01:09.33 | VirTERM | newsole: what is your problem? |
| 01:09.33 | cybvip | be back in 5 . .can u help me then . . . I'll move the machine closer so I can talk to you and look at it |
| 01:09.36 | cybvip | ok |
| 01:09.49 | VirTERM | give me 5 mins, I need a smoke |
| 01:09.50 | VirTERM | :) |
| 01:09.53 | cybvip | thanks |
| 01:10.15 | NewSole | VirTERM.... any idea |
| 01:10.35 | NewSole | why my TDM Card will only allow me to make 1 call at a time |
| 01:11.09 | *** join/#asterisk implicit (~implicit@ip68-7-149-247.sd.sd.cox.net) |
| 01:12.17 | chap | NewSole: How many modules do you have on it? |
| 01:12.41 | NewSole | just the 2.... fxs and fx0.... the dev card |
| 01:13.00 | chap | So that means you cannot do more than one outgoing call at a time... |
| 01:13.07 | *** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk) |
| 01:13.11 | NewSole | thats not what i mean |
| 01:13.17 | NewSole | I make a call |
| 01:13.33 | NewSole | then I have to stop asterisk and restart it to make another one |
| 01:13.49 | NewSole | and that should not happen |
| 01:14.49 | *** join/#asterisk phantasis (~HighOctan@137.28.208.13) |
| 01:15.39 | *** join/#asterisk TheSin (~TheSin@iphost-64-56-130-194.edm.wiband.net) |
| 01:15.50 | chap | NewSole: Are you hanging up the call in the dialplan? |
| 01:16.11 | NewSole | yes |
| 01:16.16 | cybvip | i'm back |
| 01:17.04 | *** join/#asterisk slePP (~slepp@S01060040f48412ad.ed.shawcable.net) |
| 01:17.12 | phantasis | when is a good point to use SER instead of * for SIP routing? |
| 01:17.14 | cybvip | you there VirTERM |
| 01:17.42 | Shido6 | when you dont want to build a proper dialplan and sip.conf, phantasis |
| 01:17.44 | VirTERM | yeap |
| 01:18.07 | *** join/#asterisk bmd (~bmd@rrcs-66-27-57-228.west.biz.rr.com) |
| 01:18.19 | cybvip | ok . . when I dial anything I am not seeing nothing on the console |
| 01:18.30 | VirTERM | hmmm... |
| 01:18.35 | cybvip | do I have to turn on debugging |
| 01:18.45 | cybvip | I am at the root prompt |
| 01:18.47 | VirTERM | do you have you zapata.conf configured? |
| 01:18.57 | *** join/#asterisk santiago (~santiago@63.245.86.93) |
| 01:18.59 | bmd | It's probably the wrong time of day, but does anybody have any experience with British Telecom disconnect supervision? |
| 01:19.05 | VirTERM | * has to interface with your card though this file |
| 01:19.09 | cybvip | I beleive the genzapteconf -s -d does an autoconfigure for this file |
| 01:19.22 | VirTERM | I am not familiar with the home version |
| 01:19.28 | VirTERM | just check in your /etc/asterisk |
| 01:19.37 | cybvip | ok hold on |
| 01:20.36 | cybvip | i have zapata-channels.conf, zapata-default.conf, zapata.conf, and zapata_additional.conf |
| 01:21.07 | VirTERM | hmmm....I would start with zapata.conf |
| 01:21.25 | VirTERM | you have to create channels then groups if you want to |
| 01:21.28 | cybvip | ok I have it open . .what am i looking for |
| 01:21.42 | *** join/#asterisk AgiNamu (~AgiNamu@216.230.151.230) |
| 01:21.47 | VirTERM | do you see anything under [channels]? |
| 01:21.56 | cybvip | yes |
| 01:22.00 | AgiNamu | heh, how long ago was I kicked. |
| 01:22.42 | VirTERM | ok, lets go back to the console |
| 01:22.48 | VirTERM | type zap show channels |
| 01:23.04 | cybvip | A bunch of stuff regarding echo cancellation |
| 01:23.07 | cybvip | ok hol don |
| 01:23.32 | VirTERM | you should see ALL channels which are active (ready to be used) |
| 01:23.55 | cybvip | there is no zap command |
| 01:24.06 | VirTERM | type help |
| 01:24.27 | cybvip | at the bottom of the zapata.conf file there is an #include for the zapata-channels.conf file |
| 01:24.35 | cybvip | in it there are channels specified |
| 01:24.47 | VirTERM | I see, must be the AMP or * @ home :) |
| 01:25.06 | VirTERM | if you type "help" on the console you will see the list of all available commands |
| 01:25.19 | VirTERM | you are looking for "zap show channels" |
| 01:25.36 | VirTERM | zap show channels Show active zapata channels |
| 01:26.53 | VirTERM | we have to confirm from the console that the channels are configured properly |
| 01:26.55 | cybvip | I did the help thing but there does not seem to be any show or zap commands |
| 01:26.58 | VirTERM | and visible to asterisk |
| 01:27.08 | VirTERM | hmmm, |
| 01:27.27 | VirTERM | can you see in any zapata files line starting with "context" |
| 01:27.32 | cybvip | before I did the genzaptelconf -d -s I had no dial tone on the sets and after the command I did |
| 01:28.17 | VirTERM | look for context in zapata files and make sure that you have a dialplan in extensions.conf under corresponding context |
| 01:28.40 | cybvip | jold on I'll look |
| 01:28.44 | VirTERM | zaptel is a driver |
| 01:28.51 | VirTERM | zapata is a config for * |
| 01:29.15 | *** join/#asterisk MikeJ[Jayden] (~jayden@pcp02795302pcs.roylok01.mi.comcast.net) |
| 01:30.36 | cybvip | in the zapata-default file there is a line "context=from-pstn |
| 01:30.38 | cybvip | " |
| 01:31.08 | VirTERM | ok |
| 01:31.29 | VirTERM | now look for "from-pstn" in your extensions.conf file |
| 01:31.44 | cybvip | found it |
| 01:31.45 | VirTERM | it will be in brackets from-pstn] |
| 01:31.50 | VirTERM | it will be in brackets [from-pstn] |
| 01:31.56 | cybvip | there are three lines |
| 01:32.00 | VirTERM | this is a context |
| 01:32.13 | cybvip | include => from-pstn-custom; create this context in extensions_custom.conf to include customizations |
| 01:32.13 | cybvip | include => ext-did |
| 01:32.13 | cybvip | include => from-pstn-timecheck ; this has to be included otherwise it overrides ext-did |
| 01:33.22 | cybvip | there is also an [from-internal] context |
| 01:33.24 | VirTERM | so, you have to create this context and put your sample dialplan there to call some devices or apps (voicemail) |
| 01:33.46 | VirTERM | what card do you have? fxo or fxs? |
| 01:34.08 | cybvip | I have a tdm400P wildcard with 3 fxs and 1 fxo module |
| 01:34.28 | cybvip | i have a set on line 1 and a outgoing pstn line on 4(fxo) |
| 01:34.37 | VirTERM | ok, most likely the fxo module is using context from-pstn and the other 3 from-internal |
| 01:34.45 | *** join/#asterisk IQ (~iq@65-103-165-206.omah.qwest.net) |
| 01:35.02 | VirTERM | you need to create some extensions under from-internal |
| 01:35.04 | cybvip | there is already quite a bit or text under the [from internal] |
| 01:35.16 | VirTERM | but confirm if I was correct |
| 01:35.19 | *** part/#asterisk IQ (~iq@65-103-165-206.omah.qwest.net) |
| 01:35.27 | VirTERM | like what? |
| 01:35.37 | *** join/#asterisk BBRodriguez (~BBRodrigu@pD9563686.dip.t-dialin.net) |
| 01:35.38 | VirTERM | exten=>blah blah? |
| 01:36.01 | cybvip | no, a bunch of include => lines |
| 01:36.18 | VirTERM | ok, so these lines are including other contexts |
| 01:36.33 | *** join/#asterisk IQ (~IQ@65-103-165-206.omah.qwest.net) |
| 01:36.35 | VirTERM | look into them and find out what extensions you have there |
| 01:36.46 | cybvip | then under that section is section for extension contexts |
| 01:37.05 | VirTERM | exten=>1000,1,VoicemailMain |
| 01:37.21 | VirTERM | reload extensions, then dial 1000; you should get into voicemail |
| 01:37.30 | *** part/#asterisk IQ (~IQ@65-103-165-206.omah.qwest.net) |
| 01:37.40 | cybvip | what? |
| 01:37.55 | cybvip | I don't see the exten=>1000 |
| 01:37.57 | AgiNamu | anyone around? |
| 01:38.09 | elriah | Yes. |
| 01:38.37 | VirTERM | either put this line there (this is an example) or search your dialplan for voicemailmain to know what is predefined extensions to access your voicemail |
| 01:38.57 | cybvip | there is another file called extensions_additional.conf that seems to hold the info that I entered from the amp |
| 01:39.09 | VirTERM | good |
| 01:39.21 | VirTERM | do you have this file included at the bottom of extensions.conf? |
| 01:39.38 | cybvip | hold on i'll check |
| 01:40.00 | AgiNamu | heh, I just got an email from someone who flashed their phone to IAX2. they included a SIP debug log and asked why it didnt work :P |
| 01:40.23 | cybvip | this is include at the beginning of the extensions.conf file : ; include extension contexts generated from AMP |
| 01:40.23 | cybvip | #include extensions_additional.conf |
| 01:40.40 | VirTERM | type "show dialplan" from the cli to see what's there |
| 01:40.45 | cybvip | it also says Customizations to this dialplan should be made in extensions_custom.conf |
| 01:40.59 | VirTERM | ok, must be AMP thing :) |
| 01:41.12 | cybvip | the show command is not present at the console |
| 01:41.41 | VirTERM | "show dialplan" not just show |
| 01:42.12 | cybvip | i did that and got -bash: show command not found |
| 01:42.22 | VirTERM | ok, look in the extensions_addit...... and find some valid extensions there |
| 01:42.29 | VirTERM | ?!?!? |
| 01:42.38 | VirTERM | you have to run asterisk console!!!! |
| 01:42.51 | cybvip | how do I start the asterisk console |
| 01:43.12 | VirTERM | PATH/asterisk -vvvvrc |
| 01:43.22 | cybvip | brb |
| 01:43.32 | VirTERM | more vvv gives you more verbose |
| 01:43.36 | *** join/#asterisk Qorky (~goaway@dsl-202-72-146-104.wa.westnet.com.au) |
| 01:43.56 | cybvip | ok the command works now |
| 01:44.06 | *** join/#asterisk cyberkoa (~cyberkoa@65.90.93.3) |
| 01:44.09 | VirTERM | "zap show channels" |
| 01:44.17 | VirTERM | can you see all your channels? |
| 01:44.24 | VirTERM | then do "show dialplan" |
| 01:44.42 | Qorky | i had this problem once before. and cant remember what i needed to delete: I had a cvs version of asterisk installed and now have gone back to 1.0.6 and now asterisk wont load. it complains about Loading module pbx_dundi.so failed! |
| 01:44.56 | VirTERM | remove modules |
| 01:45.07 | AgiNamu | delete /usr/lib/asterisk/modules |
| 01:45.08 | Qorky | theres a directory i had to clean out last time to get it to work again, does anyone know where that is? |
| 01:45.12 | *** part/#asterisk outsidefactor (~blah@203-206-247-72.dyn.iinet.net.au) |
| 01:45.14 | Qorky | ahh yeah thats right. |
| 01:45.14 | AgiNamu | i.e., delete everything in there. |
| 01:45.23 | cybvip | i see 4 channels but they are all show context as being "from-pstn" |
| 01:45.33 | VirTERM | unless you have g.729 :) |
| 01:45.59 | cybvip | and no extension beside them |
| 01:46.01 | VirTERM | ok, then you have to look for valid extensions under this context |
| 01:46.12 | VirTERM | you will be able to dial these extensions |
| 01:46.32 | VirTERM | remember that other contexts can be included, so you also look into them |
| 01:46.34 | cybvip | there is nothing under the extension field |
| 01:46.47 | cybvip | when i do a show channels |
| 01:47.11 | VirTERM | no, not from the console :) in your extensions....conf files |
| 01:47.19 | Qorky | cheers guys. works. |
| 01:47.27 | cybvip | ok . .hold on |
| 01:47.27 | VirTERM | show channels will show you active (in use) channels |
| 01:47.36 | VirTERM | good :) |
| 01:47.56 | VirTERM | only , if someone would help me with realtime configuration...:) |
| 01:48.34 | *** join/#asterisk cyberkoa (~cyberkoa@65.90.93.3) |
| 01:48.36 | cybvip | when i go offhook with set 1 I see it on the console |
| 01:49.09 | cybvip | ok I am in the extensions_additional.conf file |
| 01:49.21 | cybvip | what am I looking for here |
| 01:49.53 | cybvip | [globals] |
| 01:49.53 | cybvip | ZAPCHAN_201 = 2 |
| 01:49.53 | cybvip | ZAPCHAN_200 = 1 |
| 01:50.00 | VirTERM | you are looking for the same context which your zap channel is trying to use |
| 01:50.15 | VirTERM | [from-pstn] |
| 01:50.18 | VirTERM | look for this |
| 01:50.21 | cybvip | this is in the file and those are the ext numbers I programmed from the amp |
| 01:50.25 | cybvip | ok hol don |
| 01:51.05 | cybvip | the [from-pstn] is in the extensions.conf file |
| 01:51.17 | VirTERM | ok, what do you have under it? |
| 01:51.23 | cybvip | [from-pstn] |
| 01:51.23 | cybvip | include => from-pstn-custom; create this context in extensions_custom.conf to include customizations |
| 01:51.23 | cybvip | include => ext-did |
| 01:51.23 | cybvip | include => from-pstn-timecheck ; this has to be included otherwise it overrides ext-did |
| 01:51.46 | cybvip | and then there is other sections such as [from-pstn-timecheck] |
| 01:52.03 | VirTERM | do you have any lines starting with exten there? |
| 01:52.26 | cybvip | [from-pstn-reghours] |
| 01:52.26 | cybvip | exten => s,1,GotoIf($[${FAX_RX} = disabled]?from-pstn-reghours-nofax,s,1:2); if fax detection is disabled, then jump to from-pstn-nofax - else continue |
| 01:52.26 | cybvip | exten => s,2,Answer |
| 01:52.26 | cybvip | exten => s,3,Wait(1) |
| 01:52.26 | cybvip | exten => s,4,SetVar(intype=${INCOMING}) |
| 01:52.36 | cybvip | just a snippet |
| 01:52.39 | cybvip | there is way more |
| 01:52.53 | VirTERM | why don't you create some sample extension in extensions_custom |
| 01:53.05 | VirTERM | exten => 1000,1,VoiceMailMain |
| 01:53.10 | cybvip | let me open that file |
| 01:53.23 | VirTERM | then tpye "extensions reload" from the console |
| 01:53.33 | VirTERM | and dial 1000 from the phone connected to the fxs port |
| 01:54.44 | cybvip | I did that and I am still getting reorder tone by the second digit |
| 01:56.15 | libpcp | anyone has an experienced configuring SMS? i would like to know if this setup will work. GSM1-Asterisk--internet--Asterisk-GSM2. user will send SMS to GSM1 and it will send the sms message going out to GSM2 |
| 01:56.21 | VirTERM | what is the error on the console? |
| 01:56.43 | VirTERM | libpcp: yes it will work |
| 01:57.28 | cybvip | it says "starting simple switch on zap/1-1" |
| 01:57.44 | cybvip | then it says hungup 'zap1/1' |
| 01:58.02 | fugitivo | i need info about logging with asterisk |
| 01:58.10 | fugitivo | call logs, duration, etc |
| 01:58.38 | VirTERM | use sql based cdr... |
| 01:58.40 | VirTERM | areski |
| 01:58.46 | cybvip | As soon as I hit any number on the phone i get "--HUNGUP 'zap 1/1' |
| 01:59.04 | VirTERM | confirm that channel 1 zapata is in the "pstn-incoming" context |
| 01:59.16 | cybvip | in which file? |
| 01:59.17 | *** join/#asterisk Jim^^ (jd3nn1s@16.85.171.66.subscriber.vzavenue.net) |
| 01:59.42 | VirTERM | zapata.conf or any other zapata..... generated by AMP |
| 01:59.51 | libpcp | VirTERM: do you have any documentation for the SMS and maybe you can give me an idea how to accomplish that since this is my first time to configure that kind of setup, thanks |
| 01:59.58 | fugitivo | VirTERM: thanks |
| 02:00.24 | Jim^^ | what's a good silence threshold value? I'm using app_machinedetect and when I hit speaker to answer a call it never times out |
| 02:01.19 | VirTERM | libpcap: not really, but search wiki for sms asterisk |
| 02:01.34 | VirTERM | I am doing my sms through ser connected to * |
| 02:01.50 | *** part/#asterisk bmd (~bmd@rrcs-66-27-57-228.west.biz.rr.com) |
| 02:02.12 | *** part/#asterisk santiago (~santiago@63.245.86.93) |
| 02:02.37 | VirTERM | cybvip: http://asteriskathome.sourceforge.net/handbook/index.html |
| 02:02.43 | VirTERM | you should also read this |
| 02:03.06 | fugitivo | ~areski |
| 02:03.19 | phantasis | anyone running asterisk on Solaris? |
| 02:03.53 | VirTERM | good question :) |
| 02:04.15 | *** join/#asterisk gottago (visionv@c-67-167-116-116.hsd1.il.comcast.net) |
| 02:04.30 | *** join/#asterisk PMantis (~sswitzer@65-37-0-159.nrp2.roc.ny.frontiernet.net) |
| 02:04.35 | cybvip | i don;t see anything in the zapata files re: pstn-incoming |
| 02:04.55 | cybvip | Is the book from signate any good at assisting in setting up this system |
| 02:05.04 | VirTERM | please read the URL above; this should get you started |
| 02:05.08 | phantasis | i have that book |
| 02:05.10 | phantasis | its decent |
| 02:05.14 | VirTERM | no idea, |
| 02:05.16 | cybvip | I've already read that |
| 02:05.20 | cyberkoa | hi , i am newbie in asterisk . I just want to know whether can I dial from a phone connected to an Asterisk Server 1, to another Asterisk Server 2 , and call out from Asterisk Server 2 |
| 02:05.37 | phantasis | yep |
| 02:05.40 | PMantis | The asterisk.org site only lists 1.0.6 tarballs. The FTP serve has 1.0.7 but no libpri or zaptel drivers. Where are they? :) |
| 02:05.40 | cybvip | the @home version was 'supposed' to be easy to set up |
| 02:05.41 | VirTERM | it's all simply..once you have it up and running...unless you are trying to implement realtime :) |
| 02:05.44 | phantasis | with IAX |
| 02:05.59 | cyberkoa | I want to use this method to eliminate the oversea calling cost |
| 02:06.05 | *** join/#asterisk MikeJ[Laptop] (~icechat5@pcp02795302pcs.roylok01.mi.comcast.net) |
| 02:06.06 | cybvip | what version of asterisk and what flavor of linux are you using |
| 02:06.56 | VirTERM | fxs->*1->iax2->*2->pstn |
| 02:07.26 | gottago | Shido6: We agreed to work tonight. Are you listening? |
| 02:08.15 | elriah | Has anyone ever transferred a call with the * manager api? |
| 02:09.05 | *** join/#asterisk philgersekowski (~phillip@210.18.204.114) |
| 02:09.33 | MikeJ[Laptop] | manager is more of an interface than an api |
| 02:09.39 | VirTERM | PMantis: why don't you use CVS? |
| 02:09.55 | MikeJ[Laptop] | but all of the gui manager programs use that method |
| 02:10.03 | phantasis | anyone on using Solaris and Asterisk? |
| 02:10.21 | philgersekowski | I am new to asterisk, and am having problems with a Digium TDM400P Card and Linux - can anyone help ? |
| 02:10.29 | *** join/#asterisk ivesti (ivesti@ppp-68-251-35-227.dsl.chcgil.ameritech.net) |
| 02:10.39 | MikeJ[Laptop] | phantasis, not with zaptel, but at least as of recent, it ran, both on sparc and intel |
| 02:10.50 | marlowe | ea. |
| 02:10.51 | marlowe | jf2q |
| 02:10.53 | marlowe | woops |
| 02:10.55 | ivesti | hello, did anyone here ever used the Dock-nTalk device? |
| 02:11.31 | phantasis | MikeJ, I was thinking of using a sun midframe as the core asterisk router and SER and then having a cluster of x86 machines doing zaptel |
| 02:11.34 | phantasis | that work? |
| 02:11.38 | jakepdev | ivesti - that's just a cell phone dock? |
| 02:11.51 | ivesti | jakepdev,yes |
| 02:12.03 | tainted- | ivesti i was thinking about trying it out as well |
| 02:12.12 | jakepdev | ivesti - is there a question you had about it's use with *? |
| 02:12.13 | tainted- | ivesti there is also cellsocket.com by pulver |
| 02:12.38 | ivesti | tainted-, I just bought one but the sound quality isn't good at all. It's either the device either the cell phone(an old nokia 6190) |
| 02:13.00 | tainted- | is the sound quality good through the cell phone by itself? |
| 02:13.03 | ivesti | tainted-, yes but cellsocket works only with motorola phones |
| 02:13.16 | jakepdev | I had a good experience with a device similar - only that it would occasionally lock up and I needed to reset it |
| 02:13.25 | ivesti | tainted-, yes the voice quality is perfect |
| 02:13.40 | jakepdev | the echo though was kind of rough sometimes though |
| 02:13.47 | ivesti | what should I do the improve the voice quality? |
| 02:13.54 | jakepdev | (could be a cheap analog phone) |
| 02:14.13 | tainted- | ivesti did u try contacting the seller? |
| 02:14.14 | phantasis | MikeJ, is one O/S a better performer than another? |
| 02:14.21 | ivesti | tainted-, no |
| 02:15.16 | tainted- | ivesti try contacting them |
| 02:15.38 | ivesti | yeah,that's what I will do |
| 02:16.05 | PMantis | Wow, that's neat |
| 02:16.23 | VirTERM | good nite veryone |
| 02:16.51 | gottago | anyone know how to wake up Shido6? He has an appointment to work on our box tonight |
| 02:16.56 | PMantis | Now, I can get a cell phone, an extra FXO port, the dock-n-talk, then I don't have to keep my landline just for 911 support. Hmmmmmmmmmm. |
| 02:17.25 | fugitivo | VirTERM: do i have to setup asterisk to log to a database, the about of asterisk-stat is not much help |
| 02:17.39 | ivesti | PMantis ;) |
| 02:17.46 | jakepdev | pmantis - just make sure you put a ups on that dock-n-talk also |
| 02:18.20 | PMantis | True. |
| 02:18.38 | Jim^^ | I am using app_machinedetect to detect answering machines, is this what people would recommend? |
| 02:18.58 | tainted- | what does it take to provide DIDs? |
| 02:19.18 | PMantis | Hmmm. I'm also thinking this would be neat so we don't have to carry our cell phones around with us when we're home... can take the calls on our "normal" phone heh. |
| 02:19.55 | *** join/#asterisk MrBelvedr (~tt@ip68-227-209-110.dc.dc.cox.net) |
| 02:20.28 | jakepdev | pmantis - make sure you test the config out for a while though - I had several calls with echo issues. could've been a cheap analog phone though |
| 02:20.48 | cyberkoa | VirTERM , is this the answer for my question ? fxs->*1->iax2->*2->pstn |
| 02:21.28 | cybvip | VirTERM . .thanks for your help . .I think I better go do somemore reading |
| 02:21.40 | cybvip | what flavor of linux are you running? |
| 02:22.02 | PMantis | jakepdev: Ok. Of course FXO interfaces are rather cumbersome... no good hangup detect, etc. Call-waiting is also a challenge. |
| 02:22.26 | *** join/#asterisk IQ (~iq@65-103-165-206.omah.qwest.net) |
| 02:22.47 | jakepdev | now if you could somehow inteface with the cell data path to control the call... :) |
| 02:23.32 | jakepdev | some phones have a data port that function like modems. you might be able to hack it pretty easily |
| 02:24.00 | fugitivo | i'm using a motorola c333, works perfect in linux with the usb interface |
| 02:24.06 | cyberkoa | if I would like to configure in such a way : fxs->*1->iax2->*2->PSTN , any recommended resource on the Internet can I refer to ? |
| 02:24.07 | gottago | does anyone know Shido6? |
| 02:24.14 | ivesti | I have a sipura spa-3000 and I try to configure it with the dock-n-talk.The problem now is that I have an analog phone and the voice quality isn't good at all. I can barely hear something |
| 02:24.26 | jakepdev | gottago - in a biblical sense? no |
| 02:24.38 | *** join/#asterisk subtract (~subtract@ottawa-hs-209-217-99-135.d-ip.magma.ca) |
| 02:25.20 | gottago | jakepdev: nope. He was looking for work. We had a deal and now he is not answering. So I am wondering if there is anyone who knows him who can contact him. |
| 02:25.33 | jakepdev | gottago - he's a regular on here |
| 02:25.39 | tainted- | gottago stay away from him... he's a con artist |
| 02:25.39 | gottago | Yea |
| 02:25.54 | jakepdev | gottago - and on the nufone channel |
| 02:26.02 | gottago | tainted-: I would line to dcc chat with you, ok? |
| 02:26.14 | tainted- | sure |
| 02:26.14 | gottago | like too |
| 02:32.34 | fugitivo | if i have a phone with only number keys and * #, how do i transfer a call? |
| 02:32.49 | fugitivo | no transfer button |
| 02:33.42 | cbachman | gottago, he was here about about hour ago... |
| 02:33.44 | *** part/#asterisk gtigene (~gnadenx@c-67-184-112-58.hsd1.il.comcast.net) |
| 02:33.50 | *** join/#asterisk l-fy (~pchitescu@l-fy.developer.yate) |
| 02:34.14 | meanphil | fugitivo: tap the hangup thingy quickly? |
| 02:34.20 | tainted- | anyone here familiar with 5ess switches? |
| 02:34.49 | fugitivo | meanphil: mmm, i don't know if that works, it's a wireless phone |
| 02:35.05 | meanphil | oh |
| 02:35.06 | fugitivo | meanphil: no key combination for that? |
| 02:35.14 | meanphil | don't know sorry |
| 02:35.16 | fugitivo | meanphil: maybe # something |
| 02:35.17 | phantasis | is there a 64-bit version of asterisk? |
| 02:35.23 | *** join/#asterisk bparker (bparker@cable-71-8-65-183.mtv.al.charter.com) |
| 02:35.29 | fugitivo | phantasis: yes, compile it in a 64bit system |
| 02:35.43 | phantasis | is there any performance penalty? |
| 02:36.00 | phantasis | does asterisk support smp? |
| 02:37.13 | cbachman | phantasis, yes |
| 02:38.40 | phantasis | to which ? |
| 02:39.11 | cbachman | oops, to the smp. Asterisk is multi-threaded and thus should be able to take advantage of smp based systems. |
| 02:39.11 | Shido6 | boink |
| 02:39.27 | gottago | Shido6: Hello |
| 02:39.35 | gottago | we need to talk |
| 02:40.51 | gottago | GTI ring a bell? |
| 02:41.17 | *** join/#asterisk zhier (~nick@61.140.191.252) |
| 02:41.23 | *** join/#asterisk slePP (~slepp@S01060040f48412ad.ed.shawcable.net) |
| 02:41.33 | Qorky | I have a normal fax machine connected up to the same line as an x100p. when I send a fax out asterisk things theres an incomming call. is there anyway to stop that? |
| 02:46.19 | *** join/#asterisk slePP (~slepp@S01060040f48412ad.ed.shawcable.net) |
| 02:49.04 | FuriousGeorge | doesnt this gnophone thing have a gui? |
| 02:49.31 | *** join/#asterisk dan2 (dan@dan2.active.supporter.pdpc) |
| 02:49.41 | l-fy | FuriousGeorge > do you need a linux iax client? |
| 02:49.43 | webman | qorky: don't connect the X100P to the same line as the fax ?? Is the fax connected to the back of the x100p, or is there a 'double adaptor' somewhere else on the line? |
| 02:49.54 | FuriousGeorge | i-fy yeas please |
| 02:50.10 | l-fy | FuriousGeorge > yate.null.ro - gtk or qt pick your flavor :) |
| 02:50.24 | FuriousGeorge | qt |
| 02:50.32 | FuriousGeorge | with sprinkles |
| 02:51.03 | FuriousGeorge | and what is yate.null.ro |
| 02:51.30 | Qorky | double adaptor somewhere else. |
| 02:51.50 | l-fy | is a telephony engine, pretty much you have the technologies there like h323, sip, iax ,zaptel and you can use it as a client or server |
| 02:52.08 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 02:53.10 | tainted- | anyone familiar with lucent 5ess? |
| 02:53.15 | Qorky | but i could rig it up to be out of the back of the x100p. if that would help. |
| 02:53.32 | FuriousGeorge | I-fy: so what is the client for qt? |
| 02:53.43 | l-fy | FuriousGeorge > is in the sources |
| 02:53.45 | ivesti | hello,I'm trying to configure a sipura spa-300 with a dock-n-talk device.I have both of them working individually but I'm trying to connect them so the spa-picks up the incoming call from the cellphone.Can anybode help me? |
| 02:53.49 | *** join/#asterisk r0d3nt|m (nobody@wsip-24-234-241-84.lv.lv.cox.net) |
| 02:53.54 | l-fy | just take it and compile it, and don't be lazy |
| 02:54.01 | FuriousGeorge | in what sources? what do you mean? |
| 02:54.11 | FuriousGeorge | i uise gentoo |
| 02:54.24 | l-fy | FuriousGeorge > yate have a gentoo emerge |
| 02:54.34 | FuriousGeorge | emerge (what) |
| 02:55.28 | cbachman | ivesti, I just have a spa-3k, but what issues are you seeing? |
| 02:55.28 | FuriousGeorge | what is the name of a softphone for linux with a gui |
| 02:55.28 | l-fy | yate |
| 02:55.28 | FuriousGeorge | i jsut installed gnophone and when i execute it all it says is no sound and nothing happens |
| 02:55.33 | FuriousGeorge | YATE |
| 02:55.44 | FuriousGeorge | sorry |
| 02:55.44 | ivesti | cbachman, how do I connect the cables so the spa picks up the incoming pstn call and forwards it to the sip provider? |
| 02:55.54 | FuriousGeorge | i missed it |
| 02:56.52 | l-fy | n/p |
| 02:57.07 | cbachman | ivesti, Hmmm... one question first. Can you hook up the spa to the pstn and use it? |
| 02:57.27 | l-fy | FuriousGeorge > is a small client but if you know some qt you can extend that fast |
| 02:57.35 | Qwell | cbachman: with the 3000 you can |
| 02:58.08 | cbachman | qwell, I misphrased that. I should have asked if he had that already working. I have a 3k :-) |
| 02:58.26 | Qwell | cbachman: ahh |
| 02:59.21 | *** join/#asterisk dmccollum (~dmccollum@69-164-245-72.atlaga.adelphia.net) |
| 02:59.21 | FuriousGeorge | I-fy: zapchan.cpp:593: error: array bound forbidden after parenthesized type-id |
| 02:59.36 | Qwell | FuriousGeorge: iaxcomm is good too |
| 02:59.36 | FuriousGeorge | i have gcc 3.4 |
| 03:00.25 | l-fy | FuriousGeorge > just ignore zapchan, you don't need it anyway |
| 03:00.40 | FuriousGeorge | hwo does one ignore zapchan? |
| 03:01.08 | l-fy | FuriousGeorge > use configure for that |
| 03:01.15 | dmccollum | I'm trying to get the wake-up call feature to work and need a bit of help. Here's the output of the file the agi creates and puts in the outgoing directory. It clears the file, but never calls the extension. Does the contents of the file look correct? |
| 03:01.16 | *** join/#asterisk Half_Dome (~jelway@mail.westmarkinc.com) |
| 03:01.24 | dmccollum | channel: Local/2001@from-internal |
| 03:01.24 | dmccollum | maxretries: 3 |
| 03:01.24 | dmccollum | retrytime: 60 |
| 03:01.24 | dmccollum | waittime: 60 |
| 03:01.24 | dmccollum | callerid: "WakeUp" <77> |
| 03:01.24 | dmccollum | application: MusicOnHold |
| 03:01.26 | dmccollum | data: |
| 03:02.05 | *** join/#asterisk techie (gus@asterisk.horizonte.us) |
| 03:02.18 | *** join/#asterisk _Sam-- (sam@ns2.kneedraggers.com) |
| 03:02.37 | Half_Dome | Does anyone know why my Digium FXO is trying to use IRQ5 which is being used by integrated sound? I don't want to disable sound and don't think I can change it. |
| 03:03.13 | dmccollum | it's just sharing the int. That's usually normal if you have several PCI cards in the system. |
| 03:03.35 | dmccollum | You can try moving it to another PCI slot. |
| 03:03.50 | Half_Dome | I'm getting modprobe: FATAL: error running install command for wctdm and I ASSUMED it was an IRQ conflict |
| 03:03.54 | dmccollum | you might be able to select the IRQ it uses in the BIOS. |
| 03:04.01 | *** join/#asterisk al__2 (~ldli6@222.124.66.111) |
| 03:04.16 | al__2 | help on net2phone setting for asterisk@home |
| 03:04.34 | al__2 | I got this sip.conf |
| 03:04.34 | al__2 | [general] |
| 03:04.34 | al__2 | useragent = X-Lite release 1103m |
| 03:04.34 | al__2 | register => PHONENUMBER:PASSWORD@sip.net2phone.com |
| 03:04.34 | al__2 | [net2phone] |
| 03:04.34 | al__2 | type = peer |
| 03:04.36 | al__2 | host = sip.net2phone.com |
| 03:04.38 | al__2 | username = PHONENUMBER |
| 03:04.40 | al__2 | secret = PASSWORD |
| 03:04.42 | al__2 | fromuser = PHONENUMBER |
| 03:04.44 | al__2 | fromdomain = net2phone.com |
| 03:04.46 | al__2 | context = incoming |
| 03:04.48 | al__2 | insecure = very |
| 03:04.50 | al__2 | canreinvite = no |
| 03:04.50 | Half_Dome | Can you manually edit /proc/interrupts? |
| 03:04.52 | al__2 | extensions.conf |
| 03:04.54 | al__2 | [outgoing] |
| 03:04.56 | al__2 | exten => _9NXXNXXXXXX,1,Dial(SIP/net2phone/${EXTEN:1}) |
| 03:04.58 | al__2 | should i just paste it into sip.conf and extensions.conf ? |
| 03:04.59 | Qwell | ~pastebin |
| 03:05.00 | jbot | from memory, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 03:05.05 | Nivex | /kick al__2 |
| 03:05.36 | _Sam-- | if we have multiple agents, and multiple call queues....we are using the agent callback feature so the agent gets rung before the call from the queue gets connected... |
| 03:05.42 | _Sam-- | how can we tell the agent which queue is calling |
| 03:05.54 | l-fy | _Sam--> calledid? |
| 03:06.05 | _Sam-- | nope they all call in on one main number |
| 03:06.08 | *** join/#asterisk t0p (t0p@tech-mgr.chatri.com) |
| 03:06.16 | _Sam-- | then get put into the queues based on what they press |
| 03:06.22 | _Sam-- | "press 1 for sales " <salesq>... |
| 03:06.23 | _Sam-- | etc |
| 03:06.49 | _Sam-- | but some agents are part of 3 queues... |
| 03:07.18 | _Sam-- | they get rung and dont know which queue is ringing them until AFTER they accept the call from the q |
| 03:09.39 | *** join/#asterisk mhnoyes (~mhnoyes@user-2ivfnjb.dialup.mindspring.com) |
| 03:11.50 | Qwell | Would something like this work on an FXS port of a tdm400p? http://tinyurl.com/4xuxl |
| 03:11.58 | Wazb | i tried to register Asterisk to another SIP server , i am getting "1.1.1.1 does not implement 'REGISTER'" error , Please help!!! |
| 03:13.16 | tainted- | Wazb which provider |
| 03:14.26 | tainted- | any PBX gurus |
| 03:14.46 | tainted- | gottago needs help with 5ess switch... |
| 03:14.54 | tainted- | connecting to * |
| 03:15.23 | mgth | anyone have good recomendation for colo servers? |
| 03:15.55 | *** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
| 03:15.55 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 03:18.37 | cbachman | gottago, what are you connecting to it with? |
| 03:20.41 | Wazb | <tainted-> actually that SIP server is normal SIP server (only it requires IP address of client in order to register client in its configuration) |
| 03:20.50 | MrBelvedr | I am using the Manager API to originate calls. The first call always goes through successfully |
| 03:21.26 | *** join/#asterisk mishehu (mishehu@cshells.shavedgoats.net) |
| 03:21.40 | MrBelvedr | The second call *always fails* |
| 03:21.59 | MrBelvedr | anyone have any ideas why this could be happening |
| 03:22.18 | *** join/#asterisk TheEmperor (~mattn@203.121.47.100) |
| 03:22.23 | _Sam-- | what are you connecting to the manager with, php? |
| 03:22.35 | MrBelvedr | the CLI is not giving any useful information other than iax2_read: I should never be called |
| 03:22.47 | MrBelvedr | i am connecting to the manager with C# |
| 03:23.01 | MrBelvedr | it is connecting fine |
| 03:23.06 | tainted- | Wazb which provider is it? |
| 03:23.57 | tainted- | bkw_ yo |
| 03:24.10 | *** join/#asterisk slePP (~slepp@S01060040f48412ad.ed.shawcable.net) |
| 03:24.39 | MrBelvedr | I can see from the CLI that the Manager received command 'Originate' |
| 03:24.55 | MrBelvedr | but the second call never goes through |
| 03:25.21 | MrBelvedr | anyone have any idea why only the first call would be going through. After I restart asterisk the first call always goes trough |
| 03:25.38 | jontow | has anyone done anything with queue prioritization ? |
| 03:28.08 | *** join/#asterisk CosmicRay (~jgoerzen@2002:4545:7206:1:20e:a6ff:fe5c:55e1) |
| 03:30.33 | AgiNamu | Mr B, you have a C# Manager library? |
| 03:30.36 | *** join/#asterisk heison (~heison@CPE000a01d49e6f-CM00111a59bec2.cpe.net.cable.rogers.com) |
| 03:31.37 | MrBelvedr | no |
| 03:31.43 | MrBelvedr | i built the library myself |
| 03:31.46 | bkw_ | AgiNamu, dude.. you better RUUUUUUUUUUUN |
| 03:31.51 | AgiNamu | why? |
| 03:31.51 | MrBelvedr | if you want the code I can send it |
| 03:31.57 | AgiNamu | sure: mgg@Atrevido.net |
| 03:31.59 | bkw_ | boot -> ass |
| 03:32.12 | AgiNamu | boot = doesnt like information |
| 03:32.24 | AgiNamu | out of respect to mark, i took the tutorial down, pending further review. |
| 03:32.32 | bkw_ | ;) |
| 03:32.38 | bkw_ | i'm just givin jew crap |
| 03:32.40 | AgiNamu | I find it funny though |
| 03:32.43 | Qwell | tutorial? :p |
| 03:33.40 | *** join/#asterisk lately (~dougb@chi.econ.usyd.edu.au) |
| 03:33.51 | AgiNamu | I say something like "If a variable is initialized with xor edx, edx, try replacing those bytes with mov dh, 0ff" |
| 03:33.53 | AgiNamu | and get kicked |
| 03:34.00 | bkw_ | it was on how to make pigs fly |
| 03:34.02 | bkw_ | ya ya thats it |
| 03:34.05 | bkw_ | NEXT!!! |
| 03:34.06 | Qwell | ahh |
| 03:34.09 | AgiNamu | lol |
| 03:34.30 | MrBelvedr | i sent it Aginamu |
| 03:34.41 | AgiNamu | thanks! |
| 03:35.15 | MrBelvedr | np. Can anybody tell me how to debug why only the second call is not being successfully correctly |
| 03:35.24 | phantasis | analog phones are fxs or fxo? |
| 03:35.26 | MrBelvedr | not being originated successfully |
| 03:38.44 | MrBelvedr | help please |
| 03:40.45 | *** join/#asterisk poli (~poli@200-168-30-125.dsl.telesp.net.br) |
| 03:46.26 | Qwell | So, would something like this work on an FXS port of a tdm400p? http://tinyurl.com/4xuxl |
| 03:46.32 | Qwell | You'll have to excuse my stupidity today |
| 03:47.16 | cbachman | qwell, yes. |
| 03:49.28 | MrBelvedr | has anybody ever gotten a phone call where your phone was ringing, and you did not pick up yet, but you can hear the caller speaking (they don't realize you can hear them) |
| 03:49.52 | Qwell | MrBelvedr: sounds like a broken phone, heh |
| 03:50.07 | MrBelvedr | for some reason it happens sometimes on my cell phone |
| 03:50.23 | Qwell | cell, see, thats another story... |
| 03:50.32 | MrBelvedr | it is kinda funny to eavesdrop on them |
| 03:51.22 | MrBelvedr | it is a real hoot and holler |
| 03:51.30 | Qwell | They can't hear you? |
| 03:51.34 | MrBelvedr | correct |
| 03:52.21 | MrBelvedr | well actually i am not sure they can hear me or not |
| 03:52.26 | MrBelvedr | i never tried to find out |
| 03:52.29 | Qwell | I would. heh |
| 03:52.31 | DEEZED | anyone know of a quick way to change sip caller ID? Like to type in *67 as an extension and get a dial tone, then type in the caller id you want, then get another dial tone for the number you want to call? |
| 03:52.37 | Qwell | You won't be getting billed for the call. |
| 03:52.51 | AgiNamu | DEEZED, "SetCAllerIdNum" ? |
| 03:52.51 | MrBelvedr | so far when it happened it was like cc company calling or something like htat |
| 03:53.13 | DEEZED | oh ok.. so that works for out going calls as well? |
| 03:53.48 | Qwell | DEEZED: thats not a bad idea actually |
| 03:54.07 | Qwell | *67, *1, number |
| 03:54.12 | DEEZED | i stumbled upon caller id spoofing by mistake |
| 03:54.14 | Qwell | *1 will lookup a saved CID :p |
| 03:54.15 | DEEZED | now im hooked |
| 03:55.38 | *** join/#asterisk SuPrSluG (~SuPrSluG@pool-129-44-136-89.buff.east.verizon.net) |
| 03:55.47 | SuPrSluG | hi |
| 03:56.09 | MrBelvedr | what does "Ooh, voice format changed to 4" mean? |
| 03:56.27 | MrBelvedr | some of these info messages are a real hoot and holler |
| 03:56.30 | MrBelvedr | heh |
| 03:56.42 | SuPrSluG | i'm getting sonar like pings on my zaptel device. Is it an IRQ problem? |
| 03:57.05 | marlowe | cid spoofing is evil deezed |
| 03:57.10 | marlowe | DEEZED : http://pastebin.ca/8504 |
| 03:57.20 | SuPrSluG | doesn't happen on iax or sip |
| 03:57.46 | DEEZED | marlow your the man |
| 03:57.55 | DEEZED | marlowe you too |
| 03:57.59 | marlowe | thats the first thing i wrote.. BEFORE my companyies ivr |
| 03:58.03 | marlowe | err companies |
| 03:58.04 | marlowe | lol |
| 04:01.53 | *** join/#asterisk bl1tz (~brian@adsl-67-114-145-205.dsl.sntc01.pacbell.net) |
| 04:04.49 | Wazb | i am forwarding a call to SIP server , i need to send # at the end of each dialed number, How can i configure this? |
| 04:05.31 | AgiNamu | Mr B, it mean the voice format changes to #4 |
| 04:05.34 | AgiNamu | show codecs |
| 04:05.37 | AgiNamu | I think 4 is GSM |
| 04:05.38 | AgiNamu | or ULAW |
| 04:10.07 | *** join/#asterisk Legend (~Legend@24.244.142.133) |
| 04:10.46 | Moc | Anyone have access to DL polycom firmware on their site ? |
| 04:12.30 | *** join/#asterisk loud (~ariel@null0.flapping.net) |
| 04:16.39 | *** join/#asterisk mog_home (~mogorman@146.229.187.64) |
| 04:16.56 | MrBelvedr | is GSM a codec or a type of cellular network? |
| 04:17.08 | bkw_ | ding ding ding |
| 04:17.10 | bkw_ | thanks for playing |
| 04:17.11 | bkw_ | NO |
| 04:17.17 | bkw_ | its a codec too |
| 04:17.18 | Moc | gsm is a codec in voip world |
| 04:17.31 | MrBelvedr | ok |
| 04:17.33 | MrBelvedr | thx |
| 04:17.38 | bkw_ | GSM is cellular too |
| 04:17.48 | bkw_ | they are the same codec gsm-fr |
| 04:17.50 | MrBelvedr | that is what i thought |
| 04:18.03 | bkw_ | you also have gsm-efr or some shit too |
| 04:18.06 | bkw_ | but that has patents |
| 04:18.13 | bkw_ | so they can take flying LEEEEEEEEAAAP |
| 04:18.15 | *** join/#asterisk Shido6 (~greg@d57-87-253.home.cgocable.net) |
| 04:18.26 | MrBelvedr | k |
| 04:18.29 | Moc | can't find the damn polycom ip 3000 firmware !!! |
| 04:18.37 | bkw_ | polycom.com |
| 04:18.43 | bkw_ | </smartass> |
| 04:18.51 | Moc | I dont have an access for them .. |
| 04:19.32 | NewSole | anyone know why I ma getting this... |
| 04:19.34 | NewSole | RFC3389 support incomplete. Turn off on client if possible |
| 04:19.43 | bkw_ | turn off VAD |
| 04:19.50 | bkw_ | aka silence suppression |
| 04:19.51 | NewSole | from "Zap -> IAX2 -> SIP |
| 04:20.00 | bkw_ | xlite=Transmit Silence => Yes |
| 04:20.02 | bkw_ | and next time GOOGLE |
| 04:20.08 | NewSole | where |
| 04:20.19 | bkw_ | that question is answered in the first few lines of the qquery if you go to google |
| 04:20.22 | Moc | in the config |
| 04:20.29 | bkw_ | what device? |
| 04:20.39 | NewSole | TDM400P |
| 04:20.45 | bkw_ | no the sip device |
| 04:20.57 | *** join/#asterisk topping (~topping@24.210.82.196) |
| 04:21.00 | NewSole | no... sip is to provider |
| 04:21.04 | bkw_ | who? |
| 04:21.08 | NewSole | MCI |
| 04:21.10 | bkw_ | you're SOL |
| 04:21.12 | bkw_ | NEXT!!! |
| 04:21.26 | NewSole | ?? |
| 04:21.31 | bkw_ | you're shit out of luck |
| 04:21.40 | bkw_ | I don't expect them to turn off VAD |
| 04:21.43 | bkw_ | on their end |
| 04:21.46 | NewSole | I know what SOL is... but why |
| 04:21.55 | bkw_ | its MCI dude.. like they care |
| 04:22.03 | bkw_ | VAD saves them bandwidth |
| 04:22.11 | bkw_ | and lots of it if you do alot of calls |
| 04:22.11 | *** join/#asterisk iguy (~iguy@dsl093-197-234.mke1.dsl.speakeasy.net) |
| 04:22.22 | bkw_ | only transmitting when you actually talk |
| 04:22.26 | NewSole | is there not a way to adjust asterisk to support it |
| 04:22.30 | poli | I have tried a number of clients for linux, including minisip, iaxcomm, kphone and linphone. kphone and linphone are constantly giving me sig11, so I couldn't really test sound quality but as for IAXComm and MiniSIP, both using GSM, I get terrible mic sound. Has anyone had this sort of experience? |
| 04:22.31 | bkw_ | nope |
| 04:22.56 | bkw_ | poli, if you're getting sig11 you're hardware might be bad |
| 04:22.56 | Qwell | poli: iaxcomm can do more then just gsm |
| 04:23.04 | NewSole | hmmm... ok... so I will always get that Error |
| 04:23.11 | bkw_ | NewSole, yes |
| 04:23.17 | bkw_ | hold on |
| 04:23.32 | bkw_ | ok |
| 04:23.35 | bkw_ | you just broke the rules |
| 04:23.41 | bkw_ | NOTICE != warning |
| 04:23.43 | NewSole | me... |
| 04:23.56 | bkw_ | people call debug logs and notices.. WARNINGS |
| 04:24.00 | bkw_ | when infact they are not |
| 04:24.01 | bkw_ | hehe |
| 04:24.09 | MrBelvedr | hey where can I peek at some high quality code that originates a call using the Manager API |
| 04:24.12 | CoaxD | Holy Christ, Akon ROCKS |
| 04:24.15 | NewSole | so I can ignore it then |
| 04:24.25 | bkw_ | NewSole, or turn off warning in logger.conf |
| 04:24.41 | poli | bkw_, Hardware is a good 2003 Toshiba satellite, the problem is Linux itself. No matter what codec I use, it seems to a problem with the ADC or something, as even when I unplug the mic, with only the other end speaking, sound echos back and increases intensity. It is a terrible experience. |
| 04:24.49 | CoaxD | I mean, I'm not a big fan of "black rap/pop" (forgive any uninsinuated racial slurr there), but holy shit, his "Mr. Lonely" remix is one one of the best orchestrated songs i've ever heard |
| 04:24.56 | bkw_ | poli, liinux is not a desktop OS |
| 04:24.59 | NewSole | bkw... question.... |
| 04:25.00 | *** join/#asterisk dersteer (~travis@24.231.151.119.gha.mi.chartermi.net) |
| 04:25.12 | CoaxD | bkw: Thats a rather flameworthy statement there |
| 04:25.15 | NewSole | you know when you use ChanIsAvail |
| 04:25.16 | bkw_ | so |
| 04:25.20 | poli | bkw_, I use it in my desktop and notebook for 10 years now. :) So, I respectfully disagree. |
| 04:25.20 | CoaxD | bkw: (it should be noted that I agree with you.) |
| 04:25.31 | bkw_ | poli, thats ok... I atleast its not windows |
| 04:25.35 | bkw_ | and i totally respect that |
| 04:25.42 | bkw_ | I just will never use it on my desktop ;) |
| 04:25.54 | CoaxD | bkw; I concur; me either |
| 04:26.01 | poli | bkw_, I am going to test gnomemeeting with H323. |
| 04:26.02 | CoaxD | bkw: most i use it for is loading X so I can use a browser |
| 04:26.11 | CoaxD | bkw; it could never replace a real desktop |
| 04:26.12 | bkw_ | I like my mac's |
| 04:26.18 | MrBelvedr | where is some good code to look at that uses Manager api |
| 04:26.18 | bkw_ | s/like/totally LOVE/ |
| 04:26.29 | CoaxD | bkw: (tho again, many people disagree; there are billions of lines of code to make it a decent desktop OS) |
| 04:26.33 | bkw_ | MrBelvedr, dude go look on voice-info.org |
| 04:26.41 | bkw_ | decent is not good |
| 04:26.47 | CoaxD | bkw: agreed. |
| 04:26.48 | CoaxD | bkw: :) |
| 04:26.49 | NewSole | is there not like a GroupisAvail... or plans to be... epecial when calling a group of devices |
| 04:26.49 | bkw_ | I don't wanna dick with my os for hours to make it work |
| 04:26.58 | CoaxD | bkw: ME EITHER! :) |
| 04:26.58 | bkw_ | NewSole, no clue |
| 04:27.02 | CoaxD | bkw: I honestly just use Windows |
| 04:27.07 | bkw_ | my mac works great |
| 04:27.08 | NewSole | maybe an idea |
| 04:27.08 | CoaxD | bkw: I'm okay with OSX, whatever |
| 04:27.14 | bkw_ | I don't miss windows one bit |
| 04:27.15 | CoaxD | bkw: As long as it actually acts like a desktop |
| 04:27.29 | bkw_ | i was running asterisk on my ibook |
| 04:27.31 | bkw_ | it was fun |
| 04:27.32 | bkw_ | fun fun fun |
| 04:27.34 | bkw_ | :P |
| 04:27.57 | tessier | bkw_: I've been using Linux more and more without command line browsing web pages and opening MS office documents and spreadsheets and viewing pdf's and movies and all kinds of normal desktop-like things. I have been using Linux since 93 and I am quite impressed with how it is progressing on the desktop front. |
| 04:28.01 | CoaxD | bkw: Heh |
| 04:28.19 | CoaxD | tessier: Thats just it, sir. Its 'progressing' |
| 04:28.21 | bkw_ | tessier, it still has a long way to go |
| 04:28.34 | bkw_ | alot of the fuglyness must be totally hidden |
| 04:28.49 | CoaxD | tessier: Think about Windows in 1993. Windows 3.1. Windows 3.1 was about 1000x more usable than even the most good desktop manager for linux |
| 04:28.50 | bkw_ | ok people have problems using windows... you think they can get linux? |
| 04:28.58 | AgiNamu | bkw_ damn striaght |
| 04:29.01 | tessier | CoaxD: It's a lot further along than Windows 3.1 was back when it was considered a good desktop OS that's for sure. |
| 04:29.03 | AgiNamu | Windows XP is HARD for some folks. |
| 04:29.04 | CoaxD | tessier: (okay that may be pushing it, but i'm sure you understand my point) |
| 04:29.11 | *** join/#asterisk shepherd (~matt@dhcp-166.digium.com) |
| 04:29.14 | bkw_ | "what do you mean right click?" |
| 04:29.18 | chap | I don't want to try to support my family members on linux. |
| 04:29.22 | tessier | bkw_: Windows still has a long way to go. I don't think it's as far behind as you think. |
| 04:29.31 | AgiNamu | Linux has another problem with hte desktop: By the time they get to Win95-level, MS will be on longhorn |
| 04:29.31 | tessier | chap: I support family on Linux. It works great. |
| 04:29.32 | bkw_ | OS X is there |
| 04:29.39 | AgiNamu | and Lonhorn, like it or not.... is pretty amazing. |
| 04:29.42 | chap | bkw: Or, when you tell them to do a right-click, that doesn't mean right-click on everything else from then on.. |
| 04:29.44 | CoaxD | Windows XP is a very, very good desktop OS. (The *only* thing busted with it right now is the goddamn security issues.) |
| 04:29.54 | bkw_ | I hate when you do support and you get a "reader" |
| 04:29.57 | CoaxD | And quite frankly, Linux has had more fscking exploits than windows lately |
| 04:29.58 | AgiNamu | just the graphical capabilites alone are freaking awesome. NATIVE 128-bit colour |
| 04:29.59 | bkw_ | they MUSt read you everything on the fucking screen |
| 04:30.05 | AgiNamu | 3D composition |
| 04:30.05 | CoaxD | (aside from OE. OE is 1 big exploit.) |
| 04:30.07 | AgiNamu | etc. etc. all built in. |
| 04:30.13 | bkw_ | LookOut Express |
| 04:30.21 | chap | tessier: which distro? Seriously interested here. |
| 04:30.25 | tessier | CoaxD: I haven't had to fix an exploit on my Fedora box. Which exploits are you referring to? |
| 04:30.36 | tessier | chap: RedHat all along. Fedora lately. |
| 04:30.40 | bkw_ | AgiNamu, they are too late with it.. its too resource hungry.. and i'm not gonna have an OS that ties me to some registration.. activation bullshit. |
| 04:30.48 | AgiNamu | yea, YOU arent. |
| 04:30.56 | bkw_ | did they finally wake up? |
| 04:30.57 | CoaxD | tessier: There have been numerous kernel exploits within the last couple months. |
| 04:30.57 | AgiNamu | But most people will just go along with it. half of them dont even notice its there. |
| 04:31.02 | *** join/#asterisk znoG (gs@200.115.216.109) |
| 04:31.07 | tessier | CoaxD: Those are local only right? |
| 04:31.13 | CoaxD | tessier: Correct. |
| 04:31.14 | bkw_ | well I love my Mac so glad i switched |
| 04:31.22 | bkw_ | the problem is.. I can't even keep a win32 box clean for 4 hours |
| 04:31.24 | bkw_ | with all the updates |
| 04:31.25 | tessier | CoaxD: Not a big deal at all on a single user box. Sitting in front of the console is one big exploit. |
| 04:31.30 | bkw_ | so something is very wrong |
| 04:31.34 | poli | AgiNamu, Linux desktop is now waaaayyyyy better than Windows'. Much much better. That is definatly not the problem with Linux. The main problem is hardware support. |
| 04:31.45 | AgiNamu | poli, what are you smoking? |
| 04:31.45 | CoaxD | tessier: ...except that i own an ISP, and need to keep the boxen secure. :) |
| 04:31.49 | tessier | Nobody seems to look for local exploits in Windows. Why is that? |
| 04:31.52 | bkw_ | tessier, no tellin.. |
| 04:31.57 | tessier | CoaxD: Well that is a very special case then. |
| 04:31.59 | CoaxD | tessier: Nonetheless, point taken. |
| 04:32.05 | AgiNamu | tessier, lots of people do |
| 04:32.07 | AgiNamu | there's been plenty |
| 04:32.11 | chap | poli: And software that people like on windows that they want to run on linux. People want to go to the store, buy something, and run it. |
| 04:32.15 | *** join/#asterisk montoya (~montoya@200.195.84.93) |
| 04:32.16 | tessier | CoaxD: All of my desktops are single user for the most part. So local exploits are not a problem. |
| 04:32.18 | CoaxD | tessier: and note, there's a very big difference between 'local access' and 'box access' |
| 04:32.18 | AgiNamu | But, the thing is, a lot of vendors for Windows SUCK. they require admin rights. |
| 04:32.24 | tessier | I occasionally lgin to them remotely. |
| 04:32.34 | CoaxD | tessier: you may have 100,000 users. and they're not 'local' to the box |
| 04:32.42 | bkw_ | haha |
| 04:32.43 | CoaxD | tessier: 'local access' in linux means instant root. period |
| 04:32.47 | tessier | CoaxD: Right. |
| 04:32.49 | AgiNamu | local exploits are still very critical on windows, although only in certain situations |
| 04:32.54 | bkw_ | actually |
| 04:32.56 | CoaxD | tessier: ctrl-alt-del ; 'linux init=/bin/bash' |
| 04:32.58 | poli | chap, true. hardware support and people-friendly software (not user-friendly, linux has plenty of them already) |
| 04:32.59 | bkw_ | I don't hate any OS really |
| 04:33.04 | bkw_ | I use what gets the job done |
| 04:33.05 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-179-40-134.cust.bezeqint.net) |
| 04:33.06 | tessier | Where do you keep that stick anyway? |
| 04:33.19 | AgiNamu | CoaxD, that's what Encrypted File System is for |
| 04:33.21 | poli | AgiNamu, You should broaden your mind before stating that. |
| 04:33.25 | shepherd | haha.. who cares |
| 04:33.25 | bkw_ | and if you're one of those linux people that thinks linux is the answer to EVERYTHING.. I hate to break it to ya.. IT IS NOT THE END ALL |
| 04:33.30 | CoaxD | aginamu: Yes, except that i agree with the concept entirely |
| 04:33.31 | tessier | poli: You want him to smoke some weed? |
| 04:33.35 | AgiNamu | poli, I broadened it a lot. spent a few days recovering. |
| 04:33.43 | AgiNamu | still take a xanax every now and then too |
| 04:33.43 | poli | AgiNamu, Linux gives you the option to choose your desktop (WIndows does NOT) so, try looking into really good ones. |
| 04:33.45 | CoaxD | aginamu: If I have access to a box (physically), i should be able to get at the information stored within it |
| 04:33.50 | bkw_ | linux has its place... solaris has its place... *bsd has its place |
| 04:33.52 | CoaxD | aginamu: With *very* notable exceptions |
| 04:33.56 | poli | tessier, No, just google a little. :)P |
| 04:33.56 | bkw_ | what is nice we all have such a choice of OS's |
| 04:33.57 | AgiNamu | poli: get your facts straight. you don't even need to sue Win32 with Windows. |
| 04:34.01 | AgiNamu | you can get rid of explorer, etc. |
| 04:34.06 | bkw_ | just think back in the day.. the choice wasn't all that great |
| 04:34.08 | AgiNamu | and get this: Choosing your desktop is BAD |
| 04:34.09 | CoaxD | bkw; Work with UnixWare. Then see if you can say the same thing |
| 04:34.18 | CoaxD | bkw: Or perhaps, MPRAS |
| 04:34.19 | AgiNamu | the more choices mean more testing, more problems, more user education |
| 04:34.20 | bkw_ | well maybe unixware is the exception |
| 04:34.25 | tessier | bkw_: As long as the OS's try to remain compatible while still sticking to their individual philosophies that make them unique I agree that choice is a good thing. |
| 04:34.26 | poli | AgiNamu, my choise is my choise, yours is yours, but look before you talk, as I look before I do. |
| 04:34.30 | CoaxD | bkw: You really should narrow your search down a bit there, sir :) |
| 04:34.34 | tessier | But then there's MS and occasionally Apple... |
| 04:34.36 | CoaxD | bkw: *pinches your pretty little ass* |
| 04:34.41 | bkw_ | hehe |
| 04:34.55 | CoaxD | bkw: Ever ran PIC/OS? :) :) |
| 04:35.00 | bkw_ | dual G5 baby |
| 04:35.02 | bkw_ | thats on my shopping list |
| 04:35.07 | bkw_ | with 30 inch display |
| 04:35.11 | CoaxD | bkw: PICK was *DAMN* wonderful. ...except it made me feel like shooting myself in the head |
| 04:35.11 | bkw_ | muhahahhahahaha |
| 04:35.18 | AgiNamu | Poli, sure it works for you. it's not a "desktop OS" that is good at being prepackaged and shipped to 100 million desktops |
| 04:35.19 | shepherd | when is asterisk going to be its own os? |
| 04:35.22 | shepherd | hehe |
| 04:35.26 | bkw_ | shepherd, NEVAH |
| 04:35.29 | AgiNamu | anyways, gonna go teach my wife how to drive. |
| 04:35.31 | bkw_ | mark loves linux too much |
| 04:35.31 | AgiNamu | cya later |
| 04:35.31 | CoaxD | shepherd: That would be rather dumb, dont you think? |
| 04:35.36 | shepherd | nah |
| 04:35.36 | CoaxD | AgiNamu: Wait |
| 04:35.45 | bkw_ | I don't really expect asterisk to work really well on anything but linux for some time |
| 04:35.48 | CoaxD | AgiNamu: You cant just go and make a statement like that, and leave with no explanation! |
| 04:35.51 | shepherd | then there wouldn't be these stupid debates |
| 04:35.53 | CoaxD | bkw: Nor should it have to |
| 04:35.54 | shepherd | in here |
| 04:35.55 | poli | AgiNamu, Well, have you tried before stating that? I have tried Windows. :) |
| 04:36.06 | CoaxD | AgiNamu: Perhaps you could just pass a couple sentences of explanation with that?? |
| 04:36.08 | AgiNamu | that doesnt need an explanation |
| 04:36.19 | AgiNamu | go set linux up |
| 04:36.22 | AgiNamu | then go set windows up |
| 04:36.23 | CoaxD | AgiNamu: Why doesntw your wife know how to drive?? or you just dont LIKE HOW she drives, so yer gonna beat her until she does it right? :)| |
| 04:36.25 | bkw_ | hahaha |
| 04:36.25 | AgiNamu | compare the setup for a GUI |
| 04:36.29 | bkw_ | AgiNamu, has a point |
| 04:36.30 | AgiNamu | OH, tAHT statement |
| 04:36.31 | AgiNamu | lol |
| 04:36.32 | poli | AgiNamu, There are linux distributions like Knoppix for example that are simply 100% desktop-oriented. You should take a look, boots fromthe CD and uses no HD. |
| 04:36.35 | AgiNamu | No, she doesnt really know. |
| 04:36.51 | CoaxD | AgiNamu: "DRIVE, BITCH!#@$ *SMACK* NO BITCH, THATS THE GAS! NOT THE BRAKES! *SMACK*" |
| 04:36.55 | AgiNamu | poli, I'm not saying there ARENT any desktop ones. I'm saying it is not cut out to be; it's not mature enough |
| 04:36.56 | AgiNamu | LOl |
| 04:36.57 | bkw_ | poli, I just don't like how the window managers out there look |
| 04:37.02 | poli | AgiNamu, After trying, I fully accept your opinions, but not before that. :) |
| 04:37.04 | bkw_ | none really look that good |
| 04:37.04 | topping | "users need education" implies you have sysadmins to do the installs |
| 04:37.09 | AgiNamu | I've tried |
| 04:37.13 | bkw_ | it must be sexy |
| 04:37.15 | AgiNamu | I've tried Linux on the desktop since 96 |
| 04:37.16 | bkw_ | it must be easy to use |
| 04:37.16 | shepherd | come on people, we have to make it easy for people to put in the serial code to verify that we actually bought the product! |
| 04:37.20 | CoaxD | AgiNamu: She just getting her license or something?? heh |
| 04:37.21 | *** join/#asterisk shmaltz (~chatzilla@69.28.255.210) |
| 04:37.23 | poli | bkw_, HAve you seen Blanes? A Windows 2000 clone? |
| 04:37.31 | shmaltz | anybody here using polycom phone? |
| 04:37.32 | topping | if you have sysadmins, they can choose the desktop of the users |
| 04:37.32 | AgiNamu | no, its more like "Shit! This has power steering, no need to turn so sharp! Ahh! telefone poll, let me have the wheel! .... whew.... ok.... ill drive home" |
| 04:37.33 | bkw_ | poli, show me baby |
| 04:37.36 | AgiNamu | she has her license. |
| 04:37.41 | AgiNamu | Guatemala is not very stringent. |
| 04:37.46 | CoaxD | AgiNamu: Um |
| 04:37.51 | CoaxD | AgiNamu: Ohhhhhh! |
| 04:38.03 | CoaxD | AgiNamu: That isn't so unreasonable |
| 04:38.03 | AgiNamu | so im waiting until 2230 when there is no traffic |
| 04:38.07 | CoaxD | AgiNamu: Just a new car :) |
| 04:38.12 | tessier | The year 2230? |
| 04:38.14 | AgiNamu | so if she happens to mistake the clutch for the break on a hill |
| 04:38.15 | AgiNamu | LOL |
| 04:38.17 | bkw_ | foudn it poli |
| 04:38.19 | poli | bkw_, http://www.vivaolinux.com.br/screenshots/verScreenshot.php?codigo=2334 |
| 04:38.23 | AgiNamu | we wont roll back into someone's mercedes. |
| 04:38.23 | CoaxD | AgiNamu: AHHHH! Its a new car, and a STICK! |
| 04:38.25 | MrBelvedr | does anybody in here want to offer me asterisk hosting for really cheap? |
| 04:38.33 | CoaxD | AgiNamu: Go with God, sir |
| 04:38.38 | AgiNamu | oh hell yea. Automatics suck. |
| 04:38.46 | AgiNamu | dunno how anyone can drive an automatic. |
| 04:38.49 | topping | MrBelvedr: would you like root access with that too? |
| 04:38.53 | tessier | AgiNamu: Why? I've always preferred stick but... |
| 04:38.57 | CoaxD | AgiNamu: She must be very understanding to have let you buy a new car with a manual. *g* |
| 04:38.58 | bkw_ | poli, thts not bad |
| 04:39.00 | AgiNamu | not much control |
| 04:39.13 | CoaxD | AgiNamu: I prefer a manual as well. But am currently driving an auto |
| 04:39.14 | tessier | The problem with a stick is that if you want to get freak with a chick in the car you've got that stick right there between you... |
| 04:39.15 | MrBelvedr | I don't need a voip provider, just a asterisk host |
| 04:39.20 | CoaxD | Too bad most cars coming out today dont have a manual option |
| 04:39.24 | AgiNamu | tessier, she's pregant :) |
| 04:39.25 | bkw_ | poli, its got the dumbest name on the planet |
| 04:39.27 | bkw_ | but kewl |
| 04:39.27 | CoaxD | everybody wants auto |
| 04:39.30 | tessier | And that's one stick too many afaiac. |
| 04:39.30 | AgiNamu | so thats a bigger obstacle. |
| 04:39.32 | dmccollum | Can someone tell me what's wrong with this file for the /var/spool/asterisk/outgoing? |
| 04:39.34 | poli | bkw_, There is some XP-clone going on somewhere, but I really think thoses clones are to "attract" people. You can build a much more functional (and beatiful) desktop using other software. |
| 04:39.35 | tessier | AgiNamu: Doh. |
| 04:39.39 | dmccollum | [root@asterisk1 wakeups]# cat 2340.2001.call |
| 04:39.39 | dmccollum | # |
| 04:39.39 | dmccollum | Channel: Local/2001@inside |
| 04:39.39 | dmccollum | MaxRetries: 3 |
| 04:39.39 | dmccollum | RetryTime: 60 |
| 04:39.39 | CoaxD | AgiNamu: First kid? |
| 04:39.40 | dmccollum | WaitTime: 30 |
| 04:39.42 | dmccollum | Application: AGI |
| 04:39.42 | AgiNamu | yep |
| 04:39.44 | dmccollum | Data: snooze.agi |
| 04:39.44 | tessier | AgiNamu: Well, that'll learn ya. Enjoy the next 18 years! ;) |
| 04:39.45 | AgiNamu | 2 more months |
| 04:39.45 | dmccollum | Callerid: Wakeup Call Service <293> |
| 04:39.48 | AgiNamu | lol |
| 04:39.48 | CoaxD | AgiNamu: Boy, you're in for it now |
| 04:39.49 | *** kick/#asterisk [dmccollum!~brian@bkw.developer.and.friend.of.asterisk] by bkw_ (bkw_) |
| 04:39.54 | poli | bkw_, hehe The guy who develops it (actually a friend of name) is calles Blanes. :) |
| 04:39.57 | CoaxD | AgiNamu: No worries, in a year or two you'll be filing for divorce. so its all good |
| 04:39.57 | AgiNamu | I've been wanting to have kids for years. |
| 04:39.59 | bkw_ | ah |
| 04:40.00 | *** join/#asterisk dmccollum (~dmccollum@69-164-245-72.atlaga.adelphia.net) |
| 04:40.02 | shepherd | dmcollum: PASTEBIN |
| 04:40.04 | *** join/#asterisk Rick_Hunter (~rhunter@02-132.008.popsite.net) |
| 04:40.05 | bkw_ | LOOOK here don't do that again |
| 04:40.07 | AgiNamu | but she kept on complaining about needing to get marired, and have a job and so on :) |
| 04:40.10 | CoaxD | AgiNamu: (Its the american dream.) |
| 04:40.14 | bkw_ | you motherfuckers need pastebin |
| 04:40.15 | bkw_ | haha |
| 04:40.44 | CoaxD | AgiNamu: I lived the dream. And now, I have a 4 year old son to show for it. *lol* |
| 04:40.45 | dmccollum | sorry, didn't think it was a big deal with just 4 lines. |
| 04:40.45 | AgiNamu | anyways.... out to .... teach her to drive... wish me luck |
| 04:40.51 | MrBelvedr | PM me if you want to offer asterisk/linux hosting |
| 04:40.52 | CoaxD | AgiNamu: LUCK!###$ |
| 04:40.53 | AgiNamu | I want o have 8 kids |
| 04:40.58 | bkw_ | 9 lines dear |
| 04:40.59 | CoaxD | aginamu: Have fun with that, sir! |
| 04:40.59 | AgiNamu | I wanna reverse this fucking IQ trend here |
| 04:41.00 | bkw_ | 9 |
| 04:41.05 | AgiNamu | where smart people have few kids, and idiots have tons. |
| 04:41.09 | AgiNamu | cya |
| 04:41.27 | CoaxD | ...he's got a point |
| 04:41.32 | bkw_ | AgiNamu, so geniuses have ZERO |
| 04:42.04 | CoaxD | idiot women just cant figure out why they keep getting pregnant.. 'okay, but just this once.' |
| 04:42.10 | dmccollum | http://pastebin.ca/8509 |
| 04:42.28 | bkw_ | or "Can I just stick the head in...??!?!?" |
| 04:42.31 | tessier | bkw_: hehehe... |
| 04:42.34 | CoaxD | bkw: HAHAHAHAHAHA |
| 04:42.40 | CoaxD | bkw: *rotfl* |
| 04:42.55 | CoaxD | bkw: Sure. and then due to female instincts, you have about a 90% chance of the woman grabbing you and saying "FUCK ME GOD DAMNIT"! |
| 04:42.56 | shmaltz | anybody here using a polycom phone? |
| 04:42.59 | tessier | Makes me want to run out and get a vasectomy right now. |
| 04:43.36 | bkw_ | CoaxD, NEVER |
| 04:43.42 | bkw_ | hard to do that when ya can't get it up |
| 04:43.43 | CoaxD | bkw: haha |
| 04:43.50 | CoaxD | bkw: FUCK ME, WOMAN! |
| 04:43.55 | MrBelvedr | never trust a chick who swears she is on the pill so you can have unprotected sex |
| 04:43.58 | bkw_ | it woudl be about like beating two water melons together |
| 04:44.02 | MrBelvedr | that is the oldest fucking trick in the book |
| 04:44.03 | bkw_ | it would be messy |
| 04:44.06 | MrBelvedr | $ |
| 04:44.18 | CoaxD | mrbelvedr: Good man. |
| 04:44.59 | CoaxD | went to eat with my mom tonight. Realized that we were at the same restaraunt i'd eaten lunch in several hours earlier. I discovered I need a life. |
| 04:45.23 | MrBelvedr | heh |
| 04:45.45 | bkw_ | haha |
| 04:45.54 | bkw_ | CoaxD, I read that as "went to et my mom tonight" |
| 04:46.00 | CoaxD | GAH! |
| 04:46.12 | MrBelvedr | is it possible to run "virtual instances" of * on a server. That way like 10 people can get together to save money on a shared * server |
| 04:46.14 | zhier | how to configure the * as a SIP Server??? |
| 04:46.28 | bkw_ | MrBelvedr, No |
| 04:46.36 | dmccollum | sure if you're running linux jails or solaris zones in solaris 10. |
| 04:48.02 | bkw_ | or you build each to look in diffrent places for asterisk.ctl |
| 04:48.16 | MrBelvedr | yeah, cause i am cheap |
| 04:48.32 | MrBelvedr | so if somebody wants to do that i would be interested |
| 04:48.51 | MrBelvedr | i will pay $10 a month or something to get a good connection |
| 04:49.40 | MrBelvedr | can anybody reccommend a cheap linux host that costs about $10 a month that offers root? |
| 04:49.52 | CoaxD | um |
| 04:50.03 | CoaxD | you're asking for a colo (however virtual it may be) for $10/mo? |
| 04:50.15 | MrBelvedr | yeah |
| 04:50.15 | dmccollum | Anyone able to help me see why this file doesn't run in the /var/spool/asterisk/outgoing? http://pastebin.ca/8509 |
| 04:50.36 | CoaxD | MrBelvedr: Do you understand how much work has to go into your $10/mo? |
| 04:50.39 | MrBelvedr | not colo, but i have seen some hosts that offer "root" but it is not really root |
| 04:50.51 | CoaxD | MrBelvedr: Correct. Its root in a jail |
| 04:50.57 | znoG | MrBelvedr: you can get a VPS for $20/mth.. you're pushing it for $10 |
| 04:50.59 | CoaxD | MrBelvedr: There is a way to do it with usermode linux |
| 04:51.05 | CoaxD | MrBelvedr: but um, slow? |
| 04:51.17 | znoG | good luck getting any decent performance out of it for asterisk |
| 04:51.21 | MrBelvedr | is there any cheap places you know of? |
| 04:51.37 | dmccollum | why would you want to do that when you can find an old PC for next to nothing? |
| 04:51.48 | MrBelvedr | because i have cable modem |
| 04:51.50 | MrBelvedr | they block it |
| 04:51.57 | MrBelvedr | they block port 80 |
| 04:52.09 | CoaxD | haha thats pretty funny |
| 04:52.16 | MrBelvedr | and it is not reliable, i need something reliable for my app. I am not using it for a pbx |
| 04:52.24 | MrBelvedr | just for my app to place outgoing calls |
| 04:52.28 | dmccollum | oh, thought you were talking about for * |
| 04:52.36 | MrBelvedr | i am talking about * |
| 04:52.39 | znoG | you're saying they block outgoing IAX/SIP? |
| 04:52.44 | MrBelvedr | no |
| 04:52.51 | CoaxD | MrBelvedr just wants something for nothing |
| 04:53.09 | MrBelvedr | i went off on a tangent. my main problem is that my home isp is unreliable, and i need something reliable |
| 04:53.27 | dmccollum | you could get a dynamic DNS name that forwards the traffic to a port other than 80. |
| 04:53.38 | znoG | MrBelvedr: you need to make outgoing calls, right? |
| 04:53.42 | MrBelvedr | yes |
| 04:53.59 | MrBelvedr | just outgoing, but it is not as a pbx. it is a web application. |
| 04:54.25 | MrBelvedr | so i need a reliable host that offers a jail plan or whatever they are called |
| 04:55.10 | znoG | so just put together a home linux box, stick asterisk on it and away you go |
| 04:55.21 | znoG | you want a reliable host that you can run asterisk on for $10/mth? |
| 04:55.25 | MrBelvedr | yeah znog i already did that, it works great |
| 04:55.48 | MrBelvedr | yeah just a simple shared server |
| 04:55.52 | MrBelvedr | it is not that complicated |
| 04:56.00 | MrBelvedr | with jails like they said |
| 04:56.04 | znoG | i also want a 5 foot blonde, with large breasts, who loves to cook, clean, wash and have sex at least 3 times an hour... but i'm not going to get the hair colour i want am i! |
| 04:56.22 | Qwell | Why 5'? |
| 04:56.35 | znoG | 5 foot.. okay i meant 5'5 at least |
| 04:56.40 | topping | znoG: insist on dentures too |
| 04:57.42 | shepherd | m or f? |
| 04:57.44 | shepherd | heh |
| 04:58.12 | znoG | i guess i should have specified age.. :) |
| 04:58.14 | Qwell | shepherd: With all that, would it really matter? |
| 04:58.32 | MrBelvedr | lol |
| 04:58.36 | dmccollum | forget the cooking and cleaning as long as she's willing to have sex 3 times a day. |
| 04:58.45 | topping | realdoll.com |
| 04:59.12 | dmccollum | sounds like you have some experience Topping |
| 04:59.25 | topping | no, just saw it on boingboing |
| 04:59.33 | topping | thought you might be interested ;-) |
| 04:59.39 | znoG | hey if she is not gonna cook and clean, its gotta be 3 times an hour |
| 05:01.36 | dmccollum | You can pay a woman to either clean or for sex, you've hit the jackpot if you can find one that's willing to do both. |
| 05:02.36 | topping | sounds like you have some experience dmccollum |
| 05:02.57 | topping | :-P |
| 05:03.23 | dmccollum | yea. marriage. I'm paying for no sex. |
| 05:03.33 | topping | haha yah, i remember that |
| 05:05.24 | *** join/#asterisk seawolf_ (~seawolf_@sea.slackwolf.com) |
| 05:05.49 | *** join/#asterisk abatista (~Ariel@dsl-20-177.cofs.net) |
| 05:06.49 | *** join/#asterisk implicit (~implicit@ip68-7-149-247.sd.sd.cox.net) |
| 05:07.45 | MrBelvedr | what is the correct terminology for a server that offers jail root |
| 05:07.55 | MrBelvedr | "jailed root" |
| 05:08.06 | MrBelvedr | i am posting to some mailing lists |
| 05:08.34 | znoG | for $10/mth powerful enough to run asterisk? the correct terminology is "dream on" |
| 05:08.56 | *** join/#asterisk jsmith (~jsmith@smithfam.dsl.xmission.com) |
| 05:09.16 | jsmith | Anybody doing AGI programming in Python? If so, I have a quick and stupid question. |
| 05:09.44 | MrBelvedr | how about $50 a month |
| 05:09.52 | dmccollum | chroot jail |
| 05:09.59 | MrBelvedr | i am just looking for something cheaper then a colo or ded box |
| 05:10.12 | MrBelvedr | asterisk doesn't take that many resources |
| 05:10.26 | DEEZED | www.servermatrix.com |
| 05:10.52 | dmccollum | anyone here that can help me with a .call file? |
| 05:11.21 | MrBelvedr | thans deez |
| 05:12.10 | *** join/#asterisk AsteriskNoob (~asdf@24-117-23-127.cpe.cableone.net) |
| 05:12.32 | AsteriskNoob | anybody hangin around this late? |
| 05:12.37 | implicit | yep |
| 05:12.47 | *** part/#asterisk Half_Dome (~jelway@mail.westmarkinc.com) |
| 05:12.53 | AsteriskNoob | cool, hey you wouldnt happen to know how to identify which channels are what would you? |
| 05:13.06 | AsteriskNoob | or is that configured in the zaptel.conf? |
| 05:13.07 | *** part/#asterisk seawolf_ (~seawolf_@sea.slackwolf.com) |
| 05:13.19 | MrBelvedr | valueweb has used ded servers for 59 a month |
| 05:13.37 | MrBelvedr | http://www.valueweb.com/promos/dedicatedserverhosting2.htm |
| 05:13.41 | jsmith | AsteriskNoob: Zap channels are defined in zaptel.conf and zapata.conf |
| 05:13.42 | MrBelvedr | "pre-owned" |
| 05:13.56 | jsmith | AsteriskNoob: You can always type "zap show channels" at the Asterisk CLI |
| 05:14.15 | AsteriskNoob | jsmith: I've been using 2 x100p's those were easy but i just upgraded to a TE110P and a TDM400P and they arent configured yet |
| 05:14.32 | AsteriskNoob | so the only thing that shows up with "zap show channels" is the pseudo interface |
| 05:14.51 | jsmith | AsteriskNoob: Configure them in zaptel.conf and zapata.conf |
| 05:15.08 | jsmith | AsteriskNoob: Use zttool to make sure the system is seeing them. If not, load the correct kernel modules |
| 05:15.20 | AsteriskNoob | thats what i thought but maybe i've just been staring at it too long... |
| 05:17.22 | AgiNamu | WOW |
| 05:17.26 | AgiNamu | I think I have medical proof |
| 05:17.31 | AgiNamu | that women are inferior drivers. |
| 05:17.44 | AsteriskNoob | why's that ? lol |
| 05:17.47 | AgiNamu | Q: How to waste a xanax? A: Teach your wife to drive. |
| 05:18.08 | AgiNamu | OK, I'm buying a fucking driving wheel and pedals |
| 05:18.09 | implicit | yeah, that's on the info sheet for xanax |
| 05:18.12 | AgiNamu | and she can learn with GTA |
| 05:18.15 | AgiNamu | LOL |
| 05:18.22 | topping | you should have waited until 2230 |
| 05:18.29 | AgiNamu | *5* minutes to get out of the garage. |
| 05:18.42 | AgiNamu | *2* minutes to get it going down the road. |
| 05:18.48 | AgiNamu | antoher 2 to turn it around. |
| 05:18.55 | implicit | AgiNamu, is it hard for her to reach the clutch? |
| 05:19.01 | AgiNamu | nope. |
| 05:19.08 | AgiNamu | in all, I think we averaged 0.001km/hr. |
| 05:19.17 | implicit | did she stall it at all? |
| 05:19.26 | topping | well, she'll know that walking is faster from here on out |
| 05:19.29 | topping | it's a good thing |
| 05:19.48 | AgiNamu | lol |
| 05:19.52 | implicit | tell her she should just quit and get her an automatic man |
| 05:19.55 | AgiNamu | stall it at all? |
| 05:20.02 | AgiNamu | the thing almost stalls itself. |
| 05:20.06 | AgiNamu | It's a 93 Trooper. |
| 05:20.11 | implicit | automatic? |
| 05:20.13 | AgiNamu | Before had an 89 jeep korando |
| 05:20.16 | AgiNamu | hell no. manual. |
| 05:20.19 | AsteriskNoob | ok, who knows what this means : ZT_SPANCONFIG failed on span 1: Invalid argument (22) |
| 05:20.19 | implicit | good |
| 05:20.23 | AgiNamu | Im not buying a deathtrap like an automatic. |
| 05:20.38 | implicit | AgiNamu, i want to buy a new flywheel for mine |
| 05:20.48 | AgiNamu | asteriskn00b, no clue. but maybe it means you configured FXO instead of FXS or simialar? |
| 05:21.00 | implicit | get a lighter one so i can push it up a bit to 350hp |
| 05:21.01 | AsteriskNoob | trying to do the T1 span |
| 05:21.08 | AgiNamu | lol |
| 05:21.16 | AgiNamu | then i definately have no clue. |
| 05:21.23 | AgiNamu | i only have DS3s. |
| 05:21.39 | implicit | then i'll be getting it supercharged |
| 05:22.19 | AgiNamu | I just want something that's stable, and can drive over medians, small cars and old women. |
| 05:22.26 | implicit | heh |
| 05:22.26 | AgiNamu | the jeep korando fits that. |
| 05:22.34 | implicit | mine is small |
| 05:22.37 | AgiNamu | I hit an idiot who decided to cross a highway and stop in the middle of it |
| 05:22.46 | topping | can i get asterisk on a DS3 for $10 per month? anyone? |
| 05:22.49 | AgiNamu | broke his freaking tire, door, everything. |
| 05:22.54 | implicit | topping, lol |
| 05:22.57 | AgiNamu | topping $200 a month |
| 05:23.00 | topping | hehe |
| 05:23.02 | AgiNamu | with $15K down |
| 05:23.05 | AgiNamu | 1U |
| 05:23.29 | AgiNamu | actually, i could probably do it for $5K down |
| 05:23.42 | implicit | AgiNamu, man that was already a good deal |
| 05:23.44 | AgiNamu | you wouldn't be exclusive though |
| 05:23.45 | implicit | you are making yourself lose money |
| 05:23.48 | AgiNamu | hehe |
| 05:23.54 | AgiNamu | only if he pays |
| 05:23.55 | implicit | 10k down the hole |
| 05:26.32 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@l01m-20-133.d4.club-internet.fr) |
| 05:31.04 | DEEZED | my isp was nice enough to upgrade my down speed.. but upload is the same |
| 05:31.07 | DEEZED | heh |
| 05:31.50 | MrBelvedr | Spry VPS servers with full root access starting at $14.95 |
| 05:32.00 | DEEZED | are they any good? |
| 05:32.23 | topping | with only a five year contract! |
| 05:33.12 | MrBelvedr | i have no idea |
| 05:33.34 | MrBelvedr | ima email them and see what they say about putting * on there |
| 05:45.00 | Wazb | is it possible to register H323 phone with Asterisk? |
| 05:45.25 | *** join/#asterisk kpfleming (~kpfleming@207.111.174.1) |
| 05:46.36 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
| 05:51.24 | *** join/#asterisk iceyp (~icepick@202.150.105.150) |
| 05:51.24 | *** join/#asterisk ManxPower (~eric@dsl-209-16-67-160.datasync.com) |
| 05:51.29 | ManxPower | ~docs |
| 05:51.30 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 05:51.31 | ManxPower | ~mailinglist |
| 05:51.32 | jbot | from memory, mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ |
| 05:52.10 | iceyp | Shido6 any idea why i get this when trying to place a call with my 7940 ... : |
| 05:52.11 | iceyp | Mar 31 17:51:57 NOTICE[50495]: chan_sip.c:7313 handle_request: Unable to create/find channel |
| 05:52.11 | iceyp | Mar 31 17:51:59 WARNING[50495]: chan_sip.c:686 retrans_pkt: Maximum retries exceeded on call 0011219c-48830008-75d4146e-4969f82a@10.23.19.170 for seqno 101 (Non-critical Response) |
| 05:52.52 | *** join/#asterisk stdio (~stdio@pcp09745793pcs.lncstr01.pa.comcast.net) |
| 05:52.59 | iceyp | i'm using: |
| 05:52.59 | iceyp | preferred_codec: g711ulaw |
| 05:54.06 | stdio | greets all. |
| 05:54.23 | stdio | Anyone have any inkling of an idea, when 1.1 is coming out of cvs? |
| 05:56.13 | kpfleming | no, there is no specific plan at this time |
| 05:57.35 | *** join/#asterisk r0d3nt|m (nobody@wsip-24-234-241-84.lv.lv.cox.net) |
| 05:57.52 | Wazb | i am gone thorough those documentation but they are not ver helpful |
| 05:59.37 | Shido6 | whats up? |
| 05:59.38 | Shido6 | dood |
| 05:59.56 | Shido6 | shoot me 85 bucks to billing@nufone.net and schedule a session or $20 and your ssh login info :) |
| 06:00.07 | *** join/#asterisk lately (~dougb@chi.econ.usyd.edu.au) |
| 06:01.51 | *** join/#asterisk soundguy (~soundguy@zeus.blendtek.com.au) |
| 06:02.27 | *** join/#asterisk Qorky (~goaway@dsl-202-72-146-104.wa.westnet.com.au) |
| 06:03.13 | *** join/#asterisk KristinH (~KristinH@muppet.geekgirls.us) |
| 06:03.19 | KristinH | hello |
| 06:03.23 | iceyp | meh |
| 06:04.48 | KristinH | is anyone using asterisk with freeworld dialup? |
| 06:04.49 | *** join/#asterisk fedor (~fbond@office.tura.ru) |
| 06:05.19 | AsteriskNoob | hey yo... anybody seen this before? ZT_SPANCONFIG failed on span 1: Invalid argument (22) |
| 06:06.09 | Qwell | KristinH: yeah, works great |
| 06:07.37 | KristinH | i'm just now trying to set it up |
| 06:10.10 | KristinH | and I am having a problem with reaching any of the service numers |
| 06:10.14 | iceyp | 10.23.19.170 (None) 0011219c-48 00101/00104 unknow |
| 06:10.35 | *** join/#asterisk DeadDawg (~DeadDawg@CPE000d5659d23f-CM000a739a1839.cpe.net.cable.rogers.com) |
| 06:10.43 | Shido6 | AsteriskNoob |
| 06:10.48 | DeadDawg | Hey. |
| 06:10.55 | Shido6 | you have an idea of what you want to do with your gear? |
| 06:11.05 | Shido6 | or need a training session? |
| 06:11.47 | DeadDawg | Anybody familiar with MGCP? |
| 06:11.51 | Shido6 | yes |
| 06:12.08 | DeadDawg | Any word on MGCP client for asterisk? |
| 06:12.30 | Shido6 | ? |
| 06:12.35 | Shido6 | softclient? |
| 06:12.46 | Shido6 | mgcp works fine with mgcp phones |
| 06:12.49 | DeadDawg | My main local VoIP providor uses MGCP and going from there router to FXO is annoying. |
| 06:12.51 | KristinH | i keep getting "i am sorry that is not a valid extension" message on 411, 612, 55555, etc |
| 06:13.12 | Shido6 | KristinH, what are you looking to do with Asterisk? |
| 06:13.26 | KristinH | connect to FWD |
| 06:13.37 | iceyp | why doesnt asterisk and my cisco 7940 codecs work? |
| 06:13.38 | KristinH | via IAX2 |
| 06:13.45 | KristinH | which I can |
| 06:13.49 | Shido6 | you have something misconfigured, iceyp |
| 06:14.14 | DeadDawg | KristinH: all FWD numbers work expect those? |
| 06:14.20 | DeadDawg | except* |
| 06:14.24 | KristinH | none work |
| 06:14.36 | KristinH | bye the call me tool works |
| 06:14.39 | KristinH | but |
| 06:14.40 | DeadDawg | are you dialing the proper digits to specify FWD |
| 06:14.48 | DeadDawg | EG i dial 7 first |
| 06:14.58 | KristinH | i followed the install exactly |
| 06:15.07 | KristinH | 393 is the prefix |
| 06:15.43 | iceyp | preferred_codec: g711ulaw in my SIPDefault.cnf & in sip.conf ... |
| 06:15.44 | iceyp | disallow=all ; Disallow all codecs |
| 06:15.45 | iceyp | allow=gsm |
| 06:15.45 | iceyp | allow=ilbc |
| 06:15.45 | iceyp | allow=ulaw |
| 06:15.45 | iceyp | allow=alaw |
| 06:17.35 | *** join/#asterisk florz (nobody@2001:1a50:503c:0:0:0:0:1) |
| 06:17.44 | DeadDawg | KristinH: I am very sorry I have been busy in University lately and haven't been playing with Asterisk as much as I used to so I do not think I will be able to help you right now. |
| 06:17.46 | Shido6 | "/usr/src/asterisk/channels/h323/README" |
| 06:17.54 | Shido6 | Wazb |
| 06:18.02 | DeadDawg | I plan on getting back into Asterisk soon ;-) |
| 06:18.16 | iceyp | Shido6 does cisco use h323 not sip? |
| 06:18.22 | Shido6 | iceyp, did you send anything to paypal? |
| 06:18.30 | Shido6 | Wazb asked me about h323 |
| 06:18.43 | iceyp | Shido6 no |
| 06:18.56 | iceyp | i'm asking for a little help not a full pabx |
| 06:18.56 | Shido6 | [01:17] <Wazb> meanwhile can u guide me where i can find good documentation regarding Asterisk with H323 |
| 06:21.13 | DeadDawg | I am curious at what the FWD numbers start at, I am trying to see who has the lowest FWD. :-) |
| 06:21.18 | *** join/#asterisk tessier (~treed@222.253.72.192) |
| 06:22.42 | mishehu | I'm trying to build asterisk 1.0.7 addons, and I get this error for app_addon_sql_mysql.c:164:64: macro "AST_LIST_REMOVE" requires 4 arguments, but |
| 06:22.45 | mishehu | only 3 given |
| 06:23.04 | mishehu | anybody know off the top of their head what the 4 arguments should be? |
| 06:24.38 | drumkilla | mishehu: when did you download it? |
| 06:25.02 | drumkilla | you might have accidently got the tarball that was a snapshot of CVS head |
| 06:25.12 | mishehu | drumkilla: ah, was about a week ago or so I think |
| 06:25.17 | mishehu | I'll redownload it |
| 06:25.25 | drumkilla | ok, should be fixed, though |
| 06:25.32 | iceyp | anyone here got cisco 7940 working with asterisk? |
| 06:25.58 | mishehu | iceyp: do you have a specific question? |
| 06:26.10 | mishehu | I'm sure there people are who have gotten that to work. I have a 7960. |
| 06:26.16 | iceyp | yes , when making a call i get this: : |
| 06:26.16 | iceyp | Peer User/ANR Call ID Seq (Tx/Rx) Format |
| 06:26.17 | iceyp | 10.23.19.170 (None) 0011219c-48 00101/00109 unknow |
| 06:26.21 | Mother_ | yes |
| 06:26.27 | iceyp | unknown |
| 06:26.33 | iceyp | whereas im using ulaw |
| 06:26.42 | iceyp | 202.150.105.150 "Barry Mur 0011219c-48 00101/00101 ulaw |
| 06:27.28 | stdio | fyi - AST_LIST_REMOVE removes an arbitrary element from the head: |
| 06:27.28 | stdio | AST_LIST_REMOVE(headp,node1,ast_var_t,listpointers); |
| 06:28.53 | *** join/#asterisk TheEmperor (~mattn@203.121.47.165) |
| 06:29.31 | iceyp | ok the problem is because of nat |
| 06:30.22 | stdio | mishehu: more info : |
| 06:30.28 | stdio | mishehu: http://www.mail-archive.com/asterisk-users@lists.digium.com/msg82739.html |
| 06:30.36 | Shido6 | iceyp |
| 06:30.37 | Shido6 | dood |
| 06:30.51 | Shido6 | what codec do you have set in sip<macaddress>.cnf |
| 06:30.56 | Shido6 | and sipdefault.cnf |
| 06:30.57 | iceyp | set a static IP on the phone |
| 06:30.58 | *** join/#asterisk lately (~dougb@chi.econ.usyd.edu.au) |
| 06:31.00 | iceyp | and now it works |
| 06:31.12 | iceyp | i use a live ip |
| 06:31.13 | Shido6 | kewl |
| 06:31.19 | Shido6 | was that hard? :) |
| 06:31.24 | Shido6 | everything workin now? |
| 06:31.29 | iceyp | yeah |
| 06:31.33 | *** join/#asterisk subtract (~subtract@ottawa-hs-64-26-147-151.d-ip.magma.ca) |
| 06:31.34 | iceyp | now to setup nat on the phone |
| 06:31.35 | Shido6 | learn sumfin new? :) |
| 06:31.42 | Shido6 | turn on nat processing in the .cnf |
| 06:31.49 | Shido6 | and nat=yes in the peer and user |
| 06:31.53 | Shido6 | and reload chan_sip.so |
| 06:32.03 | stdio | mishehu: note, that was only just recently posted Mar 19 |
| 06:32.28 | lately | I'm trying to get my phone to log in to *. In agents.conf I have "agent => 1001,1234,Ben Dover" and my phone tries to connect but I get registration failed etc from 1001@blah. |
| 06:32.34 | iceyp | nat still doesnt work |
| 06:32.40 | lately | What have I done wrong? |
| 06:32.58 | iceyp | maybe i need to forward port 5060 from my router to the phone?? |
| 06:33.08 | subtract | I'm looking for a really basic softphone that compiles easily on OpenBSD if been trying to build linphone without success? |
| 06:33.39 | Qwell | subtract: tried iaxcomm? |
| 06:33.50 | subtract | no is it good |
| 06:33.56 | Qwell | I like it |
| 06:34.15 | subtract | have you run it on OBSD before? |
| 06:34.22 | poli | Qwell, are you running in Linux? |
| 06:34.30 | Qwell | no, but its crossplatform it claims |
| 06:34.39 | Qwell | poli: yeah, and I'm quite happy with iaxcomm...no thanks |
| 06:34.48 | drumkilla | mishehu: actually, the problem is that you didn't make install asterisk before you tried to compile addons |
| 06:35.08 | drumkilla | your version of linkedlists.h in /usr/include/asterisk/ was that from cvs head |
| 06:35.45 | subtract | Qwell: is it very gui, you know what i mean |
| 06:36.00 | *** join/#asterisk implicit (~implicit@ip68-5-148-1.oc.oc.cox.net) |
| 06:36.21 | Qwell | very gui? |
| 06:36.23 | Qwell | like, complex? |
| 06:36.29 | subtract | graphical |
| 06:36.30 | Qwell | no, its very simple... |
| 06:36.43 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
| 06:36.48 | subtract | command line? |
| 06:36.52 | lately | On my phone do I put my userid as "1001" if my agents.conf has "1001,1234,Name" ? |
| 06:36.53 | Qwell | no |
| 06:37.14 | subtract | thanks Qwell |
| 06:37.46 | subtract | i'll give it a go |
| 06:42.45 | *** part/#asterisk bl1tz (~brian@adsl-67-114-145-205.dsl.sntc01.pacbell.net) |
| 06:44.04 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) |
| 06:44.59 | *** join/#asterisk Alexis (~alexis@www.trim.it) |
| 06:45.04 | Alexis | Bonjour |
| 06:54.44 | *** join/#asterisk odie_flocon (~chatzilla@S01060011953994ee.cg.shawcable.net) |
| 06:56.15 | *** join/#asterisk ikey (ikey@220.226.41.113) |
| 06:57.13 | ikey | any one has experiance on r2mfc on asterisk ? |
| 06:57.55 | *** join/#asterisk Supaplex (supaplex@205.208.245.134) |
| 06:58.40 | *** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
| 06:58.43 | ikey | any one has experiance on r2mfc on asterisk ? |
| 06:58.53 | *** join/#asterisk florz (nobody@2001:1a50:503c:0:0:0:0:1) |
| 06:59.04 | Supaplex | is it unusual to have a channel bank w/no expansion cards, supporting just the voice lines? eg, either that 50pin centronic connector, or 24 punchdown lines, or even 24 rj11 jacks? |
| 07:05.03 | *** join/#asterisk firestrm (firestrm@S010600047577bccd.gv.shawcable.net) |
| 07:05.53 | firestrm | hey, can i set gains on a per channel basis, ie with a tdm400, set gain different on 1,2,3,4 ? |
| 07:06.05 | *** join/#asterisk pif (ldm@zenon.apartia.fr) |
| 07:06.22 | Shido6 | yes |
| 07:06.26 | Shido6 | zapata.conf |
| 07:06.49 | firestrm | Shido6, so insted of at the beginning, i would move gains to each channel def? |
| 07:07.02 | Shido6 | see how u set the context above the channel? |
| 07:07.06 | Shido6 | u can do the same for the gains |
| 07:07.20 | firestrm | ok, that makes sense.. thanks |
| 07:07.32 | Shido6 | changes made in zapata.conf |
| 07:07.35 | Shido6 | need a restart |
| 07:08.01 | firestrm | ya, i know.. thats one of the things i wish they would fix.. reload should reload everything.. |
| 07:08.36 | firestrm | or even better, have a way of tweaking gains on the fly.. would make tuning much easier |
| 07:08.48 | ManxPower | A reload is not supposed to terminate calls. chan_zap would have to terminate calls in order to reload. |
| 07:09.31 | firestrm | ManxPower, i see your point, but there should be a way of restarting without killing asterisk.. its a real P.I.T.A. |
| 07:09.47 | ManxPower | firestrm: unload chan_zap.so and load chan_zap.so |
| 07:10.00 | ManxPower | only works if there are no active Zap calls |
| 07:10.01 | firestrm | hmm.. thanks..i'll try that. |
| 07:10.23 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@host.161.115.68.195.rev.coltfrance.com) |
| 07:10.45 | firestrm | once i have my 0db reference built, i will be able to tweak with a known quantity.. |
| 07:11.08 | *** part/#asterisk stdio (~stdio@pcp09745793pcs.lncstr01.pa.comcast.net) |
| 07:11.10 | firestrm | i allmost have it working.. |
| 07:11.22 | Zeeek | morning |
| 07:11.47 | firestrm | just got to get the db monitor working, tone reference was fairly easy.. |
| 07:14.50 | firestrm | ManxPower, you were right by the way about echo problems, once you get the gains close, most of the problems go away magicly.. |
| 07:16.25 | PTG1234 | anyone alive? |
| 07:16.30 | firestrm | i am.. |
| 07:16.36 | djin | still breathing here |
| 07:16.37 | PTG1234 | k wanted to make sure irc was still working |
| 07:16.38 | PTG1234 | :) |
| 07:16.47 | firestrm | lol |
| 07:16.54 | djin | IRC is. |
| 07:16.56 | PTG1234 | any php experts alive? :) |
| 07:17.01 | firestrm | i sometimes get that feeling.. |
| 07:17.22 | djin | define expert |
| 07:17.31 | firestrm | PTG1234, ive played with it, i not exactly and expert. but my wife is.. |
| 07:17.35 | dwmw2 | define alive |
| 07:17.42 | djin | or just ask and we'll see if there's help |
| 07:17.52 | firestrm | at leas IBM thinks she is anyways.. |
| 07:18.06 | djin | define IBM thinking |
| 07:18.10 | firestrm | lol |
| 07:18.20 | firestrm | all your thoughts are belong to us |
| 07:18.26 | djin | :) |
| 07:18.44 | PTG1234 | well i need someone to spear head an admin for my new asterisk modules :) |
| 07:18.47 | firestrm | we own the rights to your dna and all deritive works.. |
| 07:18.50 | PTG1234 | i want to make them easy to config |
| 07:19.00 | PTG1234 | my replacement chan_sip, and etc |
| 07:19.10 | firestrm | PTG1234, thats a very worthy project.. |
| 07:19.21 | dwmw2 | PTG1234, you have a replacement chan_sip? |
| 07:19.42 | firestrm | brb afk |
| 07:19.53 | *** join/#asterisk ckruetze (~nospam@i3ED65A4E.versanet.de) |
| 07:20.07 | dwmw2 | does it support IPv6? |
| 07:20.12 | PTG1234 | yah i do |
| 07:20.26 | PTG1234 | um no IPv6 i don't see the point really |
| 07:20.36 | PTG1234 | and yah it is a worth project :) |
| 07:20.36 | Qwell | would help with NAT |
| 07:20.50 | PTG1234 | well IPv6 proxies are basically just NAT proxies |
| 07:20.53 | Qwell | any production * deployment shouldn't be on a NAT though, heh |
| 07:20.58 | PTG1234 | to transform the addresses to IPv6 addresses |
| 07:21.07 | PTG1234 | oh i think i got nat problems all fixed in my new chan_sip |
| 07:21.12 | Qwell | ahh |
| 07:21.22 | PTG1234 | just need some php people :) |
| 07:21.29 | dwmw2 | PTG1234, nobody really bothers with NAT-PT between IPv6 and the Legacy Internet. People just route IPv6 |
| 07:21.29 | PTG1234 | it has a full interface library i wrote |
| 07:21.31 | PTG1234 | to manage users |
| 07:21.35 | PTG1234 | just need the web part |
| 07:21.48 | dwmw2 | so you don't end up with 'IPv6 proxies', surely? |
| 07:21.53 | PTG1234 | dwmw2: yah but if the other side has no IPv6 address, then its useless |
| 07:22.08 | Qwell | There is that too, yeah |
| 07:22.12 | PTG1234 | somewhere it has to translate the IPv6 to IPv4 for use with the rest of us guys |
| 07:22.27 | dwmw2 | you listen on '::' and it accepts both IPv4 and IPv6 connections. |
| 07:22.35 | dwmw2 | most IPv6 boxes _also_ have an IPv4 address |
| 07:22.42 | Qwell | but then you're back to NAT |
| 07:22.46 | PTG1234 | well the best way is to use a proxy |
| 07:22.50 | PTG1234 | so it works for both |
| 07:22.53 | PTG1234 | if you also have an IPv4 |
| 07:22.57 | PTG1234 | then why would you use an IPv6 |
| 07:22.58 | PTG1234 | :) |
| 07:23.02 | PTG1234 | just to feel special? |
| 07:23.08 | Qwell | yes :p |
| 07:23.09 | dwmw2 | the best way is to have the server listen on both IPv4 and IPv6 addresses. Then it works for both |
| 07:23.10 | PTG1234 | No php guys want to take on a project? |
| 07:23.10 | Makenshi | i want ipv6 support |
| 07:24.17 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
| 07:24.25 | PTG1234 | well write it in :) |
| 07:24.27 | PTG1234 | how hard would it be |
| 07:24.29 | dwmw2 | Because I only have about 13 legacy IP addresses at home. |
| 07:24.38 | dwmw2 | and I have something like 2^97 IPv6 addresses. |
| 07:24.56 | dwmw2 | PTG1234, not very. I was just asking if you already did it. |
| 07:25.05 | dwmw2 | I'm looking at the existing chan_sip now |
| 07:25.17 | PTG1234 | nah like i said, i don't see the point myself.. but i welcome whoever to do it :) |
| 07:25.22 | PTG1234 | as long as they don['t put bugs into my code |
| 07:25.22 | PTG1234 | heh |
| 07:26.24 | dwmw2 | my cvs tree is a bit cluttered at the moment -- I'll probably wait till the patches in #3865 and #3889 are committed. And I have real work to do. |
| 07:26.40 | dwmw2 | should I be working against your code -- is that going to replace what's in CVS any time soon? |
| 07:27.59 | PTG1234 | i will probably release it as chan_sip2.c |
| 07:28.08 | *** join/#asterisk tafazzi (~Dario@eporroj0-1.customer.nettuno.it) |
| 07:28.18 | PTG1234 | everyone and their mother is trying to rewrite chan_sip |
| 07:30.10 | *** join/#asterisk Verliba (~Miranda@250.1-14-84.ripe.coltfrance.com) |
| 07:30.11 | Qwell | brb |
| 07:30.19 | Qwell | PTG1234: check your aim when you get a chance |
| 07:30.26 | PTG1234 | oh |
| 07:30.27 | PTG1234 | ok |
| 07:30.49 | *** join/#asterisk dopey (matth@uruk-hai.bioinformatics.unsw.EDU.AU) |
| 07:38.00 | libpcp | anyone has an experienced configuring SMS? i would like to know if this setup will work. GSM1-Asterisk--internet--Asterisk-GSM2. user will send SMS to GSM1 and it will send the sms message going out to GSM2 how do i accomplish this task? |
| 07:39.34 | *** join/#asterisk darkskiez (~darkskiez@usergc137.dsl.pipex.com) |
| 07:40.04 | Zeeek | libpcp I use SMS in both directions |
| 07:40.35 | Zeeek | you mean recv an SMS and redirect it to another GSM number? |
| 07:40.50 | Zgarbi | I cannot compile asterisk csv (today) |
| 07:40.59 | Zgarbi | channel.c: In function Á-?ast_readÁ-?: |
| 07:41.00 | Zgarbi | channel.c:1314: error: Á-?prestateÁ-? undeclared (first use in this function) |
| 07:42.35 | Zgarbi | channel.c:1314: error: (Each undeclared identifier is reported only once |
| 07:42.35 | Zgarbi | channel.c:1314: error: for each function it appears in.) |
| 07:44.35 | Zgarbi | line 1314: prestate = chan->_state; |
| 07:45.05 | Zeeek | libpcp hello? |
| 07:46.36 | Zgarbi | any solution for me? |
| 07:53.00 | *** join/#asterisk Nugget (nugget@dazed.slacker.com) |
| 07:58.44 | *** join/#asterisk sariabod (~sariabod@ip21.farheap.net) |
| 08:00.37 | *** join/#asterisk slePP (~slepp@S01060040f48412ad.ed.shawcable.net) |
| 08:01.11 | sariabod | I have 3 asterisk boxes, I use one to route incomming calls. But when I dial(IAX2/username:password@host) I cannot hear the background music or moh. In fact I cannot hear anything until the call is picked up. Anybody experiance this? |
| 08:01.38 | Nugget | why do you expect hold music when a call is being dialed? |
| 08:03.35 | sariabod | im dialing the other asterisk box, in that context I have a menu system (press 1 for... etc), this cannot be heard. The timeout context goes to a queue with moh and waits for the operator. The menu is not heard nor is the moh. |
| 08:04.09 | Nugget | that sounds like a different problem than you initially described. |
| 08:05.10 | Nugget | explain what you mean by "that context" and "the timeout context". both are unclear from what you've said so far. |
| 08:05.11 | sariabod | sorry, hard to explain, where is the whiteboard on this thing :) |
| 08:05.35 | Nugget | "the timeout context", in fact, makes no sense to me at all. |
| 08:05.40 | sariabod | lets start over |
| 08:05.45 | *** join/#asterisk Inv_Arp (junya@adsl-3-255-42.mia.bellsouth.net) |
| 08:06.43 | *** join/#asterisk Mimmus (~viggiani@ext.pitagora.it) |
| 08:06.56 | *** join/#asterisk slePP (~slepp@S01060040f48412ad.ed.shawcable.net) |
| 08:08.08 | Mimmus | I'm forwarding calls from Asterisk to Microsoft LCS via SER but I get only a 'one-way' communication |
| 08:08.28 | Mimmus | (Messenger users can hear voice but PSTN phone user cannot) |
| 08:08.42 | Mimmus | In SER log, I found: RFC3261 transaction matching failed |
| 08:09.04 | Mimmus | Googling, I found that this a known Asterisk bug |
| 08:09.07 | sariabod | [incomming] |
| 08:09.07 | sariabod | exten => s,1,Answer |
| 08:09.07 | sariabod | exten => s,2,Wait(2) |
| 08:09.07 | sariabod | exten => s,3,SetMusicOnHold(default) |
| 08:09.07 | sariabod | exten => s,4,Playback(welcome_menu); |
| 08:09.08 | sariabod | exten => t,1,Queue(operator_queue,t) |
| 08:09.21 | Nugget | you know you mispelled incoming? :) |
| 08:09.23 | Mimmus | I upgraded to 1.0.7 (Debian unstable) but bug is still present. |
| 08:09.27 | sariabod | yes |
| 08:09.32 | sariabod | but its everywhere |
| 08:09.40 | Nugget | I feel sorry for the next guy. :) |
| 08:09.45 | Mimmus | Can someone confirm thsi? |
| 08:11.17 | sariabod | when i call comes directly to box2, everything is fine. But, if the call comes into box1 then I dial box2 all the sounds cannot be heard, its just silence until somebody picks up |
| 08:11.19 | Nugget | first off, there's nothing in that dialplan that would engage any hold music. |
| 08:11.50 | Nugget | secondly, I suggest inserting some NoOp(I see a call!) log lines and then watching the console to make sure it's really doing what you think it is. |
| 08:12.27 | Nugget | most likely, your dial command isn't doing what you think it's doing. |
| 08:13.10 | Nugget | lastly, you'd be really well-served by setting up the appropriate entries in iax.conf so that you didn't need to embed login credentials into the dialplan/dial commands. there are many reasons for this, not the least of which is simplicity |
| 08:15.31 | Nugget | why do you expect dial(IAX2/username:password@host) to place the caller in the "incomming" [sic] context? |
| 08:16.21 | sariabod | the context for that "user" is set to incomming in iax.conf |
| 08:16.26 | Nugget | *nod* |
| 08:16.50 | Nugget | I've never tried to use an "s" extension as you've done, but that |
| 08:16.56 | Nugget | isn't to say it won't work/ |
| 08:17.13 | Nugget | I'd expect to see, instead, dial(IAX2/host/exten) |
| 08:17.19 | libpcp | Zeeek: are you still there pal? |
| 08:17.24 | Nugget | (assuming proper iax.conf entries for host) |
| 08:17.41 | Zeeek | yes libby |
| 08:17.49 | Zeeek | tell me your SMS problems |
| 08:18.01 | Zeeek | I feel your pain :) |
| 08:18.30 | Nugget | or, in your method, dial(IAX2/username:password@host/exten) |
| 08:18.39 | Nugget | but that's a bad habit |
| 08:18.47 | Mimmus | How can I know if BUG 0002687 is fixed? |
| 08:19.05 | Nugget | search for it on bugs.digium.com |
| 08:19.23 | Nugget | http://bugs.digium.com/bug_view_page.php?bug_id=0002687 |
| 08:19.52 | libpcp | Zeeek: i would like to ask if its possble that user send sms to gsm1 and forward it to gsm2 for out? |
| 08:20.09 | Zeeek | As far as I can see it would be |
| 08:20.12 | libpcp | something like gsm1--asterisk--internet--asterisk--gsm2 |
| 08:20.21 | Zeeek | If you are able to send and receive SMS |
| 08:20.33 | Mimmus | Nugget: I'm seeing a long bug history... |
| 08:20.43 | Zeeek | how does asterisk GET the SMS? |
| 08:20.54 | libpcp | the purpose is, the other asterisk box is located in other country and other one is in my place |
| 08:20.56 | Zeeek | mine gets it from PSTN |
| 08:21.18 | Zeeek | you need some kind of service to send/recv SMS |
| 08:21.26 | Zeeek | or maybe a cellsocket? |
| 08:21.26 | libpcp | i have gsm modem with sim card connected to zap card in both asterisk server |
| 08:21.32 | Zeeek | ok |
| 08:21.33 | Mimmus | Peraphs this is fixed in CVS but I'm not a compile guru... |
| 08:21.44 | Zeeek | so it's a question of the accounts on each gsm unit, no? |
| 08:22.30 | Nugget | running cvs head is not for the faint of heart or determination. |
| 08:22.44 | Nugget | but you could set it up on a scratch machine just to see if the bug persists |
| 08:22.55 | Mimmus | Nugget: I'm not in a production environment... |
| 08:23.15 | Nugget | it's probably worth building and fiddling, then, if you can spare the time |
| 08:23.40 | libpcp | Zeeek, if i connect a GSM phone to my asterisk box, is it possible to receive an sms message with asterisk? |
| 08:23.51 | Zeeek | yes. |
| 08:24.08 | darkskiez | Zeeek, sure? what module? |
| 08:24.10 | Zeeek | but how are you connected? |
| 08:24.29 | Zeeek | no - my bad - I was thinking of something totally different |
| 08:25.01 | Zeeek | There is a service that will handle SMS for you (I've seen it on mailing list) but I was thinking of landline SMS which I have here |
| 08:25.07 | darkskiez | libpcp, theoretically, yes, should be easy, I'm amazed nobody's knocked one up yet, could do it in an evening easily. |
| 08:25.34 | Zeeek | there is, as you all know, an SMS application which works fine |
| 08:25.48 | libpcp | darkskiez: really? how can i achieve that? |
| 08:25.51 | Zeeek | but how to get the SMS to asterisk other than thru FXO, I don't know |
| 08:25.53 | Mimmus | Nugget: do I try? Or it is an impossible mission? :-) |
| 08:25.54 | darkskiez | Zeeek, thats for over pstn lines though. |
| 08:26.00 | Zeeek | yes, mea culpa |
| 08:26.10 | Zeeek | I am not worthy of IRC |
| 08:26.13 | darkskiez | libpcp, you a coder? |
| 08:26.27 | *** join/#asterisk Delvar (~irc@83.146.53.34) |
| 08:26.33 | Nugget | Mimmus: it's pretty straightforward. follow the directions on http://www.asterisk.org/index.php?menu=download |
| 08:26.36 | libpcp | if my asterisk server can receive sms using a gsm phone or modem, how can i transfer that sms to my remote asterisk server and send out that sms to that gsm phone connected to that server? |
| 08:26.50 | Nugget | skip all the cvsup jazz, just go to the part about using cvs. |
| 08:26.59 | darkskiez | libpcp, thats a totally different question too. |
| 08:27.43 | libpcp | darkskiez: nope im not a hardcore coder |
| 08:27.44 | Zeeek | libpcp your goal is to be able to send SMS to another county free? |
| 08:28.01 | Zeeek | country |
| 08:28.21 | libpcp | Zeeek: difinetely yes |
| 08:28.41 | Zeeek | because you have a large amount of SMS traffic you want to send? |
| 08:29.12 | *** join/#asterisk fishboy1669 (proxyuser@62.69.81.129) |
| 08:29.18 | libpcp | nope, not that large traffic. its just only for a few sms message |
| 08:29.55 | Zeeek | I have a pay per call plan on my GSM phone, but it comes with free email. I can't imagine why you'd want to mess with 160 character messages |
| 08:30.12 | Zeeek | SMS is like fax! |
| 08:30.52 | dwmw2 | libpcp, asterisk can run a script of your choosing when it receives an SMS message |
| 08:30.56 | libpcp | darkskiez do you have any idea how can i achieve that? |
| 08:31.04 | fishboy1669 | hi zeeek hows things |
| 08:31.05 | RaYmAn-Bx | Zeeek: if you do need to send sms's between countries, sending them normally isn't that useful..I've had a fair amount get lost in the process... |
| 08:31.08 | dwmw2 | that script can send the same SMS message back out again via a different route |
| 08:31.17 | fishboy1669 | is anyone here from uk? |
| 08:31.25 | Zeeek | RaYmAn-Bx true, that's why SMS isn't my choice medium :) |
| 08:31.25 | dwmw2 | fishboy1669, yes |
| 08:31.31 | fishboy1669 | if so join #asterisk-uk |
| 08:31.41 | darkskiez | libpcp, not convinced you want to be using asterisk. but hey. |
| 08:31.42 | Zeeek | However I was surprised to see that sending them to another Eu country is not at all expensive |
| 08:31.57 | RaYmAn-Bx | Zeeek: not all phones support emails either anyways :P |
| 08:32.25 | Zeeek | geeze if you have a gsm phone that doesn't do GPRS or whatever that is... |
| 08:32.25 | RaYmAn-Bx | definitely..there are providers here where world-wide sms cost the same as local (and it's cheap as well) |
| 08:32.26 | libpcp | darkskiez: well if theres another option not even using asterisk is okay |
| 08:32.44 | *** part/#asterisk Mimmus (~viggiani@ext.pitagora.it) |
| 08:32.46 | RaYmAn-Bx | Zeeek: just because it does GPRS doesn't mean it can neccessarily do email :) |
| 08:33.36 | Zeeek | It does email if you provider has the web mail site |
| 08:33.46 | RaYmAn-Bx | exactly |
| 08:34.02 | Zeeek | also I wrote a little PHP thingie to read my messages via IMAP |
| 08:34.23 | Zeeek | limited to plain text and about 2000 chars but that's way better than SMS |
| 08:34.31 | Himeko | i have an email client right on my phone |
| 08:34.34 | Himeko | pop3 and imap4 |
| 08:34.46 | Zeeek | I have a whorehouse in my new Nokia |
| 08:34.57 | Zeeek | no other phone has that |
| 08:34.58 | libpcp | dwmw2: maybe you can help me how can i transfer the sms message to another asterisk server and send it out to gms modem |
| 08:35.01 | RaYmAn-Bx | heh |
| 08:35.22 | RaYmAn-Bx | Zeeek: GPRS traffic is generally a hell of a lot more expensive than sms's here |
| 08:35.39 | Zeeek | libpcp, you could have the SMS translated to audio and then have it read as voicemail :) |
| 08:36.08 | Zeeek | RaYmAn-Bx I *think* it costs about .10 to look at my mail, read a few messages |
| 08:36.15 | dwmw2 | sending the SMS message between asterisk servers is easy |
| 08:36.33 | dwmw2 | see app_sms |
| 08:36.54 | libpcp | Zeeek: thats a good idea but im sure users wont not easily go to internet cafe just to read the message that someone sent them |
| 08:37.10 | dwmw2 | when it's received by the remote server, just feed it to a script which sends it via the GSM modem. The documentation for app_sms shows you how to run a script upon receipt of a message |
| 08:37.40 | dwmw2 | If you need more help than that, I'm too lazy -- sorry :) |
| 08:37.44 | Zeeek | I think it's the receiving part that seems daunting |
| 08:37.55 | dwmw2_gone | receiving from the original GSM phone? |
| 08:38.02 | Zeeek | you're gone |
| 08:38.06 | Zeeek | but yes |
| 08:38.12 | dwmw2_gone | I've often been accused of that |
| 08:38.14 | dwmw2_gone | that isn't hard |
| 08:38.32 | Zeeek | libpcp you need to offer something to dwmw2_gone |
| 08:38.39 | Zeeek | he holds the key |
| 08:39.02 | dwmw2_gone | See ETSI GSM 07.07 |
| 08:39.03 | libpcp | Zeeek: pizza and a beer is that okay? with some girls :) |
| 08:39.12 | Zeeek | how many girls? |
| 08:39.14 | dwmw2_gone | whole bunch of AT commands for GSM phones |
| 08:39.27 | dwmw2_gone | including those you need for sending and receiving SMS messages |
| 08:39.32 | Zeeek | and you talk to it on serial port? |
| 08:39.41 | dwmw2_gone | something like libgsmme does this already |
| 08:39.53 | dwmw2_gone | Zeeek, serial, if you're old-fashioned. Or bluetooth, or whatever |
| 08:39.57 | dwmw2_gone | IrDA |
| 08:40.01 | Zeeek | aha |
| 08:40.14 | Zeeek | I don't know about any of this because I have a basic phone |
| 08:40.17 | *** join/#asterisk wasim (~wasim@203.81.218.248) |
| 08:40.48 | Zeeek | and now we have a SIP provider that offers real good rates for SMS and GSM calls |
| 08:41.12 | libpcp | dwmw2_gone: but how do i transfer the sms message to the other asterisk server? let say gsm1 receive an sms and it store in asterisk server 1, how do i transfer the sms to asterisk server 2 and send that message to gsm 2 ? |
| 08:41.48 | Zeeek | look up show application SMS and see the wiki for a full page about it |
| 08:41.53 | dwmw2_gone | libpcp, did you finish reading the app_sms documentation yet? |
| 08:42.20 | Zeeek | heh |
| 08:42.22 | dwmw2_gone | libpcp, for talking to the GSM phones, see http://gatling.ikk.sztaki.hu/~kissg/gsm/ |
| 08:42.41 | *** join/#asterisk kks (~kks@203.115.208.140) |
| 08:42.53 | dwmw2_gone | note that you can do cute things like turning on the voicemail/fax/email icons on the GSM handset, if you send the right type of messages |
| 08:43.15 | dwmw2_gone | I have asterisk running a voicemail notify script which turns the icon on/off on my gsm phone |
| 08:43.40 | *** join/#asterisk mithro (~tim@202.174.36.83) |
| 08:43.58 | mithro | anyone recommend a good softphone for asterisk for linux? |
| 08:44.12 | kks | Another newbie question, what is meant by "pass-thru mode" for g729 codec? |
| 08:44.22 | dwmw2_gone | see the documentation of 'Data Coding Scheme' field in §4 of GSM03.38 |
| 08:44.28 | *** join/#asterisk KoonS (~doudou@gate.noc.eurowan.net) |
| 08:44.40 | Zeeek | it means that asterisk won't transcode, the audio is not passing through asterisk |
| 08:45.15 | kks | ah i see, thanks Zeeek |
| 08:45.19 | dwmw2_gone | <PROTECTED> |
| 08:45.26 | dwmw2_gone | <PROTECTED> |
| 08:45.31 | libpcp | dwmw2_gone: yeah i finished reading application SMS but its a little hard for me to understand the whole function of that application |
| 08:45.53 | dwmw2_gone | there's examples of how to set it up to send and to receive messages |
| 08:46.24 | *** join/#asterisk oej (~oej@apollo.webway.se) |
| 08:46.49 | dwmw2_gone | the [smsmtrx] and [smsmorx] contexts, for sending and receiving SMS respectively. |
| 08:48.26 | Zeeek | actually that SMS doc is a little perplexing and the default directory structure where messages are stored changed without notice |
| 08:51.27 | jhiver | hi guys |
| 08:51.27 | libpcp | Zeeek: well lets assume that i can received sms message now on my asterisk server 1, my problem is how do i send it to asterisk server 2 and send that sms message out to gsm2, i know im a little stupid but i still cant figure out how to acheive that |
| 08:51.30 | libpcp | sorry |
| 08:51.43 | jhiver | I have this machine which had 1 X100P clone in it |
| 08:51.59 | Zeeek | libpcp it seems you'll have to do more research |
| 08:52.00 | jhiver | now I've put a TDM400P with 4 FXO modules in it |
| 08:52.18 | jhiver | it seems that the card is reckognized out of the box, which is cool |
| 08:52.28 | Zeeek | but... |
| 08:52.36 | jhiver | any pointers to what my zaptel.conf & zapata.conf should be? |
| 08:53.11 | Zeeek | change the line you had to 1-4 |
| 08:53.14 | jhiver | I've changed fxsks=1 to fxsks=1,2,3,4 does this make sense? |
| 08:53.17 | Zeeek | for the FXO |
| 08:53.45 | jhiver | (assuming 'kool start' is appropriate for french pstn, i have no idea) |
| 08:54.42 | jhiver | then in zapata.conf I just do channel => 1-4 and that's it? |
| 08:54.54 | Zeeek | why not try it and see? |
| 08:54.55 | *** join/#asterisk Aze` (~aze@host229-162.pool80105.interbusiness.it) |
| 08:55.08 | Zeeek | 1,2,3,4 should work too |
| 08:55.10 | jhiver | it doesn't hurt to ask see if you're doing something stupid :) |
| 08:55.16 | jhiver | ok |
| 08:55.27 | jhiver | I'll give it a shot :) |
| 08:55.27 | Zeeek | since it aonly takes about 1 minute to try, I'd have tried first |
| 08:55.47 | Zeeek | then there's always http://www.voip-info.org/wiki-Asterisk+config+zaptel.conf |
| 08:55.50 | jhiver | oh I have another issue actually |
| 08:56.21 | jhiver | is there a way to know which 'slot' at the back of the card is which channel? |
| 08:56.25 | Zeeek | which confirms that both syntaxes work (wiki) |
| 08:57.11 | Zeeek | it'll be easy once tha card is working :) |
| 08:57.15 | jhiver | lol |
| 08:57.21 | jhiver | yeah but for testing it sucks ) |
| 08:57.29 | Zeeek | the three red alarms will tell you |
| 08:57.39 | jhiver | red alarms? what's that? |
| 08:57.44 | jhiver | the stuff on the CLI? |
| 08:57.47 | Zeeek | when there's no line connected |
| 08:58.06 | jhiver | how do I 'see' those? is there a log somewhere? |
| 08:58.06 | Zeeek | so you have 4 analogue phone lines now? |
| 08:58.19 | jhiver | no, just 1 for now, the others are to arrive later |
| 08:58.33 | jhiver | i suppose I could always use a group... |
| 08:58.37 | jhiver | evil... |
| 08:58.49 | Zeeek | group for what? |
| 08:58.52 | Zeeek | outgoing? |
| 08:58.55 | jhiver | yeah |
| 08:58.58 | Zeeek | sure |
| 08:59.04 | jhiver | ok i'll do that |
| 08:59.29 | Zeeek | <PROTECTED> |
| 08:59.36 | Zeeek | from the wiki page |
| 09:02.02 | Zeeek | when you had the X100 in there didn't you ever see a red alarm? |
| 09:03.34 | jhiver | haven't hacked with Zap channels in quite a while... |
| 09:03.35 | jhiver | :) |
| 09:04.16 | Zeeek | ever try wengo? |
| 09:05.25 | libpcp | Zeeek: i have question. where does the sms message stores if i send an sms message to my gsm connected to asterisk server? |
| 09:06.14 | Zeeek | It depends on how the message is received. It comes into a directory in a certain format if you use SMS app |
| 09:06.51 | *** join/#asterisk clive- (~pirch@rndf-146-44-91.telkomadsl.co.za) |
| 09:07.40 | Zeeek | "/var/spool/asterisk/sms/" |
| 09:19.13 | *** join/#asterisk MuppetMaster (~MuppetMas@a82-92-73-185.adsl.xs4all.nl) |
| 09:19.25 | MuppetMaster | Hello |
| 09:19.30 | Zeeek | Hi! |
| 09:19.58 | MuppetMaster | Do the Digium cards have the ability to do call progress detection with Asterisk for outbound calls? |
| 09:20.25 | Zeeek | I think it depends on the country - or else the answer is no |
| 09:20.38 | Zeeek | there is a callprogress parameter ya see |
| 09:21.51 | MuppetMaster | Ah, okay. I guess it is certified in a limited number of countries. Like maybe only the US? |
| 09:22.06 | Zeeek | I don't even know if certified is the right word |
| 09:22.11 | MuppetMaster | The call progress parameter, is it controllable via the Manager API, or do you have to write a native driver to the card? |
| 09:22.15 | Zeeek | I'm pretty sure I have mine off (France) |
| 09:22.30 | Zeeek | the param is in zapata.conf |
| 09:23.00 | MuppetMaster | I will have a look and see if I can figure out how to leverage that and what countries it would work in. |
| 09:23.03 | Zeeek | in fact... |
| 09:23.05 | Zeeek | callprogress: Asterisk can attempt to monitor the state of the call to listen for a ringing tone, busy tone, congestion tone, and sounds indicating that the line has been answered. It appears that this feature is independent of the busydetect feature; it seems that both can run in parallel, and both will independently attempt to recognize a busy tone. The callprogress feature is highly experimental and can easily detect false answers, so |
| 09:23.05 | Zeeek | <PROTECTED> |
| 09:23.05 | Zeeek | "callprogress: Asterisk can attempt to monitor the state of the call to listen for a ringing tone, busy tone, congestion tone, and sounds indicating that the line has been answered. It appears that this feature is independent of the busydetect feature; it seems that both can run in parallel, and both will independently attempt to recognize a busy tone. The callprogress feature is highly experimental and can easily detect false answers, so |
| 09:23.15 | Zeeek | Hmmmmmm |
| 09:23.28 | Zeeek | Keybounce? |
| 09:23.33 | Zeeek | here's the page: http://www.voip-info.org/wiki-Asterisk+config+zapata.conf |
| 09:23.57 | MuppetMaster | Ah, I see. |
| 09:24.09 | MuppetMaster | I don't like the 'highly experimental' statement though... |
| 09:24.19 | MuppetMaster | Appears one would need to go Dialogic for that type of feature... |
| 09:24.35 | Zeeek | that's why I said I wasn't sure :) |
| 09:25.10 | MuppetMaster | AndHmmmm..... |
| 09:25.21 | Zeeek | phones really are the last bastion of non-standard standards |
| 09:25.45 | MuppetMaster | True. |
| 09:25.54 | Zeeek | so every time something doesn't work I wonder about our countrie's network |
| 09:26.08 | MuppetMaster | But, each country has a specified set of call progress tones that are well known. I was working with Dialogic and Nortels 10 years ago that could do this stuff. |
| 09:26.15 | MuppetMaster | In multiple countries and reliably. |
| 09:26.15 | Zeeek | yet, fax is universal - unless you use spandsp :) |
| 09:26.25 | MuppetMaster | Including France (a project with SEMA and FT Mobile). |
| 09:26.39 | Zeeek | FT Mobile is now Orange |
| 09:26.51 | MuppetMaster | I know, that was a long time ago... |
| 09:27.09 | MuppetMaster | Also, the 'Call Classifier' card is built into the Avaya, and it does call progress detection in most countries reliably as well. |
| 09:27.19 | MuppetMaster | If Digium is behind on this, they may be behind in other areas. |
| 09:27.29 | MuppetMaster | Like I said, this was doable 10 years ago. |
| 09:27.31 | Zeeek | must be wierd to have networks connected where in one place they reverse polarity and the other play DTMF tones |
| 09:28.00 | Zeeek | have I seen you on FWD forum by any chance? |
| 09:28.15 | MuppetMaster | Yes, you most likely have. I am a regular haunt on several forums. |
| 09:28.35 | MuppetMaster | I love Asterisk and want to be able to give back to Digium in everyway I can. But need full featured hardware too... |
| 09:28.51 | MuppetMaster | I will drop them a note and ask them where they are and where they are going on this front, as it is key to the application I am looking at. |
| 09:29.03 | Zeeek | good idea |
| 09:30.07 | facek_ | cypromis are you there? |
| 09:49.58 | *** join/#asterisk Jas_Williams (~Jason@host217-44-216-142.range217-44.btcentralplus.com) |
| 09:52.05 | *** join/#asterisk pascals (~pascal@ip503c8584.speed.planet.nl) |
| 09:53.11 | Zeeek | he's back? |
| 10:00.46 | *** join/#asterisk sob0l (~peter@uo166.internetdsl.tpnet.pl) |
| 10:00.56 | *** join/#asterisk eipi (~eipi@100-172-114-200.fibertel.com.ar) |
| 10:03.34 | *** join/#asterisk meppl (~mephisto@p3E9E2B55.dip.t-dialin.net) |
| 10:03.38 | meppl | guten morgen |
| 10:05.10 | clive- | what does sending VNAK mean ?..is it good/bad/ignorable ? |
| 10:08.54 | clive- | ~VNAK |
| 10:19.14 | *** join/#asterisk AngelGabriel (~angel@client-82-13-47-197.brhm.adsl.virgin.net) |
| 10:20.01 | AngelGabriel | Good morning people |
| 10:21.06 | libpcp | dwmw2: are you available now? |
| 10:25.48 | dwmw2 | libpcp: vaguely |
| 10:27.43 | *** join/#asterisk Mimmus (~viggiani@ext.pitagora.it) |
| 10:28.30 | Mimmus | how can I remove everything after a CVS compilation? |
| 10:30.02 | AngelGabriel | I would like to know if it is possible to announce something to the user, while he/she is on the phone, for example, the amount of time they have been on the line |
| 10:31.57 | sambal | AngelGabriel: for your girlfriend? :P |
| 10:32.14 | AngelGabriel | something like that!!! lol |
| 10:36.21 | *** join/#asterisk facek_ (faceoff@devel.acdbddh.eu.org) |
| 10:36.28 | facek_ | i have problem |
| 10:36.36 | facek_ | in show modules i dont have chan_zap |
| 10:36.41 | facek_ | how can i load it? |
| 10:39.21 | facek_ | anyone? |
| 10:41.07 | libpcp | dwmw2: i tried to connect my gsm phone to my zaptel card then i configure a context for sms but when i tried to send an sms to my gsm phone nothing happens |
| 10:41.18 | *** join/#asterisk langals (~icechat5@196.7.14.183) |
| 10:41.31 | libpcp | do you have a sample for sms incoming exten that is working? |
| 10:44.23 | langals | Hi there....wondering if anyone has used has used the Microsoft RTC Core? |
| 10:44.39 | langals | sorry about mentioning microsoft :-) |
| 10:45.00 | AngelGabriel | Is it possible to have a free incoming VOIP number? |
| 10:45.23 | facek_ | free.. i think not |
| 10:45.34 | dwmw2 | AngelGabriel: you're in the UK? see sipgate.co.uk |
| 10:45.35 | jakepdev | maybe not free - but minimal flat rate |
| 10:46.13 | dwmw2 | libpcp: how is the gsm phone connected to the zaptel card? |
| 10:46.14 | facek_ | in Poland is for one 2 dollars per month |
| 10:46.27 | facek_ | hwo to compiel asteirks with chan_zap module? |
| 10:46.28 | jakepdev | if you can get a michigan number for nufone - it's $2 a month I think |
| 10:46.53 | cypromis | just back from canada |
| 10:47.01 | dwmw2 | facek_: compile and install the zaptel and libpri stuff before building asterisk |
| 10:47.15 | dwmw2 | sipgate.co.uk do it for free |
| 10:47.43 | facek_ | dwmw2 i did that |
| 10:48.05 | facek_ | sipgate.co.uk - doesn't work |
| 10:48.40 | facek_ | dwmw2 i must have a symbolic link in /usr/src/asterisk to my source od asterisk? |
| 10:48.50 | libpcp | dwmw2: by using rj11, actually the gsm phone works if i set to another context like incoming call with IVR |
| 10:48.53 | dwmw2 | www.sipgate.co.uk |
| 10:49.08 | dwmw2 | your gsm phone has an rj11 socket? |
| 10:49.30 | libpcp | dwmw2: yeah |
| 10:49.34 | iceyp | my cisco 7940 has a rj11 jack dont know why |
| 10:49.44 | dwmw2 | bizarre. |
| 10:49.48 | dwmw2 | you don't need that. You need a _data_ connection to it |
| 10:49.53 | dwmw2 | as if you're going to use it as a modem. |
| 10:50.00 | dwmw2 | you need to be able to use AT commands |
| 10:50.22 | dwmw2 | I thought I already said that? You need to show some initiative |
| 10:50.22 | libpcp | actaully its not really a gsm phone, it is something like gsm modem |
| 10:51.23 | *** join/#asterisk mw` (id6686@p5480C262.dip.t-dialin.net) |
| 10:51.25 | mw` | hi |
| 10:51.28 | libpcp | dwmw2: ah, so i need a serial connection to my gsm modem? |
| 10:51.39 | mw` | someone knows why 2.6.11 breaks chan_capi? |
| 10:51.40 | dwmw2 | serial or IrDA or bluetooth, yes |
| 10:51.52 | *** join/#asterisk yaboo (~jsirucka@220.245.131.131) |
| 10:51.54 | dwmw2 | mw`:breaks in what way/ |
| 10:52.19 | *** join/#asterisk Gh0sty (~Ghosty@ip-81-11-220-172.dsl.scarlet.be) |
| 10:52.42 | libpcp | dwmw2: so this is impossible to work if i am connected to an RJ11 cable to my zaptel card. i thought i can make that work |
| 10:52.44 | mw` | dwmw2: hm i have only sound in one way with isdn <-> isdn and no sound with sip <-> isdn |
| 10:53.01 | dwmw2 | what isdn hardware / driver? |
| 10:53.20 | mw` | tested isdn <-> isdn with 2.6.10, works, now testing sip |
| 10:53.20 | dwmw2 | libpcp: I've never heard of a gsm phone with rj11 connector, but I assume that's only for voice. |
| 10:53.36 | mw` | dwmw2: fcclassic (suse/avm driver) |
| 10:54.50 | dwmw2 | fcclassic is AVM A1? |
| 10:54.54 | libpcp | dwmw2: yeah and i need a data cable for that to make it work |
| 10:54.58 | dwmw2 | the BT SpeedWay card? |
| 10:55.10 | mw` | dwmw2: yes, with isa conn (no pnp= |
| 10:55.39 | dwmw2 | I wasn't aware there were more drivers for that than the standard i4l hisax driver, and Karsten's new mISDN |
| 10:56.14 | dwmw2 | you could try mISDN |
| 10:56.14 | mw` | mISDN dosnt support isa cards |
| 10:56.14 | dwmw2 | ah |
| 10:56.15 | mw` | only pnp isa cards |
| 10:56.35 | dwmw2 | that can't be hard to fix, surely? |
| 10:56.47 | dwmw2 | they're basically identically apart from the detection, aren't they? |
| 10:56.48 | *** join/#asterisk MuppetMaster (~MuppetMas@a82-92-73-185.adsl.xs4all.nl) |
| 10:57.00 | mw` | dwmw2: yes ur right ;) |
| 10:57.17 | mw` | but im not that kernel hacker ;) |
| 10:57.21 | dwmw2 | not that I've been overly impressed with mISDN |
| 10:57.56 | *** join/#asterisk jhiver (~jhiver@AStDenis-103-1-14-65.w81-248.abo.wanadoo.fr) |
| 10:58.03 | mw` | but i think its the capi interface that breaks chan_capi |
| 10:58.03 | jhiver | hiya all |
| 10:58.07 | dwmw2 | mw`: even a non-kernel-hacker could manage that. |
| 10:59.01 | dwmw2 | I found a PCI AVM A1 lying around -- I was going to try setting that up over the weekend. |
| 10:59.01 | facek_ | Mar 31 12:58:51 WARNING[18312]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/chan_zap.so: undefined symbol: ast_retrieve_call_to_death |
| 10:59.13 | mw` | cause i use the same driver with 2.6.10 and 2.6.11 |
| 10:59.20 | mw` | dwmw2: isa? |
| 10:59.24 | dwmw2 | pci |
| 10:59.36 | mw` | so u use misdn? |
| 10:59.46 | dwmw2 | I was going to try :) |
| 11:00.06 | dwmw2 | I'm using mISDN on an HFC card at the moment, and it's not working well. |
| 11:00.11 | dwmw2 | it keeps reporting internal errors |
| 11:00.15 | dwmw2 | v.110 dialin doesn't work |
| 11:00.16 | mw` | there is also a small notice in voip-info.org |
| 11:00.19 | facek_ | can somebody help me with zap channels? |
| 11:01.17 | langals | Would someone be able to explain to me a codec frame rate is all about - the frame rate between my client and asterisk for G.711 is different, and thus it is not working properly |
| 11:01.19 | mw` | dwmw2: " * 5/4/05 linux-2.6.11-rc5 breaks chan_capi 0.3.5 " not more ;) |
| 11:04.50 | darkskiez | If I phone a queue from my sip phone and the call is placed in the queue in the ring state, and the phone powers off, the channels is left up active in the queue. |
| 11:04.54 | dwmw2 | hm. I don't see any relevant changes in the kernel |
| 11:06.56 | facek_ | Can somebody help me with load module for chan_zap ? ztcfg -vvv looks good. asterisk show zap channels looks good too.. but its unable to create channel when try to make call. |
| 11:09.24 | facek_ | hm? |
| 11:10.45 | *** join/#asterisk Delmar (~Delmar@222-152-57-78.adsl.inspire.net.nz) |
| 11:11.10 | Zeeek | hmmmm? |
| 11:11.21 | Zeeek | spit out the symptoms |
| 11:11.23 | Delmar | Got my TDM card today. ripped out the X100P, made NO changes, installed the TDM... No more echo. |
| 11:11.30 | Zeeek | Cool! |
| 11:11.38 | Delmar | other than drivers/modules of course. |
| 11:11.44 | Zeeek | of course |
| 11:11.51 | Delmar | no changes to zaptel/zapata or anything. |
| 11:11.56 | Delmar | it just owns. |
| 11:11.57 | Delmar | man |
| 11:11.59 | Zeeek | no channel changes? |
| 11:12.04 | Delmar | no. |
| 11:12.09 | Delmar | only had 1 X100P. |
| 11:12.12 | Zeeek | only one FXO in the TDM? |
| 11:12.17 | Delmar | yup |
| 11:12.26 | facek_ | Zeeek Can you help me little? |
| 11:12.29 | Zeeek | so your theory is that the FXO module is better? |
| 11:12.40 | Zeeek | facek_ just state your case - the court is waiting |
| 11:12.44 | Delmar | Zeeek actually my theory is quite sinister.... |
| 11:12.51 | Zeeek | Delmar let's hear it :) |
| 11:12.56 | facek_ | Zeeek i dont understand? my case? |
| 11:13.13 | Zeeek | facek_ stop asking about helping and ask your question |
| 11:13.22 | facek_ | Can somebody help me with load module for chan_zap ? ztcfg -vvv looks good. asterisk show zap channels looks good too.. but its unable to create channel when try to make call. |
| 11:13.32 | facek_ | without first " Can somebody help " |
| 11:13.33 | facek_ | solve problem |
| 11:13.59 | mw` | dwmw2: i give 2.6.10 a try and ask the author of capi_chan when he is in here |
| 11:13.59 | Zeeek | did you follow instructions from the wiki or asteriskdocs.org or somewhere? |
| 11:14.06 | Delmar | the fact is, there is a nice DSP with some good echo_can algs bolted into the silicon. there is no reason that those same tricks cant be used for an in-software DSP on the host system... and i suggest that they could do better .. but are not. |
| 11:14.52 | dwmw2 | hm, must send him my fix to make fax detection with on capi channels |
| 11:14.52 | Delmar | facek, unable to create a channel huh? |
| 11:15.05 | Zeeek | facek_ check the steps here: http://www.voip-info.org/wiki-Asterisk+Zaptel+Installation |
| 11:16.01 | Zeeek | modprobe zaptel and modprobe wcfxs or whatever other module you need |
| 11:16.03 | facek_ | thank |
| 11:16.15 | facek_ | anyone have a sipgate number? i want only call for test |
| 11:16.33 | Zeeek | sipgate has a test number |
| 11:16.43 | facek_ | 10000 |
| 11:16.44 | Zeeek | (but I forgot what it is) |
| 11:16.48 | facek_ | i know.. but call to somebody |
| 11:16.54 | AngelGabriel | Can someone explain to my what a trunk is? *no elephant jokes please* |
| 11:17.06 | facek_ | thay gave me that number 08700681943 |
| 11:17.09 | Zeeek | what you put your stuff in when taking a ship? |
| 11:17.24 | Zeeek | trunk is a combined stream that carries several channels |
| 11:17.25 | AngelGabriel | 0870? Isn't that a british number? |
| 11:17.30 | MuppetMaster | AngelGabriel: A trunk is a phone line, in simplest terms. |
| 11:17.32 | facek_ | AngelGabriel yes |
| 11:17.43 | facek_ | AngelGabriel is it possible to call for that number from other country? |
| 11:17.43 | AngelGabriel | i'm british |
| 11:17.48 | dwmw2 | AngelGabriel: sipgate.co.uk is british |
| 11:17.52 | facek_ | AngelGabriel Can you call for test? |
| 11:18.10 | MuppetMaster | So, if you say, have an E1, which is a digital line, you have 30B (bearer channels) and 2D (data channels) which use ISDN for signalling. |
| 11:18.18 | AngelGabriel | Zeeek, so, a trunk is like 10/20 lines bundeled together? |
| 11:18.20 | MuppetMaster | You would effectively have 30 trunks, each one that may carry one phone call. |
| 11:18.26 | *** join/#asterisk RestLessGemini (~umairbari@202.142.189.86) |
| 11:18.28 | MuppetMaster | In the world of VoIP, not really the same thing.... |
| 11:18.43 | dwmw2 | AngelGabriel: when I said "AngelGabriel: you're in the UK? see sipgate.co.uk" what part of that wasn't clear enough? :) |
| 11:19.03 | AngelGabriel | dwmw2, Sorry, i havn't had my hourly caffine fix yet! |
| 11:19.10 | dwmw2 | fair enough :) |
| 11:19.21 | dwmw2 | facek_: you want me to call that 0870 number from a land line? |
| 11:19.42 | AngelGabriel | I have 5 analogue lines, and I'm thinking of building an asterisk box |
| 11:19.43 | facek_ | yes |
| 11:19.46 | facek_ | dwmw2 yes |
| 11:20.18 | dwmw2 | Recorded message: "the service cannot be connected" |
| 11:20.21 | Zeeek | AngelGabriel what do you want to do (today)? |
| 11:20.23 | Delmar | Zeeek, i also reckon that the TDM, having a nice DSP on it, could filter out ADSL carrier easily. Thats exactly what the ADSL modems are doing to our voice, they use a DSP to filter the voice out. |
| 11:20.33 | Delmar | so no more ADSL line filter. |
| 11:20.43 | Zeeek | does the filter bother you? |
| 11:21.10 | dwmw2 | I'm not sure if that number can be dialled from outside the UK as +44870... |
| 11:21.11 | Delmar | yeah. you don't know what the government has put up them :P |
| 11:21.26 | facek_ | dwmw2 on 08700681943 ? |
| 11:21.26 | dwmw2 | certainly the geographic numbers ought to work. 0870 is special though |
| 11:21.28 | MuppetMaster | dwmw2: Generally no. |
| 11:21.30 | dwmw2 | facek_: yes |
| 11:21.32 | Zeeek | Delmar govt? I bought mine in an Asian store |
| 11:21.46 | Zeeek | 0870 is an expensive number to call |
| 11:21.58 | Zeeek | I had one and let it expire |
| 11:22.07 | dwmw2 | facek_: which is different to what I got after registering, but before I'd set up asterisk properly. |
| 11:22.09 | facek_ | Zeeek premium rate? |
| 11:22.10 | AngelGabriel | I want to be able to filter our incoming calls, and also route faxes to a fax machine, and have the ability to connect more than four phones to the four lines |
| 11:22.18 | AngelGabriel | @ Zeeek |
| 11:22.27 | Zeeek | yes I think it may be 3-5p or more |
| 11:22.48 | darkskiez | www.saynoto0870.co.uk |
| 11:22.53 | Zeeek | angler_ ah so now it's 4 lines? (+fax?) |
| 11:23.00 | *** join/#asterisk tzafrir_laptop (~tzafrir@62.90.10.53) |
| 11:23.34 | *** part/#asterisk MuppetMaster (~MuppetMas@a82-92-73-185.adsl.xs4all.nl) |
| 11:23.47 | Zeeek | oops AngelGabriel |
| 11:23.59 | Zeeek | darkskiez thanks for the URL! |
| 11:24.10 | Delmar | yep. im convinced that the echo_can algs in the TDM card are far superior to the in-software echo_can's being made available in the zaptel drivers. |
| 11:24.13 | AngelGabriel | Zeeek, I currently use one line dedicated for fax |
| 11:24.20 | Zeeek | <PROTECTED> |
| 11:24.32 | Zeeek | AngelGabriel so you'd want one TDM400 with 4 FXO modules |
| 11:24.32 | Delmar | they could do way more for echo_can at the host but are not. |
| 11:24.50 | AngelGabriel | I've actually got 5 lines |
| 11:24.58 | Zeeek | and then a second TDM if you want analog FXS lines for phones |
| 11:25.00 | AngelGabriel | One is dedicated to fax |
| 11:25.01 | Delmar | AngelGabriel drop the 5th 1 |
| 11:25.10 | Delmar | AngelGabriel, use spandsp |
| 11:25.13 | Zeeek | AngelGabriel No! Keep the 5th for fax only |
| 11:25.23 | Delmar | AngelGabriel, no dop it :P |
| 11:25.27 | Zeeek | no don't use spandsp (unless it doesn't matter when faxes are unreadable) |
| 11:25.29 | AngelGabriel | Drop it, keep it, drop it, keep it ... hmmm ??? |
| 11:25.37 | Zeeek | keep it! |
| 11:25.54 | Zeeek | until you can prove spandsp works for the faxers you'll have |
| 11:26.00 | Delmar | AngelGabriel how often do u get a fax that doesn't go " beeep..... beeep.. " ? |
| 11:26.21 | Zeeek | ah yeah, tyou could route them to a fax |
| 11:26.38 | dwmw2 | Delmar: most faxes go beeeep, surely? |
| 11:26.40 | Zeeek | but spandsp is imperfect in reception, many of us have problems with certain machines |
| 11:26.59 | AngelGabriel | i get about 50 faxes a day - and the phone/fax machines, you know, them combo machines, are unbelivably dumb |
| 11:27.01 | Delmar | spandsp detects the "CNG" tone.. and dont get me wrong.. there ARE some old fax machines out there that dont generate the FAX CNG tone.... |
| 11:27.05 | Zeeek | so you can drop the line but keep the fax machine :) |
| 11:27.21 | Delmar | but when i come accross someone that doesnt have a Fax CNG tone, i tell them not to fax me ever the fuck again until they fix it. |
| 11:27.30 | Zeeek | Delmar many times people still complain about distorted unreadable faxes |
| 11:27.37 | dwmw2 | CNG is the 2100Hz tone? |
| 11:27.53 | Zeeek | it isn't the tone I'm talking about but timing errors or frame shift or whatever it's called |
| 11:27.59 | AngelGabriel | Is it possible - to add a carrier to a call - for example, I have cheap calls to the US, so if someone calls a US based number, can I prefix what they dialled with my carrier codes? |
| 11:28.02 | Delmar | Zeeek, no im not talking about routing a fax over VoIP. use spandsl to receive the fax and email it via .PDF. |
| 11:28.02 | dwmw2 | er, 1100 |
| 11:28.25 | Zeeek | Delmar all beautiful in theory and it works 66.66% of the time for us |
| 11:28.28 | Delmar | AngelGabriel, can do that and more with dialplans. |
| 11:28.36 | dwmw2 | AngelGabriel: yes |
| 11:28.44 | Zeeek | so we'd only have to call back 1 in 3 customers to say we aren't capable of receiving their fax |
| 11:28.45 | Delmar | Zeeek, oh fuck here we go.. now my plan is out the window.... |
| 11:28.51 | dwmw2 | I already do that for outgoing calls to 001800 numbers |
| 11:28.55 | dwmw2 | which are free over VoIP |
| 11:28.58 | AngelGabriel | Delmar, dialplans?! GREAT! |
| 11:29.00 | Delmar | Zeeek, i have heard so much success tho.. whats the deal? |
| 11:29.16 | Zeeek | look at the mailing list or stay here long enuf |
| 11:29.24 | Zeeek | I am buying a fax machine soon |
| 11:29.37 | Zeeek | which I think is ABSURD in this day and age |
| 11:29.38 | AngelGabriel | wonders if Delmar and Zeeek are about to enter the ring! |
| 11:30.04 | Delmar | Zeeek, ah fuck. i guess im gonna go back to my good old ComShare fax switch and route to Relayfax on my server then. |
| 11:30.16 | dwmw2 | AngelGabriel: see http://bugs.digium.com/bug_view_page.php?bug_id=0003889 for example |
| 11:31.01 | dwmw2 | special routing for some numbers, but as soon as what the user dials has diverged and you know it's going out the real phone line, pick up the line, send the digits which were already dialled, and keep sending the rest of the digits as they arrive. |
| 11:31.14 | dwmw2 | so you get the proper feedback immediately from BT as you dial |
| 11:31.14 | Delmar | Zeeek, well... the trouble people are having.. is it more.. X100 or TDM.. or people with BR/Pri digital trunks using spandsp also having trouble? |
| 11:31.50 | dwmw2 | spandsp 0.0.2pre11 is supposed to be much more tolerant of timing errors |
| 11:31.52 | Delmar | i would be thinking that a TDM with no echo kinda deal should be better with spandsp than an X100. |
| 11:31.55 | Zeeek | Delvar my own is X100P |
| 11:32.08 | Delmar | right. ok |
| 11:32.11 | Zeeek | It isn't echo - some machines work, others do not |
| 11:32.22 | Delvar | ? |
| 11:32.26 | Zeeek | however, maybe the day I get a TDM with FXO it'll be great |
| 11:32.26 | Delmar | some senders fax machines work and some dont? |
| 11:32.31 | Zeeek | yes sir |
| 11:32.39 | Zeeek | For example, |
| 11:32.40 | Delmar | heh |
| 11:32.42 | Delmar | hang on dude. |
| 11:32.47 | Delmar | hold it right there... |
| 11:32.47 | Zeeek | j2.com works perfectly |
| 11:33.08 | Zeeek | I also got a couple of spam faxes with perfect reception |
| 11:33.10 | facek_ | Zeeek |
| 11:33.10 | Zeeek | BUT |
| 11:33.32 | Delmar | i was having problems with relayfax and an external modem when a supplier would send me a fax... the same supplier could never fax us when received by the PC's modem.. BUT the external modem worked fine... |
| 11:33.33 | Zeeek | I asked a customer to fax us and they have a really good machine, and the fax won't sync |
| 11:33.36 | AngelGabriel | dwmw2 - thanks for the link! *bookmarked* |
| 11:33.38 | Delmar | it almost sounds like the same shit. |
| 11:33.49 | facek_ | i want to write an context. i will put file in /var/lib/asterisk/outgoing/ and that context after connect to sb will Playback z audio file.. fut after connection not before |
| 11:34.21 | Zeeek | now, when I try to fax with a laptop fax program which works flawlessly wiuth any machine I've ever tried, spandsp won't even start reception |
| 11:34.23 | *** join/#asterisk lbarth (user@pD9563464.dip.t-dialin.net) |
| 11:34.40 | lbarth | hello |
| 11:34.46 | Zeeek | facek_ look at .call files on the wiki |
| 11:34.55 | Zeeek | it''s all there |
| 11:35.01 | AngelGabriel | I have an idea/question - I want someone to pick up the handset - and instantly hear a message, like please dial your number. Once the number is dialled, it's repeated back to the user, and then routed over the cheapest route - *international carriers, we use a few* |
| 11:35.05 | facek_ | Zeeek i know .call files. but what about context |
| 11:35.34 | lbarth | anybody out there can help me with a realtime problem? |
| 11:35.43 | Delmar | AngelGabriel, handset = what? |
| 11:35.47 | *** join/#asterisk tessier (~treed@210.245.97.9) |
| 11:35.52 | dwmw2 | AngelGabriel: that isn't hard |
| 11:35.54 | Delmar | AngelGabriel, sip client? channelbank extension? |
| 11:36.23 | AngelGabriel | Delmar, either softphone or hardphone, and device that is connected to the asterisk box, and requests an open line |
| 11:36.25 | dwmw2 | When they pick up the line and haven't dialled, they get into the 's' extension of whatever dialplan context that line is configured to start in |
| 11:36.41 | dwmw2 | you can play a message, then wait for an extension, then do whatever else you want. |
| 11:36.51 | dwmw2 | do you really need to wait and then repeat it for them? |
| 11:36.59 | dwmw2 | why not just let them dial and route it appropriately? |
| 11:37.12 | Zeeek | AngelGabriel you may mean what immediate=yes does - dials a number when the p^hone is picked up |
| 11:37.16 | dwmw2 | "The number you have dialled is 9-9-9" |
| 11:37.23 | *** join/#asterisk mw` (id1287@p5480CECC.dip.t-dialin.net) |
| 11:37.26 | Zeeek | that would be for ZAP |
| 11:37.31 | Delmar | dwmw2, so hows that going to work for a sip client? Xlite generates a local dialtone waiting for a phone number... you cant just go off hook and have it send "blank" ?? |
| 11:37.38 | mw` | dwmw2: with 2.6.10 it works perfectly |
| 11:37.49 | AngelGabriel | dwmw2, Well, no, not really, no need to repeat it to them. it's the routing that has got me worried, because we call all over the world |
| 11:37.50 | dwmw2 | Delmar: I don't think AngelGabriel was talking about SIP clients. |
| 11:37.53 | Zeeek | most SIP phones have an autodial setting, no? |
| 11:37.53 | facek_ | Zeeek in context i want start playback after connection |
| 11:38.06 | mw` | dwmw2: but theres no sound, because capi_chan doesnt detect that someone picked up the call |
| 11:38.06 | lbarth | I would like to sync 2 Servers using one realtime Database. one Server is running HEAD-01/30/05 the other Server is running today's head |
| 11:38.11 | Zeeek | facek_ what is stopping you? |
| 11:38.16 | Zeeek | The dialplan is the heart of an Asterisk system, as it defines how Asterisk should handle calls. |
| 11:38.16 | Zeeek | http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current_v1/docs-html_one/vm1.html#AEN650 |
| 11:38.25 | Delmar | lbarth, what database. |
| 11:38.27 | mw` | dwmw2: xlite still says "ringing.." |
| 11:38.29 | dwmw2 | Mw3: what if you try earlyb3? Prepend 'b' to the number you dial |
| 11:38.30 | Delmar | lbarth, sql? |
| 11:38.33 | lbarth | odbc / myql |
| 11:38.43 | dwmw2 | which isn't a fix, but might be a sane workaround |
| 11:38.43 | lbarth | the database is working fine |
| 11:38.48 | mw` | dwmw2: already tried it |
| 11:38.49 | *** join/#asterisk sandnigg0r (~crunkuser@66-55-197-254.gwi.net) |
| 11:38.59 | Delmar | lbarth, have you searched for howto slave/master sql database? |
| 11:39.00 | facek_ | Zeeek but Dial is maked by file in outgoing directory |
| 11:39.13 | Zeeek | what? |
| 11:39.14 | dwmw2 | AngelGabriel: the routing is easy enough. That mantis URL gives examples including 001800. You just add others to match. |
| 11:39.18 | lbarth | @Delmar sqlsync is not the Problem. |
| 11:39.23 | mw` | dwmw2: do you use chan_capi? |
| 11:39.25 | *** join/#asterisk cjk (~cjk@80.92.64.103) |
| 11:39.34 | Zeeek | facek_ there is a firled for context in the .call file - you know that since you read the doc |
| 11:39.36 | dwmw2 | Mw3: I've just switched to using it so that I can get earlyb3, yes. |
| 11:39.38 | dwmw2 | on 2.6.10 |
| 11:39.39 | Zeeek | a field |
| 11:39.54 | dwmw2 | hehe |
| 11:39.55 | Delmar | lbarth the problem is that they are not the same * version? |
| 11:39.56 | dwmw2 | users. |
| 11:40.12 | mw` | dwmw2: dont upgrade to 2.6.11 *duck* :) |
| 11:40.17 | dwmw2 | I blame the admins who let them use windows boxen |
| 11:40.22 | Delmar | yes, stay away of 2.6.11 |
| 11:40.30 | lbarth | the problem is that I alway's revcive db.c:177 ast_db_get: Unable to find key 'XXXX' in family 'SIP/Registry' |
| 11:40.31 | Zeeek | Windows is great |
| 11:40.40 | dwmw2 | mw`: actually I was going to do that this weekend. I'll deal with that when it happens |
| 11:40.43 | AngelGabriel | dwmw2, so, can i prefix any dialled numbers with an 0800 access number, followed by an access code? |
| 11:40.46 | lbarth | it seams tha the current head is not making correct SQL Lookups |
| 11:40.52 | mw` | Delmar: do you also have problems with 2.6.11? |
| 11:40.55 | dwmw2 | AngelGabriel: yes |
| 11:41.06 | *** join/#asterisk FaithX (~Faithful@202-6-145-116.ip.adam.com.au) |
| 11:41.07 | AngelGabriel | dwmw2, brilliant - that's what i wanted to know |
| 11:41.17 | Delmar | mw` i gave it a shot.. went right back to 2.6.10 |
| 11:41.29 | dwmw2 | so if you have special call rates for France via some carrier, for example, you'd do |
| 11:41.39 | Zeeek | AngelGabriel you might find these articles interesting as an intro |
| 11:41.40 | Zeeek | Starter tutorial: |
| 11:41.40 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html |
| 11:41.40 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html |
| 11:41.40 | Zeeek | http://www.automated.it/guidetoasterisk.htm |
| 11:41.40 | Zeeek | THE reference of the moment: |
| 11:41.42 | Zeeek | http://www.asteriskdocs.org |
| 11:41.54 | AngelGabriel | thanks Zeeek |
| 11:42.01 | Delmar | the problem i had which only cropped up after going to 2.6.11.4 was .. 30 seconds of audio (via my voip provider) then audio dead... went back to 2.6.10 and problem went away. |
| 11:42.06 | Delmar | that was enuf of a problem for me. |
| 11:42.07 | Zeeek | I know they are basic but they do give a great leg up |
| 11:42.27 | dwmw2 | exten => _0039+,1,Dial(Whatever/0800123123123123${EXTEN}) |
| 11:42.40 | dwmw2 | can't remember how to insert pauses off-hand, but it shouldn't be hard |
| 11:42.46 | Zeeek | no + |
| 11:42.56 | dwmw2 | Zeeek: see http://bugs.digium.com/bug_view_page.php?bug_id=0003889 |
| 11:43.05 | Zeeek | ah something new? |
| 11:43.26 | AngelGabriel | i have to go deal with two idiot OE users - who have fucked thier machines |
| 11:43.28 | AngelGabriel | <PROTECTED> |
| 11:43.31 | Zeeek | is that gonna be in the next version? |
| 11:43.36 | dwmw2 | yeah. Actually it probably won't be the '+' character by the time it gets applied, because people use '+' at the beginning of numbers for other reasons |
| 11:43.37 | lbarth | have a look at mantis bug 0003509 i think there is a problem |
| 11:43.54 | dwmw2 | Zeeek: not sure. I was expecting a response from Mark last night but there was none |
| 11:44.08 | Delmar | lbarth ah i remember helping someone in here not long ago with an SQL problem... something needed changing because it broke. |
| 11:44.19 | dwmw2 | but we need _some_ kind of wildcard which will work both with DTMF and non-DTMF dialling |
| 11:44.25 | Delmar | lbarth, im a bit too tired to remember tho. it wasnt anything too hard. |
| 11:44.35 | Zeeek | that's why the + sucks |
| 11:44.44 | lbarth | was it something in the code or in the config? |
| 11:44.51 | Zeeek | too bad the 16key pâds weren't standard |
| 11:44.54 | Delmar | i hate all those +'s in patches. i do it all manually :P |
| 11:44.56 | Zeeek | with ABCD |
| 11:45.05 | dwmw2 | in the example above, '0039' would match DTMF immediately and start the outgoing connection, but you'd need '_0039.' to match something like a SIP client dialling the full number up-front |
| 11:45.29 | Delmar | lbarth, that time it was a change to the way things are done in the * config to make the SQL query work again dude. |
| 11:46.14 | Delmar | but hey, it might not even be the same deal but its a safe bet that if it works ok on one CVS version, and now it dont work on another...something needs to be un-b0rked. :P |
| 11:46.18 | facek_ | Zeeek my .call file have a Channel: Context, Extension and PRiority |
| 11:46.30 | facek_ | and.. hwo to start that ocntext after connection not before |
| 11:46.56 | Zeeek | facek_ you aren't really explaining your problem very well - or at least I don't understand it. |
| 11:47.09 | Zeeek | there can be no connection until you dial |
| 11:47.11 | Delmar | ok im gonna hit the sack before i face plant this keyboard. |
| 11:47.12 | Delmar | l8r |
| 11:47.17 | Zeeek | ehh |
| 11:47.22 | lbarth | delmar, I'll have again a look at the 3 new parameters... thanks a lot |
| 11:47.23 | Zeeek | or heh |
| 11:47.24 | facek_ | I want that.. i put file in /var/lib/asterisk/outgoing/ with phone number to call and extension to run |
| 11:47.45 | facek_ | asterisk ic calling for that number.. and when sb answer call. he * is playing some file |
| 11:47.48 | facek_ | how |
| 11:47.54 | Zeeek | and in that extension you will answer the call |
| 11:48.07 | Zeeek | by reading the elementary documentation about dialplans |
| 11:48.22 | facek_ | Answer call in extension? no.. this is calling for my cellular phone |
| 11:48.39 | Zeeek | you could have mentioned that one hour ago |
| 11:49.17 | facek_ | y-y |
| 11:49.29 | Zeeek | so, to sum up and in case anyone knows if there is way to do this.... |
| 11:49.56 | Zeeek | is it possible to initiate file playback from asterisk afetr an asterisk initiated call is answered? |
| 11:50.09 | Zeeek | I don't see how, but there may well be one |
| 11:50.27 | Zeeek | maybe as an option of dial |
| 11:50.54 | Zeeek | yes, this looks like it miht work |
| 11:50.57 | Zeeek | <PROTECTED> |
| 11:51.11 | Zeeek | facek_ look at the show application Dial |
| 11:52.10 | facek_ | ok |
| 11:52.58 | dwmw2 | AngelGabriel: btw are you aware of the existence of #asterisk-uk ? |
| 11:53.26 | *** part/#asterisk lbarth (user@pD9563464.dip.t-dialin.net) |
| 11:53.40 | *** join/#asterisk lbarth (user@pD9563464.dip.t-dialin.net) |
| 11:54.14 | facek_ | Zeeek you knwo how yo use that in me example? |
| 11:54.31 | Zeeek | I just tested it |
| 11:56.55 | facek_ | and what effect |
| 11:58.11 | lbarth | hi |
| 11:58.38 | lbarth | is there any problem with the cvs HEAD? I can no longer compile it |
| 11:58.50 | lbarth | channel.c: In function `ast_read': |
| 11:58.50 | lbarth | channel.c:1314: error: `prestate' undeclared (first use in this function) |
| 11:58.50 | lbarth | channel.c:1314: error: (Each undeclared identifier is reported only once |
| 11:58.50 | lbarth | channel.c:1314: error: for each function it appears in.) |
| 11:58.51 | lbarth | make: *** [channel.o] Error 1 |
| 11:59.40 | lbarth | i'll wait a few minutes.. maybe someone is just commiting stuff |
| 12:01.19 | *** join/#asterisk zoa (~zoa@pirus.securax.be) |
| 12:02.50 | zoa | anyone awake with knowledge on iax2 native bridging and huge delay with iaxclient ? |
| 12:03.48 | Mimmus | lbarth: I'm compiling CVS just now! |
| 12:04.01 | Mimmus | lbarth: same error. You need to compile zaptel and libpri too |
| 12:04.11 | *** join/#asterisk shadebob (~shadebob@rnis-162-206-192-81.marocconnect.com) |
| 12:05.14 | lbarth | Mimimus, ah.. thats caused by the update to the bridging code |
| 12:05.21 | lbarth | I' ll try it |
| 12:07.06 | *** join/#asterisk zotz (~zotz@24.231.32.191) |
| 12:07.40 | lbarth | i've yust make && make install libpri.. still the same error :- |
| 12:08.19 | Mimmus | lbarth: firstly compile zaptel |
| 12:08.54 | lbarth | mimmus: ok, I didn't wanted to install zaptel.. now i have to |
| 12:09.19 | Mimmus | zaptel: it is neede in some way; read http://www.voip-info.org/wiki-Asterisk+Zaptel+Installation |
| 12:09.42 | clive- | zoa speak to stevek |
| 12:10.01 | lbarth | i know how to install it. But it was nice to have * running without it. |
| 12:10.36 | lbarth | mimmus: I dont linke this kernel module, compiling stuff :-) |
| 12:10.47 | Mimmus | lbarth: It is needed at least for ztdummy |
| 12:11.33 | dwmw2 | the wiki lies |
| 12:11.39 | dwmw2 | you don't need a full kernel source |
| 12:11.47 | lbarth | no i dont? |
| 12:11.54 | dwmw2 | the distribution should have enough in /lib/modules/`uname -r`/build |
| 12:12.13 | Mimmus | lbarth: I followed wiki instructons |
| 12:12.24 | clive- | lbarth why do you need libpri to do native bridging? |
| 12:13.01 | dwmw2 | you don't need the full source -- only the headers and top-level Makefiles |
| 12:13.21 | dwmw2 | in Fedora this is part of the kernel rpm (in <= FC3) or kernel-devel rpm (FC4+) |
| 12:13.23 | lbarth | clive: the latest heads need it |
| 12:13.37 | dwmw2 | other distributions are similar, I believe |
| 12:13.48 | clive- | lbarth strange...oh well |
| 12:13.51 | lbarth | I have suse 9.1 running over here |
| 12:14.17 | dwmw2 | lbarth: look in /lib/modules/`uname -r`/build |
| 12:14.49 | lbarth | dwmw2: /lib/modules/../build is linked to the non existing dir /usr/src/linux-2.6.9 :-) |
| 12:15.00 | dwmw2 | doh |
| 12:15.03 | dwmw2 | file a bug then :) |
| 12:15.15 | lbarth | at SUSE :-) |
| 12:15.15 | dwmw2 | or check if there's a 'kernel-devel' package or something similar |
| 12:15.17 | dwmw2 | yes |
| 12:15.33 | lbarth | I'll solve it somehow.. |
| 12:16.24 | lbarth | urghs.. the kernel is commpiled with gcc 2.95 and the system is running 3.3.3 |
| 12:16.38 | lbarth | so I need to compile a new kernel.. shit |
| 12:20.33 | Mimmus | I just finished to compile Asterisk from CVS for my very first time |
| 12:23.18 | Mimmus | I need chan_capi: do I need to compile this too or can I use Debian module? |
| 12:24.39 | lbarth | mimmus:compiling is quite sipmle but for the head you'll need a patch |
| 12:25.41 | lbarth | http://www.levigo.de/VoIP/chan_capi-0.3.5-cvs-HEAD-patch.tar.bz2 |
| 12:26.58 | Mimmus | lbarth: do I need to apply this against chan_capi source? |
| 12:27.53 | darkskiez | cvs head doesnt compile if ZAPTEL_OPTIMIZATIONS is not defined. |
| 12:28.00 | darkskiez | if anyone cares. |
| 12:29.23 | dreamcode | how do i get call parking to work, i have include = parkedcalls in my dialplan , what's wrong ? |
| 12:31.52 | *** join/#asterisk lately (~doug@80.4.233.220.exetel.com.au) |
| 12:32.23 | dreamcode | pls .. help |
| 12:33.47 | lately | In what cases would I want to use the command Switch over Dial(IAX2/...) ? |
| 12:38.40 | Mimmus | lbarth: I applied patch successfully but compiling gives me a lot of errors |
| 12:39.12 | Mimmus | as these: |
| 12:39.15 | Mimmus | ras:~/chan_capi-0.3.5# make install chan-capi |
| 12:39.15 | Mimmus | gcc -pipe -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/usr/include/a |
| 12:39.15 | Mimmus | sterisk -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DCAPI_ES -DCAPI_GAIN -DCAPI |
| 12:39.15 | Mimmus | _SYNC -DUNSTABLE_CVS -Wno-missing-prototypes -Wno-missing-declarations -DCRYPTO |
| 12:39.15 | Mimmus | <PROTECTED> |
| 12:39.16 | Mimmus | In file included from /usr/include/time.h:38, |
| 12:39.18 | Mimmus | <PROTECTED> |
| 12:39.20 | Mimmus | <PROTECTED> |
| 12:39.20 | facek_ | Zeeek ? |
| 12:39.22 | Mimmus | <PROTECTED> |
| 12:39.24 | Mimmus | re "typedef" |
| 12:39.25 | Mimmus | In file included from /usr/include/pthread.h:21, |
| 12:39.28 | Mimmus | <PROTECTED> |
| 12:39.29 | Mimmus | <PROTECTED> |
| 12:39.33 | clive- | flood,,help |
| 12:39.55 | Mimmus | Sorry, right-click made the tragedy! |
| 12:42.18 | *** join/#asterisk dmccollum (~dmccollum@69-164-245-72.atlaga.adelphia.net) |
| 12:42.25 | clive- | mimmus seems like the patch isnt up to date |
| 12:42.49 | wt | is there such a thing as a 802.3af powered ATA? |
| 12:43.14 | dmccollum | anyone here that might could help me try to figure out why a .call file isn't working? |
| 12:43.22 | *** join/#asterisk riksta (~rick@81-178-219-201.dsl.pipex.com) |
| 12:43.23 | facek_ | dmccollum i can try |
| 12:43.33 | Mimmus | clive-: I downloaded it just now from http://www.levigo.de/VoIP/chan_capi-0.3.5-cvs-HEAD-patch.tar.bz2 |
| 12:43.49 | Makenshi | wt, would be nice :) |
| 12:43.50 | dmccollum | ok. let me post the .call file and the asterisk console output up at pastebin. |
| 12:44.03 | Makenshi | wt, though not all 802.3af devices are made equal |
| 12:44.20 | facek_ | sb wrote chan_skype? ;] |
| 12:44.51 | wt | I am thinking of http://www.thevoipconnection.com/store/catalog/product_16172_NETGEAR_242_Managed_Switch_with_PoE.html |
| 12:44.59 | wt | for the switch side |
| 12:45.12 | Makenshi | i have some poe switches and power injectors for our phones |
| 12:45.42 | Makenshi | 3com make a 802.3af ata, however it requires nbx 4.2 or later |
| 12:45.56 | wt | nbx 4.2? |
| 12:45.56 | *** join/#asterisk Gostt (~Paulo@200.223.105.24) |
| 12:45.58 | wt | what is that? |
| 12:46.08 | *** join/#asterisk LeoB (~chatzilla@c-66-31-41-1.hsd1.ma.comcast.net) |
| 12:46.21 | clive- | mimmum which cvs head is that ?...I mean what date? |
| 12:46.30 | Zeeek | I just tested itok |
| 12:46.31 | Makenshi | operating software |
| 12:46.36 | Zeeek | whoa there |
| 12:47.06 | Makenshi | they do say that it will be sip compatible in the future |
| 12:47.26 | LeoB | novice question: can anyone help me connect a softphone to asterisk? |
| 12:47.53 | wt | LeoB: which softphone? |
| 12:48.03 | Mimmus | my problem with chan_capi compiling seems to be present somewhere... I found some references in Google |
| 12:49.29 | dmccollum | facek: I posted the info at http://pastebin.ca/8526 |
| 12:50.44 | Zeeek | anyone about IAX, I have a weird thing I just saw |
| 12:50.46 | Zeeek | http://pastebin.ca/8527 |
| 12:54.06 | *** join/#asterisk Taqyon (Taqyon@wblv-146-211-102.telkomadsl.co.za) |
| 12:54.19 | Taqyon | Hi All |
| 12:54.26 | Taqyon | Is there an FAQ for this channel? |
| 12:54.45 | dmccollum | facek: I just solved my problem. I didn't have the context set right in the scripts. |
| 12:55.15 | facek_ | dmccollum ok |
| 12:55.30 | wt | Makenshi: where can I find info about that device? |
| 12:55.36 | wt | I can't find it on 3com.com |
| 12:55.37 | facek_ | anyone have some interested PBX and IVR dialplans.. with most feautres possible in asterisk |
| 12:55.40 | dmccollum | amazing how much your clarity is in the morning after a good nights sleep. heh |
| 12:57.46 | Mimmus | dmccollum: here in Italy it's 3:00 PM and I'm beginning to be very tired and bored |
| 12:57.47 | *** join/#asterisk scanna (~scannachi@81-174-16-211.f5.ngi.it) |
| 12:58.06 | Makenshi | wt, the model number is 3C10400 |
| 12:58.55 | fugitivo | anyone using asterisk-stat ? |
| 13:06.21 | lbarth | mimmus: Im back.. sorry for the delay |
| 13:06.47 | Mimmus | lbarth: no problem, I'm still struggling with chan_capi compilation |
| 13:06.53 | lbarth | have you solved our porblem |
| 13:07.08 | lbarth | oh. you have to download the app_faxcapi patch |
| 13:07.28 | Mimmus | lbarth: /usr/lib/gcc/i586-mandrake-linux-gnu/3.4.1/include/stddef.h:213: error: |
| 13:07.28 | Mimmus | syntax error before "typedef" |
| 13:07.50 | lbarth | it is mentioned in the REDEM of the patch. Download it an just copy the app_fax*.c into the chan_capi folder |
| 13:07.55 | Mimmus | lbarth: another patch? |
| 13:07.58 | lbarth | jes |
| 13:08.04 | lbarth | sorry.. I forgott this |
| 13:08.16 | lbarth | mom... |
| 13:08.25 | lbarth | what version of chan_capi do you use? |
| 13:08.29 | Zeeek | as I was saying, before I was interrupted by a call, http://pastebin.ca/8529 shows an ZAP->IAX2 call that times out after being answered. Anyone comment? |
| 13:08.45 | langals | HI there.....the G.729 codec that comes with Asterisk - is that a free implementation? |
| 13:08.46 | Mimmus | chan-capi-0.3.5 |
| 13:08.56 | Zeeek | same call from IAX was ok. Same call in reverse (IAX call same ZAP) worked too |
| 13:09.06 | lbarth | mimmus: thats fine |
| 13:09.11 | Aze` | Can update Cisco 7912 to sip v.2.0 ? |
| 13:09.34 | clive- | langals * does not come with g729 |
| 13:09.41 | Mimmus | lbarth: do I need another patch? |
| 13:09.42 | Damin | Sip v2.0? |
| 13:09.56 | Damin | Aze`: I didn't know there was a 1.0. ;) |
| 13:10.02 | lbarth | mimmus: please read the readme that comes with the patch |
| 13:10.04 | langals | clive - but it is listed under "show audio codes" |
| 13:10.28 | *** join/#asterisk ClayReiche123 (fwuser@acxexch1.accxx.com) |
| 13:10.38 | lbarth | mimmus: http://www.mlkj.net/asterisk/chan_capi-0.3.5-patch.tar.bz2 |
| 13:10.48 | Aze` | Sip 2.x = RFC 3261 |
| 13:10.55 | clive- | yes, * can do g729, but you need to buy the codec from digium, or scale the illgal one |
| 13:11.16 | *** join/#asterisk rowter (~Drake@201.133.210.80) |
| 13:11.20 | lbarth | I've justed copied the missing .c file to the chan_capi source folder |
| 13:11.25 | lbarth | than just run make |
| 13:11.36 | rowter | am trying to compile openh323 support, am getting /usr/include/ptlib/semaphor.h:92: macro or `#include' recursion too deep |
| 13:12.16 | ClayReiche123 | OK, can someone PLEASE help me? I posed this question a couple days ago and noone could help... hoping someone new is here today. |
| 13:12.32 | Mimmus | lbarth: I read it, it says only run 'make install' after applying patch |
| 13:12.50 | jluk | so that's the question ClayReiche123 ? |
| 13:12.58 | jluk | s/that's/what's/ |
| 13:13.18 | lbarth | yes. thats wrong.. :-) but hte other patch is also mentioned. I've just compiled and installed it this morning |
| 13:13.44 | pif | hi, is there a way that certain parts of an extension be _not_ echo'ed at the * console? |
| 13:13.50 | *** join/#asterisk KristinH (~KristinH@muppet.geekgirls.us) |
| 13:14.05 | KristinH | good morning |
| 13:14.12 | Mimmus | lbarth: if you say me where I can find this patch, I'll try immediatley! |
| 13:14.31 | KristinH | anothere here familiar with using freeworlddialup via iax? |
| 13:15.02 | lbarth | mimmus: I've send you the patch |
| 13:15.04 | ClayReiche123 | I have configured a telephone adapter to use ONLY g711ULAW. I am using SIP and asterisk is connecting to a remote gateway for all PSTN calls. Asterisk is giving the remote gateway an option of G711ULAW AND g729, even though my device doesn't support it in this configuration. |
| 13:15.45 | Mimmus | lbarth: I'm not an IRC guru, what do I need to do to receive it? |
| 13:15.52 | jluk | ClayReiche123: so disallow=g729 |
| 13:16.09 | lbarth | ive send you the link; http://www.mlkj.net/asterisk/chan_capi-0.3.5-patch.tar.bz2 |
| 13:16.47 | ClayReiche123 | jluk: I've done that in the peer setup. Not in my global sip.conf settings because I want to allow g729 for most of my perrs. |
| 13:16.50 | *** join/#asterisk Dabba (~d@ipv6.mfnx.ip6net.net) |
| 13:16.58 | Mimmus | lbarth: sorry, I lost it in this river of words... |
| 13:17.16 | Zeeek | KristinH you have a problem? It works usually |
| 13:17.34 | lbarth | im not an irc guru as well but I've just tried to contact you in a "private" chat window |
| 13:17.38 | ClayReiche123 | jluk: I'm sorry... I do a "disallow=all" and then "allow=ulaw" |
| 13:17.51 | Zeeek | KristinH I just called 612 and it worked |
| 13:17.52 | KristinH | well it registers and i can use their "call me" webpage to ring my phone |
| 13:18.01 | langals | Clive: thanks. I have tried GSM, and the quality isn't great |
| 13:18.08 | KristinH | but if I try dialing 393 612 it doesnt work |
| 13:18.17 | KristinH | i get invalid extension |
| 13:18.20 | jluk | hmm ClayReiche123 that should work, and should mean that that peer could never establish a g729 |
| 13:18.45 | ClayReiche123 | jluk:for outgoing and incoming? |
| 13:18.45 | jluk | ClayReiche123: I think you might better off asking this when the yanks are awake. |
| 13:18.51 | fenlander | ClayReiche123: make sure you have disallow=all for the gateway - not just the phone |
| 13:19.37 | fenlander | CkayReiche123: but it should translate in that case... |
| 13:19.56 | Zeeek | KristinH when you dial, what is the exact Dial command? |
| 13:20.22 | ClayReiche123 | fenlander:The thing is, I can't restrict the gateway to g711 only. It is terminating ALL my PSTN calls and I want most of my users on g729. There are only a few that I want using g711.(for faxing) |
| 13:20.29 | *** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com) |
| 13:21.17 | KristinH | zeeek, exten => _393.,3,Dial(IAX2/${FWDNUMBER}:${FWDPASSWORD}@iax2.fwdnet.net/${EXTEN:3},60,r) |
| 13:21.22 | *** join/#asterisk UPM-eduardo (~UPMeduard@tauro2.dit.upm.es) |
| 13:21.58 | *** join/#asterisk shuric (alexander@alexander.office.inter-telecom.net.ru) |
| 13:22.10 | fenlander | CkayReiche123: sorry - misread |
| 13:22.17 | shadebob | hi, just a little question... Someone can explain what is the difference between a Voip server and a Voip proxy? |
| 13:22.27 | Zeeek | KristinH here is mine: Dial(IAX2/${FWDUSERID}@IAXpeerFWD/${EXTEN:3},45) |
| 13:22.45 | Zeeek | so you see I have a peer entry for FWDIAX |
| 13:22.50 | KristinH | basically the same |
| 13:23.11 | Zeeek | except mine works... |
| 13:23.27 | Zeeek | ulaw only you know? |
| 13:23.29 | *** join/#asterisk TheEmperor (TheEmperor@218.111.51.93) |
| 13:23.31 | bjohnson | shadebob: one answers .. the other just forwards |
| 13:23.43 | dreamcode | on what version does atxfer => *2 work ? |
| 13:23.58 | bjohnson | shadebob: in practice the difference is quite so clear cut |
| 13:24.08 | bjohnson | s/is/isn't |
| 13:24.15 | KristinH | I'm setup as a user with them |
| 13:24.28 | Zeeek | what does CLI say after the dial |
| 13:24.31 | Zeeek | ? |
| 13:24.40 | KristinH | PM ? |
| 13:24.49 | Zeeek | no I don't do pm |
| 13:24.55 | KristinH | ok |
| 13:25.01 | bjohnson | mine is just Dial(IAX2/IAXpeerFWD/${EXTEN:3},45) |
| 13:25.07 | Zeeek | do I look like that kind of guy? :) |
| 13:25.15 | bjohnson | I specify the username and secret in the iax peer |
| 13:25.23 | Zeeek | KristinH you did register for IAX, right? |
| 13:25.31 | KristinH | I followed the directions as shown on the site |
| 13:25.32 | bjohnson | ahh .. and wait |
| 13:25.35 | KristinH | yes i did |
| 13:25.37 | bjohnson | what site? |
| 13:25.44 | Zeeek | including asking for IAX on your account? |
| 13:25.44 | KristinH | fwd |
| 13:25.50 | bjohnson | url? |
| 13:26.21 | KristinH | hold on |
| 13:26.50 | KristinH | http://account.freeworlddialup.com/index_new.php?section_id=112 |
| 13:26.59 | shadebob | bjonhson : and what is advantage of a sip proxy... A server and distant sip phone make the same things? |
| 13:27.02 | bjohnson | KristinH: your dial command should work if fwd is expecting iax and you don't have any iax.conf entries conflicting with it |
| 13:27.39 | KristinH | i dont want to flood the channel so that is why i asked to pm |
| 13:27.40 | tzanger | shit man |
| 13:27.44 | tzanger | kram's coming to toronto? |
| 13:27.45 | Zeeek | bjohnson this is the best FWD link for now: http://www.freeworlddialup.com/content/view/sitemap/2 |
| 13:27.49 | bjohnson | shadebob: sometimes a sip proxy (like SER) is better for handling high volumes of traffic |
| 13:27.58 | tessier | tzanger: Hide the good silver! |
| 13:28.03 | tzanger | tessier: :-) |
| 13:28.15 | Zeeek | KristinH do you register with them? |
| 13:28.15 | tzanger | I was thinking I'll just give him my panasonic cordless phone and the NDA then |
| 13:28.28 | KristinH | I register |
| 13:28.43 | dwmw2 | +// Stolen from chan_zap |
| 13:28.43 | dwmw2 | +static void capi_handle_dtmf_fax(struct ast_channel *ast) |
| 13:28.44 | dwmw2 | hmmm |
| 13:28.45 | Zeeek | and you say you followed http://www.freeworlddialup.com/advanced/iax |
| 13:28.54 | dwmw2 | that's generic code and really doesn't want reimplementing in every channel driver |
| 13:29.00 | KristinH | i followed those instructions on that url |
| 13:29.13 | Zeeek | weird because that works for the rest of us |
| 13:29.15 | bjohnson | KristinH: I meant what url are you following for setting up iax.conf |
| 13:29.20 | Zeeek | ah wait... |
| 13:29.30 | Zeeek | bandwidth=low in iax.conf??? |
| 13:29.40 | KristinH | = high |
| 13:29.43 | bjohnson | I assume you mean http://www.freeworlddialup.com/advanced/iax |
| 13:29.49 | KristinH | bjohnson, yes |
| 13:30.01 | bjohnson | try show peer |
| 13:30.03 | Zeeek | ok I'm out of ideas unless their IAX auth system is down for new accounts |
| 13:30.06 | bjohnson | iax2 show peer |
| 13:30.11 | fenlander | ClayReiche123: you could probably use ${SIP_CODEC} to set the codec? |
| 13:30.54 | shadebob | bjonhson : sometimes ... maybe I can use proxy to generate local cdr stats |
| 13:31.40 | KristinH | I'm not a peer, I am a user |
| 13:31.57 | bjohnson | shadebob: most people just use * |
| 13:31.59 | Zeeek | you should see fwd in the peer list though if you register |
| 13:32.15 | bjohnson | KristinH: user is for incoming .. not for dial commands |
| 13:32.27 | KristinH | dont you mean iax2 show registry? |
| 13:33.00 | *** join/#asterisk VirTERM (~virterm@shiva.kanatek.com) |
| 13:33.09 | Zeeek | KristinH my iax2 show peers shows: IAXpeerFWD/2,,,, |
| 13:33.16 | KristinH | 65.39.205.121:4569 636192 192.168.0.100:60564 60 Registered |
| 13:33.19 | Zeeek | you need a peer entry to dial |
| 13:33.30 | facek_ | Zeeek did you do what wa taling about? |
| 13:33.33 | Zeeek | not registry (tho that's comforting to see) |
| 13:33.47 | Zeeek | facek_ yes the macro parameter works fine |
| 13:34.10 | facek_ | Zeeek yes. but how to run that from .call file.. where i don't run Dial.. co I cant use a M(macro} |
| 13:34.11 | Zeeek | it's good because that means you can call a hotel and play "Room 100" when they answer |
| 13:34.30 | KristinH | well it appears their webpage is missing the peer part :p |
| 13:34.34 | Zeeek | facek_ how do you connect to the cellphoen if you don't dial? |
| 13:34.34 | facek_ | Zeeek example? |
| 13:34.50 | facek_ | Zeeek i put file iv /var/spool/asterisk/outgoing/ |
| 13:35.39 | Zeeek | KristinH you're right! I never noticed |
| 13:36.01 | KristinH | care to share your peer? |
| 13:36.06 | KristinH | config |
| 13:36.11 | darkskiez | anyone built the latest cvs head? where does the uncompress function pbx_dundi calls come from? |
| 13:36.31 | Zeeek | KristinH sure just a sec |
| 13:36.35 | KristinH | thank you |
| 13:36.48 | KristinH | i've been pulling my hair long enough over this :p |
| 13:36.52 | facek_ | Zeeek ? |
| 13:37.12 | Zeeek | KristinH here: http://pastebin.ca/8530 |
| 13:37.37 | Zeeek | someone should write to them about that |
| 13:38.10 | Zeeek | facek_ what is the problem you have exactly? I told you to try the M parameter with the dial. NOw you say you don't dial. How do you connect? |
| 13:38.17 | *** join/#asterisk MikeJ[Laptop] (~icechat5@65.170.43.34) |
| 13:38.30 | facek_ | Zeeek using the .call files |
| 13:38.35 | facek_ | Zeeek can you show me example |
| 13:38.46 | Zeeek | the call file will have to dial the phone in its context |
| 13:39.14 | Zeeek | what does the context look like? go to pastebin.ca and post the .call file and the context in it |
| 13:39.28 | LeoB | novice question: could not register my softphone with Asterisk. can anyone help? |
| 13:39.38 | Zeeek | LeoB more info |
| 13:39.55 | Zeeek | OS, NAT, what phone... |
| 13:40.17 | bjohnson | KristinH: the way they show their dial example .. you don't need a peer entry. But I could never get it to work that way. Also .. leaving your peer entries and secrets in iax.conf simplifies your extensions.conf .. more so when you have multiple voip accounts |
| 13:40.30 | LeoB | windows, no nat, sjphone and xlite |
| 13:40.35 | shadebob | bjonhson : thanks for informations |
| 13:41.11 | bjohnson | LeoB: I don't use windows so dont know those programs .. but have you checked the wiki for info? |
| 13:41.13 | Zeeek | bjohnson true and true |
| 13:41.18 | KristinH | so then do I just need that one entry in iax.conf now? |
| 13:41.39 | LeoB | I did, but somehow it didn't work |
| 13:41.40 | Zeeek | LeoB you need to show what you see on the CLI when you try to call |
| 13:41.41 | bjohnson | KristinH: I suggest 2 entries in iax.conf |
| 13:41.41 | *** join/#asterisk abatista (~Ariel@dsl-20-177.cofs.net) |
| 13:42.03 | KristinH | both with the same context? |
| 13:42.14 | bjohnson | KristinH: one user for incoming calls and one peer for outgoing. Gives you more control over authentication, cdr accounts, etc, etc |
| 13:42.31 | Zeeek | does context matter for outgoing? I think not |
| 13:42.32 | bjohnson | KristinH: I find type=friend is useful for internal extensions but nothing else |
| 13:42.43 | KristinH | k |
| 13:42.47 | bjohnson | context= is only for incoming |
| 13:42.54 | bjohnson | eg .. user |
| 13:42.56 | ManxPower | And many service providers use differnet username/secret for incoming .vs. outgoing and so you HAVE to use user/peer types. |
| 13:43.06 | tzanger | bjohnson: mark's coming to toronto on the 21st |
| 13:43.07 | bjohnson | fwd is actually one |
| 13:43.07 | facek_ | Zeeek this is my .call file with my telephon numer http://pastebin.ca/8532 |
| 13:43.11 | Zeeek | ManxPower I havea IAX question for you |
| 13:43.13 | bjohnson | they use rsa keys for incoming |
| 13:43.20 | *** part/#asterisk langals (~icechat5@196.7.14.183) |
| 13:43.41 | facek_ | Zeeek adn this is extension http://pastebin.ca/8533 |
| 13:43.44 | bjohnson | tzanger: yeah .. I read about that a few weeks ago I think. Asticon in TO or something? |
| 13:44.01 | bjohnson | LeoB: was there also info about setting up sip.conf? |
| 13:44.05 | Zeeek | I called an IAX softphone today and the person answered, but the Dial command timed out and went to email after the 15 seconds (dial(IAX2/user,15) |
| 13:44.07 | tzanger | bjohnson: this is the first I've heard of it :-) |
| 13:44.19 | ManxPower | Zeeek: no idea. |
| 13:44.22 | LeoB | bjohnson, yes |
| 13:44.22 | sudhir492 | I get the following error: Ooh, voice format changed to 256 |
| 13:44.22 | sudhir492 | Asked to transmit frame type 256, while native formats is 1 (read/write = 1/1) |
| 13:44.53 | ManxPower | sudhir492: You've been here long enough to know what that message means and how to fix it. |
| 13:44.56 | Zeeek | ManxPower this happened only with ZAP->IAX, nowhere else |
| 13:45.05 | bjohnson | LeoB: give us the url to what you were following to make sure it looks right |
| 13:45.06 | Zeeek | I've never seen anything like that |
| 13:45.21 | VirTERM | morning |
| 13:45.23 | bjohnson | LeoB: just paste the url here .. don't dcc me the page |
| 13:45.48 | VirTERM | anyone using realtime with iax trunks/peers? |
| 13:45.53 | LeoB | bjohnson: which url? |
| 13:45.54 | sudhir492 | yes, g723.1 but the provider insists that they are using g729. |
| 13:45.57 | bjohnson | LeoB: also .. can you please confirm to us that you aren't trying to go through a router |
| 13:46.02 | Zeeek | facek_ I asked for the .call file with the extension |
| 13:46.05 | bjohnson | LeoB: the one you were following |
| 13:46.08 | LeoB | no router |
| 13:46.09 | Faithful | zaphfc works will guys!!! |
| 13:46.09 | ManxPower | sudhir492: they lie. But just force it on your side. |
| 13:46.16 | sudhir492 | ManxPower: I think I wll trust * |
| 13:46.18 | facek_ | Zeeek i gave you.. but sorry for in two other files |
| 13:46.40 | Zeeek | facek_ ok I didn't see it above |
| 13:46.42 | Zeeek | I got it |
| 13:46.58 | ManxPower | sudhir492: don't use a bandwidth= line, use disallow=all allow=g729 in the peer and user for that service provider. |
| 13:47.00 | LeoB | bjohnson, here's what I get from CLI: Mar 31 08:42:21 NOTICE[3880]: chan_sip.c:8428 handle_request: Registration from '<sip:softphone-usr@192.168.2.15:5060>' failed for '192.168.2.11' |
| 13:47.40 | KristinH | gentlemen thank you. it is now working |
| 13:47.48 | Zeeek | cool! |
| 13:48.01 | KristinH | time for a shower and then get ready for work |
| 13:48.03 | bjohnson | LeoB: authentication problem .. make sure the user and secret in the softphone match the user and sercret in the type=user or type=friend section of sip.conf |
| 13:48.04 | Zeeek | KristinH what are you doing later tonight? :) |
| 13:48.09 | KristinH | i really appreciate the help |
| 13:48.12 | Zeeek | dinner? |
| 13:48.16 | KristinH | heh |
| 13:48.20 | Zeeek | in Paris? :) |
| 13:48.31 | bjohnson | Zeeek: yeah .. you eat at your house and I'll eat at mine |
| 13:48.31 | Druken | mmmmmm, mcdonalds breakfast... |
| 13:48.42 | KristinH | that sounds nice but i dont think my SO would appreciate me leaving the country |
| 13:48.44 | Zeeek | bjohnson I wouldn't tougch that one |
| 13:49.03 | bjohnson | SO? |
| 13:49.09 | KristinH | significant other |
| 13:49.10 | bjohnson | Senior Officer? |
| 13:49.13 | Zeeek | KristinH no more than mine would appreciate me dining when she gets home at 10PM |
| 13:49.19 | KristinH | heh |
| 13:49.23 | facek_ | Zeeek and, you know what i am doing wrong? |
| 13:49.24 | Zeeek | anyway cheers |
| 13:49.47 | Zeeek | facek_ in short, you probably want to look up the Local pseudo channel on the wiki |
| 13:49.59 | Makenshi | how can i get asterisk to reload musiconhold.conf without restarting it? |
| 13:50.02 | Druken | Zeeek: you mean you have dinner before 10pm ? |
| 13:50.04 | facek_ | Zeeek ok |
| 13:50.07 | KristinH | mine is a football player with not much of a sense of humor regarding dinner with strangers |
| 13:50.16 | Zeeek | facek_ If you write Channel: Local/9000 for example |
| 13:50.27 | Druken | KristinH: soccer? :) |
| 13:50.34 | KristinH | no football |
| 13:50.35 | Zeeek | and then put a dial comand at extension 9000 |
| 13:50.40 | KristinH | the real football :) |
| 13:50.47 | facek_ | Zeeek so i must create that pseudo channel.. and in extension run Dial.. and run macro.. so i need one extension and one macro, right? |
| 13:50.48 | KristinH | youknow, NFL |
| 13:50.52 | Druken | as in the pigs skin ? |
| 13:50.54 | Zeeek | and use the famous M parameter, you will be rewarded with success |
| 13:50.55 | Druken | ahh, excelent |
| 13:51.04 | Zeeek | I hate football more than any other sport |
| 13:51.17 | KristinH | well i gtg. |
| 13:51.20 | KristinH | thank you again! |
| 13:51.21 | Zeeek | facek_ that's it - I think that is your answer |
| 13:51.22 | Druken | ya never know with the damn euro's in here, calling soccer football |
| 13:51.28 | *** part/#asterisk KristinH (~KristinH@muppet.geekgirls.us) |
| 13:51.51 | *** join/#asterisk subtract (~subtract@ottawa-hs-209-217-110-148.d-ip.magma.ca) |
| 13:52.50 | VirTERM | Realtime; anyone? |
| 13:53.01 | Druken | realtime == fun :) |
| 13:53.10 | VirTERM | I know.... |
| 13:53.23 | VirTERM | drunen: have you got it working with iax trunks? |
| 13:53.33 | Druken | nope.. never tried |
| 13:53.43 | subtract | hello does anybody use asterisk on an OpenBSD platform? if yes could you recommend a basic SIP softphone that works well on OPENBSD. |
| 13:54.14 | *** join/#asterisk Supaplex (supaplex@205.208.245.134) |
| 13:54.32 | VirTERM | realtime => headache |
| 13:54.46 | Druken | subtract: well, what does it matter is asterisk is on openbsd? your looking for a softphone... |
| 13:54.59 | subtract | yes |
| 13:55.02 | Zeeek | which for any unix isn't easy |
| 13:55.03 | Druken | VirTERM: i guess... realtime is a bit buggy |
| 13:55.34 | VirTERM | well, I know that, but still need to get it to work |
| 13:57.02 | subtract | is it possible to run Asterisk from command line (place calls, receive calls) |
| 13:57.20 | subtract | without needing a softphone |
| 13:57.46 | subtract | i've tried linphone |
| 13:57.56 | Druken | asterisk is the server... |
| 13:58.10 | subtract | yes |
| 13:58.31 | Druken | you've sorta lost me... |
| 13:58.35 | *** join/#asterisk markak2 (~twist@ndn-165-150-215.telkomadsl.co.za) |
| 13:58.48 | markak2 | hi all. |
| 13:58.55 | subtract | I need a softphone for OpenBSD |
| 14:00.36 | dwmw2 | subtract: yes, it's possible if your kernel supports OSS -- which some BSDs do, right? |
| 14:00.45 | markak2 | does anyone know how one would aproach this. lets says the board has 2 pstn lines on an fxo card. and 2 people are currently on calls then a thrid person wants to make a call. at present the phones just hangup because asterisk sees no way to place the call. could you have that person parked and then asterisk can dial the number on the next open port. |
| 14:01.35 | markak2 | at present asterisk plays music while a call is being connected and connects once the call is set up. basicaly keep on playing the music until line is available |
| 14:01.42 | facek_ | anyone know that problem |
| 14:01.43 | facek_ | <PROTECTED> |
| 14:01.49 | ManxPower | markak2: That is a non-trivial thing, but I suspect you can do it. |
| 14:02.16 | markak2 | i know but would be incredibly useful |
| 14:02.55 | facek_ | Zeeek llok at that http://pastebin.ca/ i think is good.. but this start playing before connect |
| 14:02.58 | markak2 | zeeek are you there ? |
| 14:03.14 | Zeeek | yes I am in fact |
| 14:03.22 | facek_ | Zeeek http://pastebin.ca/8534 |
| 14:03.28 | markak2 | do you have any ideas on this |
| 14:03.47 | Druken | markak2: i would personally get an outbound host for 20 bux a month and shunt all my calls out that, rather then using my fxo lines.. :) |
| 14:04.10 | Druken | that way you can have people calling out, and still have two incoming lines |
| 14:04.17 | ManxPower | Oh, that sucks. Jerry Falwell is feeling better. |
| 14:04.35 | Druken | damnit.. i knew i should have used more arsenic |
| 14:04.38 | markak2 | im in south africa no termination points in this country yet. all route via london |
| 14:04.40 | Zeeek | markak2 it's an intriguing question I admit :) |
| 14:05.24 | Druken | markak2: the only way i could see that happening then, would be with a looped dialplan... with a few gotoif's in it... |
| 14:05.32 | tzanger | markak2: I am guessing the channel disappeared before echo cancel could get turned on |
| 14:05.37 | Zeeek | facek_ remove the goto s,1 |
| 14:05.41 | ManxPower | Druken: Should have used more. He's a lot heaver than Strom Thurmand 8-) |
| 14:06.09 | Druken | ManxPower: :) |
| 14:06.12 | Zeeek | facek_ then yoiu're missing the logic to get the number and make the call |
| 14:06.23 | Druken | markak2: use the exten => _1800NXXXXXX,1,ChanIsAvail(Zap/g1) |
| 14:06.54 | *** join/#asterisk klictel (~klictel@207.107.208.137) |
| 14:06.54 | Druken | type of thing to check if there's an available channel, if not, waitmusiconhold(2) then try again |
| 14:06.55 | markak2 | druken : and then put some timeout and try again until open ? |
| 14:06.59 | klictel | morning all |
| 14:07.48 | Druken | only thing is.. the moh might be a little choppy cause of the looping... |
| 14:07.58 | facek_ | hm |
| 14:08.23 | *** part/#asterisk _Sam-- (sam@ns2.kneedraggers.com) |
| 14:08.30 | facek_ | Zeeek macro is working as extensions.. i can make the aplayback.. press one to reply and |
| 14:08.34 | markak2 | ill use a small file that repeats please be patient |
| 14:08.41 | facek_ | exten => 1,1,Goto(s,1) |
| 14:11.10 | facek_ | Zeeek this is only mechanism to inform me about some specific events on server |
| 14:11.23 | tzanger | eh? |
| 14:13.11 | facek_ | yh? |
| 14:13.30 | Zeeek | facek_ perhaps WaitExten |
| 14:14.03 | Zeeek | or Background(please-enter) |
| 14:14.19 | facek_ | Zeeek yes but in macro-play yes? |
| 14:14.33 | Zeeek | instead of playback |
| 14:14.45 | Zeeek | not sure about the return part |
| 14:14.54 | Zeeek | use-w |
| 14:15.12 | ariel_ | morning everyone |
| 14:15.32 | facek_ | Zeeek ok |
| 14:15.38 | facek_ | ariel_ hi |
| 14:15.56 | Zeeek | I have to go now, bye all |
| 14:16.03 | facek_ | where to put specified file for language.. for saydigit and others |
| 14:16.06 | *** part/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
| 14:17.10 | Mimmus | Can someone tellme where can I can find /usr/include/asterisk/channel_pvt.h file? |
| 14:17.30 | Mimmus | It is not in current CVS and chan_capi doesn't compile |
| 14:18.36 | ariel_ | facek_, /var/lib/asterisk/sounds/ is the location of all to the sound files. |
| 14:19.03 | VirTERM | Realtime; anyone? |
| 14:20.05 | antifuchs | VirTERM: unless you state a slightly more specific question, the answer is likely to remain "" |
| 14:20.29 | VirTERM | :) realtime for iax peers/trunks |
| 14:20.44 | VirTERM | I am having problem getting it to work |
| 14:21.02 | VirTERM | looking for WORKING table structure |
| 14:22.36 | VirTERM | for iax stuff |
| 14:24.35 | *** join/#asterisk mentat (~Mentat@pcp01260498pcs.nhaven01.ct.comcast.net) |
| 14:24.48 | *** join/#asterisk julianjm (~julianjm@250.Red-80-59-67.pooles.rima-tde.net) |
| 14:27.54 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
| 14:28.34 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 14:28.34 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:29.27 | *** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
| 14:29.27 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 14:32.36 | *** join/#asterisk SpaceBass (~sp@c-24-125-33-214.hsd1.va.comcast.net) |
| 14:32.38 | SpaceBass | morning |
| 14:33.04 | epoch | sigh |
| 14:33.05 | SpaceBass | i have a sip phone behind a firewall that I cannot get to register... all ports are open and forwarded to my asterisk box |
| 14:33.20 | epoch | I tried to unsubscribe from the asterisk-users mailing list yesterday |
| 14:33.23 | epoch | got a confirmation |
| 14:33.30 | epoch | and am still receiving messages... |
| 14:34.02 | jaiger | epoch, sounds like lag in the mail queue |
| 14:37.24 | epoch | yeah, I geuss |
| 14:37.25 | epoch | "guess" |
| 14:37.41 | epoch | it's all good just as long as it doesn't persist past tomorrow :) |
| 14:37.57 | epoch | I think that's when my e-mail address is gonna get disabled |
| 14:38.00 | epoch | or rerouted, or something |
| 14:38.44 | ManxPower | SpaceBass: is your Asterisk box behind nat? |
| 14:39.35 | SpaceBass | ManxPower yeah |
| 14:39.53 | SpaceBass | (its actually two internal networks with nat b/t them) |
| 14:40.15 | ManxPower | SpaceBass: your configuration is beyond even my help |
| 14:40.28 | SpaceBass | lol |
| 14:40.31 | SpaceBass | well thanks for offering |
| 14:40.48 | tzanger | http://www.bce.ca/en/news/releases/bc/2005/03/30/72235.html |
| 14:45.06 | *** join/#asterisk Dabba (~d@ipv6.mfnx.ip6net.net) |
| 14:45.36 | *** join/#asterisk JerJer[mobile] (~nonyobizn@65.173.197.174) |
| 14:46.36 | dwmw2 | two internal network with nat between them |
| 14:46.40 | dwmw2 | IPv6 baby :) |
| 14:47.24 | SpaceBass | dwmw2 its for a wifi sip phone that doesnt support WPA, so i dont want a WEP only access point on my network |
| 14:47.31 | SpaceBass | and IPv6 isn't an option |
| 14:48.27 | dwmw2 | so the WEP access point is behind a firewall? |
| 14:48.34 | SpaceBass | dwmw2 yeah |
| 14:48.54 | dwmw2 | hm. You get to fight with NAT then. Or perhaps install Asterisk on the firewall too :) |
| 14:49.24 | SpaceBass | dwmw2 well, it does seem to be a nat issue... but I have litterally forwared all ports |
| 14:49.30 | ClayReiche123 | fenlander: you still here? |
| 14:49.36 | SpaceBass | i have net=yes in my config |
| 14:49.52 | dwmw2 | if there's only one device on it and you're forwarding all ports -- why not just give it an IP address, and firewall it hard _without_ NAT. |
| 14:50.15 | dwmw2 | let it talk _only_ to the asterisk server, on certain ports |
| 14:50.19 | dwmw2 | but let it have a real IP address |
| 14:50.28 | pif | hi, I'm trying to set the ALERT_INFO on * to change the ring type of a Polycom |
| 14:50.43 | markak2 | Druken : i have it working perfectly waits for a channel and connects when available after playing a small message do you know if one can play music during a wait() period |
| 14:50.51 | pif | using: exten => s,2,SetVar(ALERT_INFO=<CUSTOM_1>) |
| 14:50.58 | pif | is that correct? |
| 14:51.22 | *** join/#asterisk cbachman (~chatzilla@victory.ece.northwestern.edu) |
| 14:52.02 | SpaceBass | dwmw2 probably an option, but beyond my expertese to set it up |
| 14:52.07 | *** join/#asterisk SuPrSluG (~SuPrSluG@pool-129-44-136-89.buff.east.verizon.net) |
| 14:52.30 | *** join/#asterisk Moc____ (~mochouina@64.235.210.66) |
| 14:53.02 | SpaceBass | option B is to trow another nic in the * box and put the AP right on that |
| 14:53.24 | SpaceBass | b/c then I am really only compermising my * box |
| 14:54.34 | *** part/#asterisk ClayReiche123 (fwuser@acxexch1.accxx.com) |
| 14:55.48 | *** join/#asterisk scorpion68 (~chatzilla@HSE-Toronto-ppp186743.sympatico.ca) |
| 14:56.31 | *** join/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
| 14:57.20 | Moc____ | morning |
| 14:57.36 | *** join/#asterisk wazquis (~akv@lnxbx.dk) |
| 14:57.37 | bjohnson | or do like dwmw2 suggested and make another * server and use iax between the 2 * servers |
| 14:58.04 | wazquis | is it possible to dial from asterisk console? maybe even from a bash? |
| 14:58.13 | *** join/#asterisk zotz (~zotz@24.231.32.191) |
| 14:58.25 | *** part/#asterisk fedor (~fbond@office.tura.ru) |
| 14:58.33 | SpaceBass | bjohnson well, I might do that... but for a home "project" expirement this is already getting out of control |
| 14:59.21 | *** join/#asterisk Egonis (~chultay@69.194.211.129) |
| 14:59.21 | Egonis | I am trying to emerge asterisk, but get many error messages in zaptel -- I am using udev, not devfs |
| 14:59.21 | Egonis | and zaptel fails to emerge |
| 15:02.22 | ManxPower | IRQ Misses: 1502 |
| 15:02.26 | markak2 | does anybody know if you can play music in the background during a wait(10) |
| 15:02.32 | ManxPower | I think the system is a tad ill. |
| 15:02.57 | *** join/#asterisk Blissex (~Blissex@82-69-39-138.dsl.in-addr.zen.co.uk) |
| 15:03.05 | Egonis | ManxPower: Not that I know what I am talking about, but are you using IRQ Balancing in 2.6? |
| 15:03.18 | ManxPower | Egonis: Hell no! |
| 15:03.25 | Egonis | ManxPower: LOL, why not? |
| 15:03.30 | ManxPower | Egonis: I use 2.4 |
| 15:03.37 | *** join/#asterisk viLeR (1000@ip-47-252.telesat.com.co) |
| 15:03.42 | Egonis | ManxPower: Is the fact that I use kernel 2.6 an issue? |
| 15:03.50 | Egonis | ManxPower: maybe that's my problem |
| 15:03.59 | ManxPower | An IRQ miss for a Zap card is kind of like if suddently gravity was reversed for 1 second. |
| 15:04.05 | ManxPower | Egonis: 2.6 works fine. |
| 15:04.12 | drumkilla | ManxPower: in cvs head, there is a StartMusicOnHold and StopMusicOnHold app that would work for that |
| 15:04.13 | Egonis | ManxPower: ahh, okay |
| 15:04.20 | ManxPower | Egonis: But packaged versions of asterisk are frequently screwed up |
| 15:04.42 | ManxPower | drumkilla: in cvs stable has tnose applications too |
| 15:04.42 | Egonis | ManxPower: I am using Gentoo, but can't get zaptel to emerge correctly, although I am using udev |
| 15:05.05 | drumkilla | ManxPower: ... really? I sure didn't know they were there :) |
| 15:05.09 | ManxPower | Egonis: does it work if you download zaptel and build it yourself? |
| 15:05.42 | Egonis | ManxPower: That's what gentoo does, is build it from source... and no, it doesn't help.... I get various 'disabling devfs' messages, since I don't use it -- then it fails |
| 15:05.55 | Egonis | ManxPower: Zaptel 0.9.1 |
| 15:06.16 | blitzrage | drumkilla: !!!! |
| 15:06.17 | ManxPower | Egonis: That is so old don't expect it to work. |
| 15:06.24 | ManxPower | Egonis: the current version is 1.0.7 |
| 15:06.31 | Egonis | ManxPower: I will try that one then |
| 15:06.49 | blitzrage | what should an "Asterisker" be called? |
| 15:06.57 | Pj386 | an "asterisker" ? :) |
| 15:07.01 | ManxPower | blitzrage: Ast Master, of course. |
| 15:07.02 | drumkilla | hey blitzrage :) |
| 15:07.17 | drumkilla | Asterisk user? I mean, I know that's boring, but ... |
| 15:07.36 | drumkilla | Astriholic? ... you'd have to credit twisted on that one |
| 15:07.47 | blitzrage | drumkilla: well, I have the words "young Jedi" in documentation... but I want to replace that with "young ......" |
| 15:07.50 | Pj386 | AsteRisker ? |
| 15:07.57 | blitzrage | Pj386: like that one :) |
| 15:08.13 | Pj386 | Yeah, experience inspired me :) |
| 15:08.22 | *** part/#asterisk wazquis (~akv@lnxbx.dk) |
| 15:08.23 | blitzrage | don't we know it |
| 15:08.33 | ManxPower | blitzrage: I like "grasshopper" |
| 15:09.03 | blitzrage | I don't get that term... I mean... you would think it should be some animal that changes when it grows up... |
| 15:09.10 | Egonis | which version of asterisk should I use? Gentoo suggests 0.9.0 |
| 15:09.19 | blitzrage | Egonis: CVS |
| 15:09.31 | Pj386 | Egonis: forget gentoo |
| 15:09.31 | blitzrage | Egonis: or Stable 1.0.7 |
| 15:09.39 | Egonis | blitzrage: Cool, will try |
| 15:09.42 | blitzrage | Egonis: yah... don't use the Asterisk in portage |
| 15:09.43 | Egonis | Pj386: Why? |
| 15:09.50 | Pj386 | I'm using gentoo too, and _love_ it, but you should just forget about emerge for asterisk |
| 15:09.50 | blitzrage | Egonis: it's OOOOOOOOOOOOOOOOOOOOOOOLD |
| 15:09.52 | dwmw2 | SpaceBass: setting up the box which routes to the WEP network to _route_ instead of doing NAT really won't be hard |
| 15:09.59 | Egonis | blitzrage: hah, okay |
| 15:10.06 | Pj386 | 'cause it doesn't work with it, and it works without |
| 15:10.43 | Egonis | Pj386, blitzrage: Trying an emerge of 1.0.7 first, if it fails, I will do it the old fashioned way |
| 15:10.50 | Pj386 | If you're motivated to find out how to make a clean ebuild, then please do so, but right now it didn't work for me, didn't work for other, and, well, obviously not for you either :) |
| 15:11.45 | blitzrage | Egonis: you're going to have to learn how to build Asterisk at some point. Its not hard, so don't use emerge as a crutch. |
| 15:11.58 | *** join/#asterisk loud (~ariel@null0.flapping.net) |
| 15:12.08 | Egonis | blitzrage: Okay, I understand... it's going to be a learning experience anyway I suppose.. :) |
| 15:12.27 | blitzrage | Egonis: its really not that hard, and the docs on asteriskdocs.org should step you through it. |
| 15:12.40 | Egonis | blitzrage: cool, will check that out |
| 15:12.51 | blitzrage | export CVSROOT=:pserver:anoncvs@cvs.digium.com:/usr/cvsroot |
| 15:12.53 | blitzrage | cvs login |
| 15:12.57 | blitzrage | (pass is anoncvs) |
| 15:13.02 | blitzrage | cvs co asterisk |
| 15:13.07 | Egonis | blitzrage: FYI, the 1.0.7 ebuild worked |
| 15:13.18 | blitzrage | Egonis: don't care, still a crutch :) |
| 15:13.38 | Egonis | blitzrage: LOL |
| 15:14.02 | Mimmus | Starting CVS Asterisk, I get this error: |
| 15:14.03 | Mimmus | WARNING[11319]: loader.c:388 __load_resource: cdr_custom.so: load_module failed, returning -1 |
| 15:14.03 | Mimmus | <PROTECTED> |
| 15:14.04 | Mimmus | Mar 31 17:09:36 WARNING[11319]: loader.c:509 load_modules: Loading module cdr_custom.so failed! |
| 15:14.06 | Egonis | What kind of IP Phone should I use w/ Asterisk? (Physically) H.323, ...? |
| 15:14.52 | *** part/#asterisk loud (~ariel@null0.flapping.net) |
| 15:14.52 | *** join/#asterisk gtigene (~gnadenx@c-67-184-112-58.hsd1.il.comcast.net) |
| 15:14.52 | *** join/#asterisk eKo1 (~bernd@63.245.57.70) |
| 15:14.52 | pif | is iaxtel still on? |
| 15:14.52 | blitzrage | Egonis: I like 7960's, heard good things about Polycom's. You can use any phone you with if you have a TDM400P with FXS interfaces. |
| 15:14.59 | *** join/#asterisk loud (~ariel@route.flapping.net) |
| 15:15.05 | blitzrage | pif: I've not had luck with IAXtel for over a year, and finally just removed it a few months ago |
| 15:15.08 | Blissex | Egonis: ideally a IAX2 phone, but you can also use an FXS box... |
| 15:15.34 | eKo1 | Is it just me or is the SPA-2000 the shittiest SPA ever. |
| 15:16.12 | gtigene | Where can I read about whats in /var/log/asterisk/messages |
| 15:16.38 | pif | blitzrage: what should we use now? |
| 15:17.17 | facek_ | if i am setting variable in .call fiel SetVar: cos=123 |
| 15:17.20 | Egonis | Blissex, blitzrage: My supplier has SIP, H.323, and MGCP IP |
| 15:17.21 | facek_ | i will see that in macro? |
| 15:18.01 | VirTERM | anyone using realtime with iax trunks/peers? |
| 15:18.06 | Blissex | Egonis: H323 is really rather complicated. There is not a lot of stuff that does MGCP, and lots of stuff with SIP. |
| 15:18.21 | *** join/#asterisk chrislwade (~clwade@river104.bigriver.net) |
| 15:18.48 | mentat | eKo1: that doesn't sound good, I was thinking about getting one |
| 15:18.51 | Egonis | Blissex: So SIP would definitely be the easiest, right? -- Dumb question, I have Asterisk installed, is it simple to configure? i.e. how quickly can I get the SIP Phone talking on the network? |
| 15:18.52 | Blissex | Egonis: in practice the ideal configuration is to run Asterisk locally, use IAX2 to connect to a IAX2 gateway provider and use a SIP phone locally to connect to Asterisk. |
| 15:19.38 | bkw_ | ITS OVER WITH finally |
| 15:19.38 | Blissex | Egonis: problem is, Asterisk is a bitch to configure, for now. I am writing a ''generic'' sample config and config docs that I think make it not quite as obnoxious as it is now. |
| 15:19.59 | gtigene | In /var/log/asterisk/messages we get "NOTICE Unable to create channel of type Zap". Does this mean I am dropping calls, I wonder? Have searched for documentation of log file, not found it yet. |
| 15:20.34 | Blissex | Egonis: the reason why ideally you want to use IAX2 to connect to an external service provider is that it uses rather less bandwidth and it passes thru NAT a lot more easily. |
| 15:20.37 | Egonis | Blissex: And what constitutes an IAX2 gateway? I have 2 IP Phones with a carrier (Primus Canada) |
| 15:20.59 | Egonis | Blissex: Going into a D-Link box |
| 15:21.04 | Blissex | Egonis: but locally on a LAN the bandwidth and NAT are not an issues, and it is much easier to find SIP phones. |
| 15:21.56 | *** join/#asterisk pupfuzz (~pupfuzz@pcp01687453pcs.wchstr01.pa.comcast.net) |
| 15:23.24 | *** join/#asterisk Milligan (~stephen@wkstn7.gnwd-noc.valuelinx.net) |
| 15:23.44 | Milligan | Hey all.... |
| 15:24.13 | Aze` | Anyone use IVCALL ? |
| 15:25.12 | Zeeek | bkw_ This behavior "is attributable to brain stem and forebrain functions that are reflexive, rather than cognitive"? |
| 15:25.40 | facek_ | Zeeek i am setting var in .call file liek that SetVar: wolne=20 |
| 15:25.51 | Zeeek | that's nice |
| 15:25.53 | facek_ | but i donyt see that when run NoOp(${wolne}) do you know why? |
| 15:26.13 | Zeeek | I don't. Unless yiou can set variables in a .call file? |
| 15:26.28 | facek_ | i set yes |
| 15:26.33 | facek_ | SetVar: wolne=20 |
| 15:27.09 | blitzrage | bkw_: when it's over, it's really just begun |
| 15:27.16 | *** join/#asterisk scrubb (~scrubb@OCI-19-41.onecall.net) |
| 15:27.24 | *** join/#asterisk mhnoyes (~mhnoyes@user-2ivfin7.dialup.mindspring.com) |
| 15:28.47 | *** part/#asterisk kpfleming (~kpfleming@207.111.174.1) |
| 15:28.47 | pupfuzz | Hi folks, can someone explain to me how to log the the information necessary to identify a problem with SIP registration? Everything has been working fine until recently - something changed in CVS HEAD since 3/27/05 which causes outages in my SIP phones (asterisk log says Registration from '<sip:1136@10.10.10.19>' failed). |
| 15:30.59 | bjohnson | canucks - who is interested in getting a HP 2550L Colour Laser printer for $310 ? |
| 15:31.09 | ManxPower | pupfuzz: You didn't see anything on the asterisk-cvs mailing list that might help? |
| 15:31.24 | blitzrage | pupfuzz: if you know when the change went into CVS, you should check your CVS logs (if you're running HEAD, you're obviously subscribed to the CVS mailing list) and check for changes that may have caused the problem |
| 15:31.26 | ManxPower | Does anyone here know how to use patlooptest? |
| 15:31.29 | ChkDigit | bjohnson: Ok, I'm interested |
| 15:31.36 | scrubb | ok, I'm having trouble getting the definitive answer on this. Will the TE410P work in a PCI-X port? |
| 15:32.06 | scrubb | It looks like from digiums page it should, but digium never comes out and says PCI-X. |
| 15:32.16 | bjohnson | ChkDigit: look into the Staples.ca price match garantee .. they offer 150% of difference up to $50 off lower price. They have for $699 |
| 15:32.21 | darkskiez | popfuzz: http://bugs.digium.com/bug_view_page.php?bug_id=0003782 maybe |
| 15:32.29 | fishboy1669 | hi anyone here know how to guestimate system resources for voicemail the intal spec is 10 boxes expanding up to 200 |
| 15:32.38 | fishboy1669 | is it just down to hd size |
| 15:32.39 | tzanger | bjohnson: not a fan of colour laser |
| 15:32.39 | bjohnson | ChkDigit: Future Shop starting tomorrow is having sale on it for $350 |
| 15:33.27 | blitzrage | ok, what the hell. Updated Asterisk and now my menu prompts don't play to the end user |
| 15:33.41 | blitzrage | CLI says its playing them, but I don't hear anything... |
| 15:33.59 | bjohnson | ooops .. sale tomorrow is $380 |
| 15:34.00 | scrubb | anyone got an answer on the PCI-X thing? |
| 15:34.07 | facek_ | http://pastebin.ca/8541 |
| 15:34.14 | facek_ | can sb look and tell my with Goto doesn't work ? |
| 15:34.33 | bjohnson | also .. you can use coupon code 86814 to get $20 off of Staples order |
| 15:34.38 | pupfuzz | ManxPower: I see changes to chan_sip.c, but i'm not good enough of a coder to understand what the changes are doing. So I can do only one of two things: 1. revert back to an older version of CVS HEAD which works and then increment until the cause of problem. OR, log enough data that hopefully someone else can identify the cause. Option 1 would take a while since the bug is not immediately repeatable. |
| 15:34.46 | mgth | bjohnson: on what? |
| 15:34.50 | mgth | sale? |
| 15:35.17 | bjohnson | brings it down to $380 - $50 - $20 for this http://www.staples.ca/ENG/Catalog/cat_sku.asp?CatIds=88%2C341%2C5957%2C467,473&webid=567293&affixedcode=WW |
| 15:35.35 | *** join/#asterisk iq (~iq@70-57-182-28.omah.qwest.net) |
| 15:35.37 | blitzrage | facek_: doesn't work because it never hits extension 1 |
| 15:36.00 | facek_ | blitzrage why? i press 1 |
| 15:36.05 | Zeeek | facek_ I told you youhave to use Background to get the digits |
| 15:36.07 | blitzrage | facek_: you need Background() |
| 15:36.13 | blitzrage | facek_: not Playback() |
| 15:36.27 | blitzrage | facek_: Playback() doesn't accept digits |
| 15:36.41 | facek_ | but how make backgorund and saydigit? |
| 15:36.58 | facek_ | i want in background 'for reepeat press:' |
| 15:37.00 | *** part/#asterisk chrislwade (~clwade@river104.bigriver.net) |
| 15:37.07 | facek_ | and in saydigit '1' |
| 15:37.34 | blitzrage | facek_: restate, I don't understand what you want |
| 15:37.39 | pupfuzz | darkskiez: Thanks. Reading bug 3782 now. The problem appeared somewhere in the past week or so, so this could be related. |
| 15:37.56 | blitzrage | facek_: just s/Playback/Background |
| 15:38.07 | facek_ | blitzrage i want to play that "For repeat press:" + "one" - but one is by SayDigit |
| 15:39.50 | blitzrage | facek_: you're going to need to use Background() |
| 15:40.04 | blitzrage | facek_: once you get to SayDigits(), it won't accept input anymore |
| 15:40.29 | facek_ | exten => s,3,Background(aby-powtorzyc-wiadomosc-nacisnij) |
| 15:40.29 | facek_ | exten => 1,1,Goto(s,2) |
| 15:40.32 | facek_ | not working still |
| 15:40.33 | blitzrage | facek_: if you need to use SayDigits(), might need to add a Read() |
| 15:40.50 | blitzrage | facek_: did you reload your extensions.conf? |
| 15:40.58 | facek_ | yes |
| 15:41.07 | blitzrage | facek_: Background() accepts digits while the file is playing |
| 15:41.08 | Zeeek | might be handy to READ some docs too since this issue has been on the agenda all day now |
| 15:41.15 | facek_ | ye is looking for 1 but no in macro-play, he is looking in [info] |
| 15:41.21 | facek_ | aaa |
| 15:41.37 | blitzrage | facek_: you need to learn fundamental dialplan concepts. Reading docs is the only way to do that. |
| 15:41.44 | Zeeek | and with that in mind |
| 15:41.46 | Zeeek | http://www.voip-info.org/wiki-Asterisk+Dialplan+Introduction |
| 15:41.46 | Zeeek | http://www.voip-info.org/wiki-Asterisk+config+extensions.conf |
| 15:41.46 | Zeeek | http://www.voip-info.org/wiki-Asterisk+Extension+Matching |
| 15:41.46 | Zeeek | http://www.voip-info.org/wiki-Asterisk+config+extensions.conf+sorting |
| 15:41.57 | Zeeek | and |
| 15:42.01 | SpaceBass | back |
| 15:42.06 | *** join/#asterisk hajekd (~hajekd@mail.idoox.com) |
| 15:42.13 | Zeeek | and http://asteriskdocs.org |
| 15:42.27 | SpaceBass | ok, so so I eliminated the 2 network issue, i put the access point on the same subnet as my * |
| 15:42.32 | SpaceBass | and the phone still won't register |
| 15:42.39 | hajekd | have you seen this error before when starting asterisk? |
| 15:42.39 | hajekd | Mar 31 05:39:32 WARNING[5633]: /usr/lib/asterisk/modules/app_readfile.so: undefined symbol: ast_read_textfile |
| 15:42.39 | hajekd | Mar 31 05:39:32 WARNING[5633]: Loading module app_readfile.so failed! |
| 15:43.28 | facek_ | ok |
| 15:43.56 | *** join/#asterisk alecm (~alec@adsl1.kennedysgroup.com) |
| 15:44.24 | SpaceBass | anyone ever used a zyxel wifi sip phone? |
| 15:44.24 | Egonis | Do I need zapata? |
| 15:44.33 | SpaceBass | (and save the bad reviews, I've heard them all) |
| 15:44.37 | Zeeek | no but I've heard less than kind things... |
| 15:44.41 | Zeeek | oops |
| 15:44.46 | SpaceBass | lol |
| 15:44.46 | SpaceBass | :) |
| 15:44.49 | pupfuzz | Can someone tell me how to log the output of sip debug to a file? |
| 15:45.04 | SpaceBass | I have heard a mixed bag... but the bottom line is that it was $100 on ebay |
| 15:45.12 | Zeeek | so enjoy! :) |
| 15:45.19 | SpaceBass | and I've tried a regular cordless on a ATA and the quality was so bad people stopped taking my calls |
| 15:45.34 | Zeeek | funny, I've had good luck with the IAXy |
| 15:45.35 | SpaceBass | i just cannot get the thing to register |
| 15:45.41 | SpaceBass | Zeeek this is a pa168 |
| 15:45.44 | Zeeek | that was part of the reviews... |
| 15:46.00 | Zeeek | SpaceBass are there other IAX clients on the same LAN? |
| 15:46.14 | SpaceBass | Zeeek the pa168 is the only IAX2 client at all |
| 15:46.19 | SpaceBass | the rest are sip softphones |
| 15:46.27 | Zeeek | ok because I had some problems with that |
| 15:46.49 | Zeeek | I hear also that they don't also set the port right (they=PA168 stuff) |
| 15:46.56 | mentat | anyone have experience using iaxy or spa-200x "on the road" through NAT and crap? |
| 15:47.09 | Zeeek | not on the road but through NAT |
| 15:47.13 | SpaceBass | well, it connected just fine, but the audio quality of the outgoing end was too poor |
| 15:47.20 | SpaceBass | i heard people a-ok |
| 15:47.23 | Zeeek | IAXy? |
| 15:47.29 | SpaceBass | but they said I sounded like I was being run over by a train underwater |
| 15:47.39 | *** part/#asterisk lbarth (user@pD9563464.dip.t-dialin.net) |
| 15:47.39 | Zeeek | and of course the phone was a good one |
| 15:48.09 | SpaceBass | the cordless? vtech 2.4G |
| 15:48.13 | SpaceBass | never had problem using it on ptsn |
| 15:48.30 | Zeeek | odd |
| 15:48.36 | bannerman | hm.. when dialing certain 800 numbers (18008880262 for HP for instance) using livevoip, I just get a fast busy signal .. any ideas? |
| 15:48.57 | Zeeek | bannerman sometimes it's because of callerid |
| 15:49.04 | Zeeek | what are you sending? |
| 15:49.05 | bannerman | Zeeek: thanks, I bet that's it |
| 15:49.18 | bannerman | Zeeek: not the right thing ;-) |
| 15:49.22 | Zeeek | JerJer put me onto that a while back |
| 15:49.22 | SpaceBass | Zeeek I tried it with SIP and IAX2 neither worked well... was pretty fustrating b/c I have heard great things about the 168 chipset |
| 15:49.35 | Zeeek | I have a 168 phone |
| 15:49.44 | SpaceBass | I'm about to give up on my asterisk as a home voip project expirement |
| 15:49.46 | Zeeek | kinda neat at $70 new |
| 15:49.55 | SpaceBass | so far all I have is a glorified answering machine (I do like that part a lot!) |
| 15:49.59 | Zeeek | but the handset is bad - needs a new mic |
| 15:50.20 | Zeeek | SpaceBass I have mine at the office so it's a cheap pbx AND a glorified answering system |
| 15:50.22 | SpaceBass | bought two IP phones off e-bay... they look awesome, but are lacking firmware apparently and the company is out of business |
| 15:50.41 | Zeeek | Maybe the publicly available firmware would work? |
| 15:50.47 | SpaceBass | and now I cannot get this wifi sip phone to work... so I'm thinking I call it a day... which makes me sad |
| 15:50.47 | Nugget | SpaceBass: do NOT buy the zyxel wifi phone. it's a tragic piece of crap. |
| 15:50.52 | Zeeek | heh |
| 15:50.54 | *** join/#asterisk visionv (~sonye@c-67-184-112-58.hsd1.il.comcast.net) |
| 15:51.00 | *** join/#asterisk drinker (~lawrence@adsl1.kennedysgroup.com) |
| 15:51.00 | Zeeek | too late |
| 15:51.01 | Nugget | possibly the worst piece of hardware I've ever owned |
| 15:51.03 | Nugget | heh |
| 15:51.05 | SpaceBass | Zeeek unfortunatly the two IP phones are pretty propritary... i.picasso |
| 15:51.06 | *** part/#asterisk Alexis (~alexis@www.trim.it) |
| 15:51.10 | SpaceBass | Nugget too late... ebayed it |
| 15:51.12 | Zeeek | ah hear about that |
| 15:51.15 | Zeeek | heard |
| 15:51.23 | Jas_Williams | SpaceBass: I am using a Pulver Wisip with zytel firmware works for me |
| 15:51.30 | Zeeek | well if you have ,nothing to lose... |
| 15:51.39 | Nugget | why did you ask our opinion if you didn't want to hear the bad reviews, anyway? |
| 15:51.57 | SpaceBass | Nugget wasn't asking an opinion... was asking for support really :) |
| 15:52.12 | SpaceBass | Jas_Williams any problems getting it to register? |
| 15:52.50 | Zeeek | SpaceBass take a lookhere: http://www.aredfox.com/edownloads.htm |
| 15:52.51 | Jas_Williams | SpaceBass: No after I fixed it from failling to reregister after 30 mins |
| 15:52.56 | *** join/#asterisk RaYmAn-Bx (user@213.237.12.147.adsl.vby.tiscali.dk) |
| 15:53.00 | zoa | anyone from digium here ? |
| 15:53.09 | *** join/#asterisk _Sam-- (sam@ns2.kneedraggers.com) |
| 15:53.29 | SpaceBass | Zeeek thanks, I actually have that bookmarked already |
| 15:53.30 | SpaceBass | :) |
| 15:53.39 | Zeeek | might try the new (March 22) one |
| 15:53.46 | SpaceBass | thats what I'm running now |
| 15:53.49 | *** part/#asterisk _Sam-- (sam@ns2.kneedraggers.com) |
| 15:53.52 | Zeeek | and? no good? |
| 15:54.03 | Zeeek | what letter? S? |
| 15:54.24 | SpaceBass | have to check the letter |
| 15:54.28 | Zeeek | just curious |
| 15:54.35 | SpaceBass | v |
| 15:56.12 | SpaceBass | think s is different? |
| 15:56.27 | *** join/#asterisk techie (gus@asterisk.horizonte.us) |
| 15:57.27 | Jas_Williams | SpaceBass: Was this the IP Phone http://www.telradconnegy.com/ipicasso.htm |
| 15:57.33 | Jas_Williams | looks cool |
| 15:57.58 | SpaceBass | Jas_Williams yeah thats it... got two of em... they LOOK amazing |
| 15:58.11 | SpaceBass | but when they power on, they say "initializing.... checking for new version" |
| 15:58.21 | SpaceBass | and the company is now out of business... guess its missing firmware |
| 15:58.27 | *** join/#asterisk TheEmperor (TheEmperor@218.111.50.148) |
| 15:58.33 | Zeeek | SpaceBass I was on the phone - no the letter is for a type of chip I think |
| 15:58.42 | SpaceBass | Nugget beinging to agree with you re: the zyxel 2000w... it just registered for no reason... decided it wanted to... |
| 15:58.46 | Zeeek | so assuming you need the v... |
| 15:58.55 | SpaceBass | Zeeek I tried a few and V was the only one that worked |
| 15:58.56 | *** join/#asterisk JerJer[mobile] (~nonyobizn@65.173.197.174) |
| 15:59.12 | Zeeek | Maybe if you open the phone up you can see which one it uses? |
| 15:59.19 | SpaceBass | might do that |
| 15:59.46 | Zeeek | that really is a cool screen |
| 16:00.03 | SpaceBass | isn ti |
| 16:00.05 | SpaceBass | it? |
| 16:00.05 | Zeeek | can you send naked girl photos to it? |
| 16:00.12 | SpaceBass | came with a stylis too... |
| 16:00.13 | Zeeek | that'sd be hot |
| 16:00.16 | Zeeek | oooo |
| 16:00.16 | Nugget | a few people in here speak fondly of their hitachi wifi phones. That's probably what I'll try next. |
| 16:00.19 | SpaceBass | ohhh naked girl photos... AWESOME |
| 16:00.28 | Zeeek | with the stylus...ooohhh |
| 16:00.37 | Zeeek | where to put it? |
| 16:00.38 | SpaceBass | Nugget I've heard goot things about them as well... might shell out for one or ebay a cisco |
| 16:00.42 | SpaceBass | lol |
| 16:01.23 | Nugget | the cisco is a waste of time. |
| 16:01.24 | Zeeek | what is wrong with it? It doesn't register? |
| 16:01.26 | ManxPower | get a polycom |
| 16:01.27 | Nugget | it won't do SIP |
| 16:01.35 | Nugget | polycom makes a wifi phone? |
| 16:01.37 | SpaceBass | Zeek the I.picasso? doesnt boot |
| 16:01.39 | bannerman | Geez.. I'm stumped. I can't figure out how to set callerid out outbound calls. It just sends my extension number (107) |
| 16:02.04 | facek_ | hwo make backgroudn to be repeated under sb presss sth or under will be time out ? |
| 16:02.14 | Nugget | bannerman: sounds like you ARE setting callerid, then. |
| 16:02.27 | bannerman | apparently so, yes :) it's pulling it from my sip.conf |
| 16:02.32 | ManxPower | bannerman: you cannot send NAME to the telco |
| 16:02.57 | blitzrage | bannerman: SetCIDNum() |
| 16:02.58 | oej | bannerman: Use option "o" in dial or setcallerid |
| 16:03.04 | Zeeek | bannerman no DID on this same account? |
| 16:03.17 | SpaceBass | ohhh bad echo on the ZyXel |
| 16:03.18 | bannerman | I do have a did on this account |
| 16:03.26 | Zeeek | maybe you MUST use that number? |
| 16:03.31 | bannerman | I am :) |
| 16:03.33 | bannerman | trying anyhow |
| 16:03.35 | Zeeek | *oh. |
| 16:04.17 | bannerman | blitzrage: so I should do SetCIDNum(xxxxxxxxxx) right before my Dial(xx)? |
| 16:05.03 | ManxPower | bannerman: yes. Callerid numbers do NOT have special characters in them like space, dash or parends |
| 16:05.30 | ManxPower | Does anyone here know how to use patlooptest? |
| 16:05.51 | *** join/#asterisk lyoungz_ (lyoungz@x40347751.ip.e-nt.net) |
| 16:06.46 | bannerman | sweet, that works. thanks, guys |
| 16:06.59 | bannerman | You know, for all the tons of info in the wiki, it's sure devoid if useful information sometimes :-P |
| 16:07.27 | blitzrage | bannerman: its there, you just need to find it |
| 16:08.21 | bannerman | blitzrage: It may be there, but it doesn't come up with searching, and it's not linked to from any related pages... |
| 16:09.31 | *** join/#asterisk Los415 (~los415@c-24-126-63-233.hsd1.ca.comcast.net) |
| 16:09.33 | blitzrage | bannerman: you probably could have found what you were looking for by doing a "show applications" and "show application ........" from the CLI |
| 16:10.06 | Zeeek | bannerman I've found that giving google callerid wiki asterisk usually brings the stuff up on the first page (without site:) |
| 16:10.25 | Zeeek | of course you have to know the word you're looking for |
| 16:11.07 | Zeeek | for example google .call file brings up the right page |
| 16:11.36 | bannerman | Sure, I knew about SetCIDNum(). But how was I to know that I had to use that instead of setcallerid, and where to use it? How was I to know that it was intended to be used for sending cid to the telco, instead of putting callerid=xxx in my peer section in iax.conf? |
| 16:11.52 | Zeeek | both have their uses |
| 16:11.59 | bannerman | That's good |
| 16:12.05 | Nugget | bannerman: you don't have to use it instead of setcallerid. |
| 16:12.07 | Nugget | both work. |
| 16:12.11 | SpaceBass | Zeeek does your PA168 have audio settings? |
| 16:12.12 | JerJer[mobile] | bannerman: because it is an application that gets used on the dialplan |
| 16:12.21 | Zeeek | tons, SpaceBass |
| 16:12.26 | JerJer[mobile] | ala extensions.conf |
| 16:12.44 | JerJer[mobile] | i use the callerid fields for internal calleridi |
| 16:12.57 | Zeeek | SpaceBass do you have Windows? There is a loader for use with it |
| 16:13.00 | JerJer[mobile] | then override the internal with SetCIDNum for sendng out to the pstn |
| 16:13.26 | Nugget | yeah, that's the sane way to do it. |
| 16:13.31 | ManxPower | I have an AGI that does that., |
| 16:13.34 | SpaceBass | Zeeek yeah, pretty much have the OS spectrum covered here |
| 16:13.41 | SpaceBass | Zeeek by loader do you mean for the firmware? |
| 16:13.48 | Nugget | if you're calling another internal extension, callerid should be 107.. |
| 16:13.50 | Zeeek | SpaceBass did you find the win utility that does all the loading? |
| 16:13.53 | bannerman | I know. I'm coming here as a newb to VoIP. I've worked in IT since I was 16. I know how to google for information, I'm the guy people call when they don't want a booklearned exper, but they awnt somoene who can make things work. And I'm trying to let the general community know that the information needed to properly do the job isn't readily available on the wiki. |
| 16:13.58 | Zeeek | firmware AND config |
| 16:14.06 | SpaceBass | is that the PA tool? |
| 16:14.08 | SpaceBass | i haven't tried it |
| 16:14.10 | Nugget | bannerman: I disagree completely. |
| 16:14.14 | Zeeek | yeah it's great |
| 16:14.15 | bannerman | Not trying to make a point or beg for help or anything |
| 16:14.27 | Nugget | everyone in this channel managed to learn it. obviously it can be done |
| 16:14.27 | SpaceBass | been using the web config |
| 16:14.29 | Zeeek | no one's saying that bannerman |
| 16:14.31 | SpaceBass | which leaves something to be desired |
| 16:14.33 | ManxPower | I agree with bannerman, but not enough to spend the next 6 months of my life fixing the Wiki |
| 16:14.37 | Zeeek | heh |
| 16:14.45 | Nugget | the wiki is hard to search, but the information is there. |
| 16:15.03 | Zeeek | I have to say, people should force themselves to update it and add to it |
| 16:15.03 | SpaceBass | Zeeek the problem is that the audio from my end is very garbled and muted |
| 16:15.07 | bannerman | Nugget: Not all of it. Things that you take for granted, that seem obvious to you, are not obvious to other people. |
| 16:15.15 | SpaceBass | unless I yell into the thing, it drops off |
| 16:15.35 | Zeeek | SpaceBass my audio recvd is good, the xmitted audio sucks but it's the handset mic, I tried another one to see |
| 16:15.35 | bannerman | And it's not all because people are stupid. Just most of it. |
| 16:15.50 | Nugget | nothing about asterisk seemed obvious to me. |
| 16:15.59 | Pj386 | bannerman: I agree completely with the fact the info on the wiki isn't all ready and ovious |
| 16:16.02 | Pj386 | obvious |
| 16:16.06 | Zeeek | bannerman it's true. There's a lot of obscure shit that needs to be more widely known |
| 16:16.13 | Nugget | but I think you're overstating the situation |
| 16:16.25 | SpaceBass | you know, now that this ZyXel is working, it seems to be ok... good audio and great coverage... i just have to have a damn wide open AP on my network |
| 16:16.29 | Nugget | and, as ManxPower hinted at, if you really dislike it -- go fix it. |
| 16:16.37 | Nugget | the wiki is the fault of all of us. |
| 16:16.38 | bannerman | I can't, I don't know how yet ;) |
| 16:16.38 | Zeeek | However, it's also true (this is not directed at you bannerman) that a large majority of newbies haven't taken much trouble to serahc |
| 16:16.45 | Zeeek | search |
| 16:16.45 | Pj386 | bannerman: but will you ? :) |
| 16:16.51 | bannerman | Pj386: hell no. :) |
| 16:16.56 | Zeeek | because if they did, they'd not be asking |
| 16:17.09 | Pj386 | bannerman: so just shut up and read ;) |
| 16:17.19 | bannerman | Pj386: /salute :-P |
| 16:17.21 | Zeeek | I have added obscure stuff twice in one year - does that qualify? |
| 16:17.51 | Pj386 | You can't say you're expecting things from people that people cannot expect from you |
| 16:17.54 | bannerman | I'm not complaining, just stating the obvious that I think flies under the radar for a lot of people. |
| 16:18.05 | *** join/#asterisk anonymous^ (~strace@ADSL-F49-S197-critical-coi.nortenet.pt) |
| 16:18.07 | Pj386 | Ohhh ok |
| 16:18.08 | anonymous^ | hey gang |
| 16:18.16 | Pj386 | then you're stating the obvious alright ;) |
| 16:18.23 | anonymous^ | how do I allow anonymous SIP phones to place calls? |
| 16:18.27 | Pj386 | And it's a lot better than a year ago |
| 16:18.35 | bannerman | Pj386: I'll complain a bit too in a few minutes when I have more time |
| 16:18.39 | bannerman | if you want |
| 16:18.52 | *** join/#asterisk outtolunc (~me@adsl-66-218-53-170.dslextreme.com) |
| 16:19.02 | ManxPower | I firmly believe there are far too many Asterisk documentation sites out there. Everyone should close their site and donate their entire site to the asteriskdocs.org site |
| 16:19.09 | SpaceBass | Zeeek what are jitter size and audio frames? |
| 16:19.39 | Zeeek | stuff to play with :) |
| 16:19.42 | SpaceBass | lol |
| 16:19.44 | SpaceBass | :) |
| 16:20.00 | *** join/#asterisk lesouvage (~lesouvage@cc341200-a.assen1.dr.home.nl) |
| 16:20.03 | Zeeek | wait a sec I have a paper here |
| 16:20.11 | SpaceBass | opinions on codec? g711u or a? |
| 16:20.39 | Zeeek | mine says frames=2 jitter... can't find |
| 16:20.42 | Pj386 | ManxPower: true |
| 16:22.09 | Zeeek | I usually try ulaw first |
| 16:22.17 | Zeeek | especially if the phone is local |
| 16:22.23 | TheEmperor | hi all |
| 16:22.36 | Zeeek | these phones do 729 and iLBC as well as gsm |
| 16:23.01 | TheEmperor | can anyone tell me why is it when i call an extension, i can hear them great but they can't hear me and there is a delay sometimes? |
| 16:23.10 | Zeeek | SpaceBass try the utiklity and see if it talks to the phone (set the ip first) |
| 16:23.10 | ManxPower | TheEmperor: no. |
| 16:23.24 | Zeeek | how can there be a delay if they can't hear it? |
| 16:23.28 | SpaceBass | Zeeek using the utility now, and its working |
| 16:23.33 | *** join/#asterisk jeffik (~jeffik@CPE00c049565af7-CM0012256ead9e.cpe.net.cable.rogers.com) |
| 16:23.35 | Zeeek | it's pretty neat |
| 16:23.41 | SpaceBass | yeah it is |
| 16:23.43 | Zeeek | faster than the web page |
| 16:23.57 | TheEmperor | what I mean is sometimes the other party can't hear me at all |
| 16:23.58 | Zeeek | sleeker than tftp |
| 16:24.04 | TheEmperor | and if they can hear me, there is a delay |
| 16:24.10 | SpaceBass | i like it |
| 16:24.11 | ManxPower | TheEmperor: I know what you mean. That can be caused by about 400 things. |
| 16:24.41 | TheEmperor | ManxPower: so where do I start?.. |
| 16:24.50 | ManxPower | TheEmperor: Nat/ |
| 16:25.14 | SpaceBass | but still doesnt fix my shitty audio :) |
| 16:25.15 | TheEmperor | ManxPower: also, it gets worse when I use LiveVoip or Nufone to call out |
| 16:25.30 | Zeeek | SpaceBass there are a few volume settings too |
| 16:25.50 | ManxPower | TheEmperor: Start by showing us your network diagram. Example: Telco(T-1)<->(T-1)Asterisk<->Internet<->Router(NAT)<->Phone |
| 16:26.04 | ManxPower | TheEmperor: sounds like your ISP or router sucks |
| 16:26.17 | TheEmperor | ManxPower: I am using a linksys router, would that be the problem? |
| 16:26.29 | SpaceBass | Zeeek i have played with those a lot... i have the handset vol all the way up... thats the best audio I can get |
| 16:26.36 | ManxPower | TheEmperor: not normally |
| 16:26.43 | ManxPower | TheEmperor: I'm still waiting for your diagram |
| 16:26.48 | TheEmperor | ManxPower: it's on a 1mbps leased line |
| 16:26.51 | TheEmperor | ManxPower: I am using a linksys router, would that be the problem? |
| 16:26.56 | ManxPower | TheEmperor: I'm still waiting for your diagram |
| 16:28.11 | ManxPower | TheEmperor: Start by showing us your network diagram. Example: Telco(T-1)<->(T-1)Asterisk<->Internet<->Router(NAT)<->Phone |
| 16:28.25 | Zeeek | SpaceBass mayeb the timezone is wrong? |
| 16:29.02 | TheEmperor | ManxPower:ISP(1Mbps)<->Router<->Asterisk<->Internet<->Router<->Phone |
| 16:29.03 | sudhir492 | JerJer[mobile]: I modified the Makefile in h323, and compiled with PWLib 1.8.1, Openh323 1.17.1 |
| 16:29.17 | TheEmperor | ManxPower: I hope that's right |
| 16:29.18 | ManxPower | TheEmperor: so everything is on a piblic IP? |
| 16:29.18 | bannerman | should I send a 10 digit number, or add a 1 to the start? |
| 16:29.32 | ManxPower | TheEmperor: sorry, I cannot help you further |
| 16:29.43 | ManxPower | bannerman: 10 digits. Always |
| 16:29.46 | sudhir492 | Seems to work fine so far, though see a lot of messages: RFC3389: 1 bytes, level 256... |
| 16:29.47 | TheEmperor | ManxPower: Yes, my Asterisk server is on a public IP |
| 16:29.49 | Bonbon | i need a russian incoming number |
| 16:30.30 | TheEmperor | ManxPower: I have been having this problem for a while, I can always hear the other party very well, but they either can't hear me or there is a substantial delay |
| 16:31.29 | TheEmperor | can anyone help on this? |
| 16:32.00 | Zeeek | TheEmperor what is the lag time to your provider? |
| 16:32.13 | jakepdev | always hav that problem with my wife |
| 16:32.30 | facek_ | Zeeek Agents are used whan I want to make a callcanter.. for only answet phone, yes? |
| 16:32.32 | bannerman | ManxPower: thakns |
| 16:32.41 | bannerman | ManxPower: And thanks, as well. |
| 16:32.54 | SpaceBass | jakepdev LOL!!! |
| 16:33.01 | facek_ | but if i wants to use queues for nromal user u must use queues.conf end set nromal user as member |
| 16:35.26 | AmaDEE0_ | When using a DB as a back end for logs or config, is Asterisk fairly tied to MySQL? It seems the Wiki talks about it quite a bit. |
| 16:35.39 | AmaDEE0_ | Not much mention of alternates (a bit on Postgresql). |
| 16:36.30 | TheEmperor | Zeeek: what do you mean? |
| 16:36.58 | Zeeek | how far in milliseconds is your service provider? |
| 16:38.44 | TheEmperor | Zeeek: I have a leased line 1 mbps |
| 16:39.08 | Zeeek | oh that's great |
| 16:39.17 | Zeeek | but it doesn't answer the question |
| 16:39.30 | TheEmperor | Zeeek:how can I check? |
| 16:39.33 | *** join/#asterisk faux-jenn (~fjenn@216.199.225.11) |
| 16:39.38 | faux-jenn | hello everyone |
| 16:39.44 | Zeeek | SIP show peers |
| 16:39.54 | Zeeek | or IAX2 if it's IAX |
| 16:40.03 | jhiver | TheEmperor: looks like you need to be doing some traffic shaping |
| 16:40.20 | Zeeek | don't get him confused |
| 16:40.34 | jhiver | it could be that your line upstream is saturated, hence crappy sound for ppl you call |
| 16:41.25 | AmaDEE0_ | How much bandwith does a IAX2 or SIP call take up? Roughly. Between say 3Kb and 7Kb sec? |
| 16:41.52 | Hmmhesays | ahah the predecessor to C was B I never knew that |
| 16:42.25 | zoa | AmaDEE0_ see www.astertest.com ppt |
| 16:42.26 | jhiver | Zeeek: my ISP is like 400 ms aways and I can still make decent calls |
| 16:42.36 | AmaDEE0_ | Also, how do you deal with internal QoS when setting up VoIP in an office? Make them get a QoS switch? You can't have the calls frap out when Bob decides to download a DVD. |
| 16:42.49 | jhiver | if jitter and packet loss is not too bad then it's ok |
| 16:43.20 | *** join/#asterisk cyberkoa (~cyberkoa@65.90.93.3) |
| 16:43.31 | jhiver | AmaDEE0_: I use linux to make an ethernet bridge between the lan and the dsl and I shape on the bridge |
| 16:43.49 | Nugget | it would take quite a bit of internal traffic to saturate most networks. |
| 16:45.10 | AmaDEE0_ | jhiver: But once it gets on the ADSL you can't be sure any other routers will obey the QoS right? |
| 16:45.44 | jhiver | no you can't |
| 16:46.14 | jhiver | but some companies such as equant do guaranteed bandwith VPNs |
| 16:46.21 | Bonbon | any news on a russian number? |
| 16:46.28 | jhiver | in which case you can - but it's more expensive than DSL... |
| 16:46.49 | Nugget | I use an openbsd bridge, much like jhiver's I presume, to shape my dsl traffic. it works like a champ. |
| 16:47.06 | jhiver | that being said, _if_ you do prioritize properly on both ends, then it'll prolly be fine |
| 16:47.28 | jhiver | if you suspect an ISP is fucking about with VoIP traffic, just use a VPN (such as OpenVPN) between the two |
| 16:48.02 | jhiver | openbsd sounds good cause you can also 'attempt' to shape inbound traffic - which Linux doesn't do |
| 16:48.25 | jhiver | but in my experience with any DSL shaping outbound is good enough |
| 16:48.41 | Nugget | I went with openbsd mainly for its ability to do bidirectional filtering in a three-interface bridge, which nothing else could do at the time. (and maybe that's still the case) |
| 16:48.44 | jhiver | another thing that looks good is netequalizer, but it's commercial only :( |
| 16:48.51 | faux-jenn | I've yet to install Asterisk, but want to know if you can set outgoing caller-id to anything you'd want... |
| 16:49.02 | faux-jenn | using VoicePulse also |
| 16:49.11 | Nugget | faux-jenn: voicepulse connect will permit that, yes. |
| 16:49.20 | jhiver | Yeah openbsd looks awesome for network routing |
| 16:49.22 | Nugget | callerid number, that is. you can't set the name. |
| 16:49.28 | Nugget | altq+pf is an amazing tool. |
| 16:49.31 | SpaceBass | anyone got a recomendation for calling waiting and PSTN? |
| 16:49.31 | faux-jenn | is the setting in Asterisk or the SoftPhone (Xlite) |
| 16:49.39 | Nugget | faux-jenn: asterisk |
| 16:49.40 | *** join/#asterisk [Pi] (Pi@mstr195175-29290.dial-in.ttnet.net.tr) |
| 16:49.52 | faux-jenn | -->Nugget:Thanks. Can't wait to try it. |
| 16:49.58 | jhiver | but you know, for now linux is good enough soooo... it's not broke so I won't fix it :) |
| 16:50.44 | Nugget | when it does break (and since it's linux, it will :), consider openbsd. |
| 16:51.01 | jhiver | yeah eventually :) |
| 16:51.09 | faux-jenn | Nugget:Any config differneces running Asterisk on OpenBSD versus Linux? |
| 16:51.11 | jhiver | don't know, i've slapped a knoppix on a box |
| 16:51.15 | jhiver | removed anything I could |
| 16:51.20 | jhiver | use the 2.6 kernel |
| 16:51.22 | Nugget | since linux seems to completely replace the packet filtering tool on each kernel rev, it's only a matter of time. :) |
| 16:51.23 | jhiver | works great |
| 16:51.30 | Nugget | faux-jenn: I don't know, I don't run asterisk on openbsd. |
| 16:51.42 | jhiver | does shaping, bridge, vpn, asterisk, you name it :) |
| 16:51.44 | faux-jenn | Nugget: What DO you run it on, if I may ask. |
| 16:51.49 | Nugget | freebsd and linux |
| 16:51.51 | Nugget | and os x |
| 16:51.58 | faux-jenn | Nugget:Thanks. |
| 16:52.02 | jhiver | also a nice software raid with a couple of 200 gb disks for backups |
| 16:52.12 | jhiver | incremental backups with rsync... |
| 16:52.12 | [Pi] | hi all... I am learnin asterisk new... and I wonder if Asterisk can transfer Voices from anybodies' computer to anothers'... from one country to another... any idea? |
| 16:52.14 | jhiver | nice :) |
| 16:52.21 | Nugget | linux is the only platform I'd recommend for asterisk, unless you know what you're doing. |
| 16:52.29 | jhiver | [Pi] yes it can |
| 16:52.40 | Nugget | anything but linux is a second-class citizen in the asterisk world and you'll invariably have some difficulties eventually |
| 16:52.58 | Nugget | it's getting better, which is encouraging, but it's not "there" yet |
| 16:53.06 | [Pi] | jhiver: Is there any client software for taking to other computers? |
| 16:53.17 | jhiver | [pi] yes |
| 16:53.19 | Shido6 | talking to other computers? |
| 16:53.25 | jhiver | you can use a bunch of softphones |
| 16:53.35 | jhiver | so basically you have asterisk as a server |
| 16:53.46 | jhiver | and softphones like firefly as clients |
| 16:54.08 | jhiver | you can also use hardphones (i.e. VoIP phones that look like a proper phone) to talk to asterisk |
| 16:54.15 | [Pi] | i mean I want to have a software and want to talk with people (with computers) in other countries.... |
| 16:54.18 | [Pi] | is it possible? |
| 16:54.26 | tzanger | pif: why not? |
| 16:54.29 | tzanger | er [Pi] |
| 16:54.37 | jhiver | yes, that's what it's meant to do... |
| 16:54.54 | jhiver | but maybe you should be starting out with skype to ease the learning curve :) |
| 16:54.54 | [Pi] | so I purchased a server machine .. and I want to install asterisk to fedora core 2 |
| 16:55.31 | [Pi] | I dont want skype.. I want to use my own server... |
| 16:55.39 | jhiver | ~search asterisk fc2 rpm |
| 16:55.48 | jhiver | is it ~search or ~google? |
| 16:55.54 | *** join/#asterisk ^HeLL^ (~admin@217.11.115.168) |
| 16:55.57 | jhiver | ~google asterisk fc2 rpm |
| 16:56.07 | ^HeLL^ | hello all |
| 16:56.13 | jhiver | HeLL lo |
| 16:56.25 | SpaceBass | i have usecallwaiting=yes in my zapata.conf |
| 16:56.27 | ^HeLL^ | :) |
| 16:56.31 | [Pi] | hi ^HeLL^ |
| 16:56.36 | SpaceBass | but when a call comes in from the pstn I have no way switching over to it |
| 16:56.48 | jhiver | guys why do use nicks that are incredibly hard to type? |
| 16:57.02 | SpaceBass | do I need to configure something on the sip or IAX client end? |
| 16:57.08 | jhiver | you know, ^HeLL^ is kind of a bitch to sort out :) |
| 16:57.24 | Zeeek | two keys from here |
| 16:57.33 | Zeeek | ^ <tab> |
| 16:57.34 | jhiver | I wish all nicks were lower case, [a-z] and maybe '-' only |
| 16:57.42 | Shido6 | tab completion |
| 16:57.47 | Nugget | does jbot have a nickometer? |
| 16:57.55 | jhiver | don't know I use xchat (aka n00b's irc client) |
| 16:58.05 | Nugget | jbot: nickometer jhiver |
| 16:58.05 | ^HeLL^ | tab is the best friend for the kids. :) |
| 16:58.10 | tzanger | oh he's back |
| 16:58.11 | Nugget | jbot:nickometer ^HeLL^ |
| 16:58.14 | [Pi] | and one last question.... Do I have to integrate this server a Wildcard for transfering voice packets from one country to another? |
| 16:58.14 | tzanger | jbot you dumbass |
| 16:58.26 | Zeeek | ~got milk |
| 16:58.27 | jbot | ACTION chugs down a carton |
| 16:58.29 | AngelGabriel | what's a nickometer? |
| 16:58.31 | SpaceBass | call waiting coming in from PTSN, how do I "flash" over to it |
| 16:58.42 | jhiver | oh tab completion does work! cool.... |
| 16:58.45 | Zeeek | SpaceBass on my SIP hardphone it's the R key |
| 16:58.51 | AngelGabriel | jbot:nickometer AngelGabriel |
| 16:58.52 | jhiver | ^HeLL^: ^HeLL^ ^HeLL^ |
| 16:58.54 | jhiver | yeah! |
| 16:58.57 | bannerman | Well, I do for sure have my CID set correctly, but I still get a fast busy signals when calling certain toll free numbers. I guess I have to wait for LiveVOIP to get back to me. |
| 16:59.11 | Zeeek | bannerman is your DID a tollfree? |
| 16:59.15 | bannerman | yes |
| 16:59.30 | SpaceBass | Zeeek I dont have a flash or r key on my hardphone and on my cordless->ATA flash does nothing |
| 16:59.31 | Zeeek | some tollfree numn will NOT accept calls from tollfree |
| 16:59.35 | bannerman | oh.. hm |
| 16:59.41 | jhiver | jbot:nickometer 7e|-|_r0x0r3d |
| 16:59.44 | Zeeek | I have real life experience with this |
| 16:59.46 | jhiver | heh |
| 16:59.48 | SpaceBass | Zeeek do I have to configure anything special in sip.conf for the phone? |
| 17:00.01 | Zeeek | my solution was to put my mother's number and hope they don't bother her |
| 17:00.31 | jhiver | jbot:nickometer j00f001! |
| 17:00.32 | Zeeek | SpaceBass I don't think so |
| 17:00.38 | bannerman | heh thanks Zeeek, problem solved |
| 17:00.46 | SpaceBass | jbot:nickometer spacebass |
| 17:00.52 | jhiver | cool game :) |
| 17:00.55 | Zeeek | once again, JerJer was at the origin of that solution |
| 17:00.59 | bannerman | I guess I should have two dialing rules, one for toll free and one for everything else, toll free gives them our local number and everything else gives them the 800 number |
| 17:01.00 | SpaceBass | yea jbot... you go |
| 17:01.05 | ^HeLL^ | anybody happends anything like this when you compile chan_h323 ? |
| 17:01.05 | bannerman | JerJer: I indirectly love you. |
| 17:01.07 | ^HeLL^ | In file included from chan_h323.c:53: |
| 17:01.18 | ^HeLL^ | usr/include/asterisk/lock.h: In function `ast_mutex_init': |
| 17:01.26 | Nugget | jbot: nickometer SpaceBass |
| 17:01.28 | ^HeLL^ | usr/include/asterisk/lock.h:299: `PTHREAD_MUTEX_RECURSIVE' undeclared (first use in this function) |
| 17:01.34 | Nugget | hrm, different than I'm used to. |
| 17:01.38 | *** join/#asterisk RaYmAn-Bx (user@213.237.12.147.adsl.vby.tiscali.dk) |
| 17:01.40 | SpaceBass | lol |
| 17:01.45 | Zeeek | ^HeLL^ there's a bunch of old messages on the mailing list about this issue |
| 17:01.49 | Nugget | the original infobot code would have penalized you for the BuckyCaps |
| 17:02.07 | bannerman | Next problem: The entire company has gone overtime on cell minutes. How do I make the minutes smaller |
| 17:02.18 | SpaceBass | ahhh tab complete taks care of that :) |
| 17:02.23 | Zeeek | declare a mùinute as 90secs? |
| 17:02.28 | AngelGabriel | bannerman, Tell them to call less |
| 17:02.42 | Zeeek | AbsoluteTimeout(59) |
| 17:02.49 | ^HeLL^ | Zeeek : i know, but adding the #define _GNU_SOURCE do nothing |
| 17:02.59 | Zeeek | That's all I know |
| 17:03.29 | ^HeLL^ | almost, this error converts into a undefined ast_.... variables... |
| 17:06.21 | *** join/#asterisk sandnigg0r (~crunkuser@66-55-197-254.gwi.net) |
| 17:06.38 | SpaceBass | how can I get asterisk to recognize commands (* or 1-9) during a call? like if I want to start recording the call or want to conference or something? |
| 17:06.45 | *** join/#asterisk vidia22 ([U2FsdGVkX@vidiamob4.vidiacom.com) |
| 17:06.57 | SpaceBass | i would search, but not ever sure what to call that |
| 17:07.40 | Zeeek | look at features.conf |
| 17:07.45 | *** join/#asterisk mass_murderer (murderer@200-171-226-51.customer.telesp.net.br) |
| 17:07.56 | Zeeek | and barge |
| 17:07.58 | vidia22 | newbi x100p installation question... When I boot up the box with the new x100p cards should I allow kudzu (hardware manager) to configure them..(it thisks they are tiger cards) |
| 17:08.05 | *** join/#asterisk heison (~heison@ns.somanetworks.com) |
| 17:08.17 | SpaceBass | Zeeek thanks... i cannot believe I hadnt checked out that .conf yet |
| 17:08.40 | SpaceBass | vidia22 no, use genzaptelconf |
| 17:08.52 | bannerman | AngelGabriel: Brilliant! I'll do that :) |
| 17:08.58 | Zeeek | I think there isd a barge for sip et al |
| 17:09.08 | SpaceBass | barge? |
| 17:09.20 | Zeeek | errr monitor? Something? |
| 17:09.20 | vidia22 | SpaceBass: so I should tell kudzu at boot up to ignoe or do nothing? |
| 17:09.29 | SpaceBass | vidia22 i think so |
| 17:09.37 | SpaceBass | Zeeek is that what is monitoring for the command? |
| 17:09.51 | SpaceBass | for instance, in features.conf I have parkext = 70 |
| 17:09.58 | SpaceBass | but dialing 70 during a call does nothing |
| 17:09.59 | Zeeek | monitor is for recording i b'leive |
| 17:10.01 | heison | anyone here using SER with Asterisk? |
| 17:10.20 | *** join/#asterisk [cc]smart (~smart@gw.ptr-62-65-149-158.customer.ch.netstream.com) |
| 17:10.20 | Zeeek | parking would be #700 |
| 17:10.33 | Zeeek | and you must include the parking context |
| 17:10.42 | SpaceBass | Zeeek why? if I have 70 set up in my features.conf? |
| 17:10.52 | Zeeek | I think you still need the # |
| 17:10.56 | Zeeek | or is it * ? |
| 17:11.04 | Zeeek | but I know nothing |
| 17:11.28 | SpaceBass | Zeeek http://pastebin.ca/8547 |
| 17:11.39 | SpaceBass | if you have a second |
| 17:12.08 | Zeeek | why is there a # ? |
| 17:12.22 | Zeeek | bad comment? |
| 17:12.28 | vidia22 | SpaceBass: is genzaptelconf part of the zaptel dist or do I need to go and get it? |
| 17:12.38 | vidia22 | I haent used that before... |
| 17:12.43 | SpaceBass | vidia22 should be part of the zaptel dist |
| 17:13.01 | SpaceBass | Nugget you are right, this thing is a POS... doesnt support call waiting |
| 17:13.09 | Zeeek | SpaceBass dial #70 and it should work |
| 17:13.20 | Zeeek | but there must be a 't' or 'T' option in the dial command |
| 17:13.35 | Zeeek | this is my old knowledge, that all may have changed |
| 17:13.55 | vidia22 | SpaceBass; THANKS :) mucho appreciated :) |
| 17:14.03 | SpaceBass | vidia22 good luck! |
| 17:14.55 | SpaceBass | Zeeek asterisk doesnt seem to recognize any commands internally while in a call |
| 17:14.56 | *** join/#asterisk nel (~oeo@199.75.114.196) |
| 17:15.07 | Zeeek | show application dial |
| 17:15.31 | Zeeek | The option string may contain zero or more of the following characters: |
| 17:15.31 | Zeeek | <PROTECTED> |
| 17:15.34 | nel | I need some help with a tdm04b card and echo |
| 17:16.07 | nel | I have been using ztmonitor and I can see my voice in the rx when I seppek |
| 17:16.09 | nel | speek |
| 17:16.10 | nel | speakj |
| 17:16.26 | loud | does anyone have any news regarding livevoip and their did numbers ? |
| 17:16.33 | nel | is there any echo cancellation feature on polycom phones? |
| 17:16.34 | *** join/#asterisk pluto70 (~me@80.70.179.76) |
| 17:16.55 | SpaceBass | Zeeek gotcha |
| 17:17.11 | SpaceBass | Zeeek that helps a lot, thanks |
| 17:17.11 | *** join/#asterisk _Sam-- (sam@ns2.kneedraggers.com) |
| 17:17.12 | epoch | nel: I don't think so |
| 17:17.16 | epoch | nel: I don't think it's really the right place for it, either |
| 17:17.20 | epoch | there is a jitter buffer though |
| 17:17.26 | heison | does anyone know if Asterisk is capable of handling realms with SIP? |
| 17:17.28 | _Sam-- | mog you around |
| 17:17.45 | *** join/#asterisk pointer-gaim (~pointer@64.18.101.22) |
| 17:17.49 | _Sam-- | im on hold with digium support ....would be nice to get someone from here :) |
| 17:18.05 | *** join/#asterisk Ro[b]ert (~acidburnn@cust.7.204.adsl.cistron.nl) |
| 17:18.16 | nel | epoch: do you know of any other tool to try to adjust echo cancellation besides ztmonitor? |
| 17:18.23 | Ro[b]ert | i have installed one of them FXO cards you can buy at Ebay... |
| 17:18.43 | Ro[b]ert | now the zaptel.conf file shows fxsks=1 |
| 17:18.54 | Ro[b]ert | doenst that have to be fxoks=1?? |
| 17:19.06 | epoch | nel: I think that's pretty much it... look through echo cancellation stuff in voip-info.org |
| 17:19.20 | epoch | nel: it's a pretty difficult thing to get exactly right, unfortunately |
| 17:19.21 | nel | also when I change things on zapata.conf it always sounds the same, it seems like if I change tx and rx values don't change anything, every time I change something I do ztvcfg and stop and start asterisk , am I doing something wrong? |
| 17:19.33 | epoch | and unfortunately, you'll never get rid of 100% of echo, if you're using an analog interface :( |
| 17:20.11 | nel | epoch: I'm using 4 fxo cards , but in the future we are planning on implementing pris...would I have the same problems with echo? |
| 17:21.01 | Ro[b]ert | the zaptel.conf file shows fxsks=1 |
| 17:21.10 | Ro[b]ert | doenst that have to be fxoks=1?? |
| 17:21.12 | Ro[b]ert | ?? |
| 17:21.57 | nel | Ro[b]ert: I think you have to put the opposite, if you have a fxo card you have to use fxsks |
| 17:22.27 | epoch | nel: we used to have 6 FXOs here, terrible unsolved echo problems... |
| 17:22.33 | epoch | nel: installing PRI fixed 99% of it :) |
| 17:22.44 | nel | epoch, that's good news |
| 17:22.44 | nel | :D |
| 17:22.52 | nel | I'm doing a test here |
| 17:22.55 | epoch | yeah, PRI is awesome |
| 17:22.57 | nel | to buy pri cards |
| 17:23.07 | epoch | *however* we still get some echo when calling certain people |
| 17:23.09 | nel | but it would be a nightmare if i had echo in evry line |
| 17:23.34 | epoch | there's a good document written by Cisco about echo cancellation linked off the wiki |
| 17:23.41 | nel | I can hear myself, but the other people don't hear the echo |
| 17:23.44 | epoch | it's pretty depressing ;) |
| 17:23.47 | epoch | yeah, exactly |
| 17:23.48 | nel | hahahah |
| 17:24.36 | epoch | the echo exists with other PBXes and regular analog phones, but it's amplified when you bring VoIP into the mix |
| 17:25.14 | *** join/#asterisk embo (~embo@ss-dsl-153.southslope.net) |
| 17:25.21 | nel | epoch: if I mess around with zapata.conf, how can I be sure that the changes are been taken by the fxo card? |
| 17:25.40 | Zeeek | nel by doing a restart of asterisk |
| 17:25.52 | nel | I do that but I hear no difference |
| 17:26.02 | nel | i.e if I change the values on tx and rx gain |
| 17:26.03 | embo | Anyone have time to help a newbie pick out hardware for an Asterisk installation for a small call center? |
| 17:26.04 | Zeeek | if you mean zaptel you may have to reload drivers |
| 17:26.15 | nel | it always sounds the same to me even if i use txgain=-40.0 |
| 17:26.27 | *** join/#asterisk dca[laptop] (~dca[lapto@sta-207-174-139-178.rockynet.com) |
| 17:26.34 | Zeeek | then try 100 |
| 17:26.38 | nel | Zeeeek: how do you reload drivers? |
| 17:26.38 | SpaceBass | nel how did you restart? |
| 17:26.57 | nel | Spacebass: stop gracefully, and then asterisk -cvvvvv...and also ztcgf -v |
| 17:27.03 | SpaceBass | cool |
| 17:27.04 | facek_ | hmm |
| 17:27.05 | SpaceBass | just checking |
| 17:27.07 | *** part/#asterisk _Sam-- (sam@ns2.kneedraggers.com) |
| 17:27.09 | Zeeek | or just restart gracefully |
| 17:27.17 | *** part/#asterisk Ro[b]ert (~acidburnn@cust.7.204.adsl.cistron.nl) |
| 17:27.30 | nel | but is ztcfg the right way to restart zaptel drivers? |
| 17:27.39 | epoch | nope |
| 17:27.45 | Zeeek | rmmod zaptel rmmod wcfxs |
| 17:27.49 | epoch | you could rmmod them |
| 17:27.52 | Zeeek | except it isn't wcfxs |
| 17:27.54 | epoch | then modprobe them again |
| 17:27.59 | nel | hmmm I would have to recompile the kernel then |
| 17:28.03 | epoch | Zeeek: yes it is |
| 17:28.07 | epoch | he's using a tdm400p |
| 17:28.09 | epoch | recompile!? |
| 17:28.11 | epoch | why? |
| 17:28.17 | Zeeek | no there's a new driver for FXO |
| 17:28.21 | nel | because the kernel I compiled does not accept module unloading |
| 17:28.26 | epoch | in CVS? |
| 17:28.32 | Zeeek | wctdm or something (but not quite) |
| 17:28.43 | nel | what is wctdm? |
| 17:28.48 | Zeeek | who's on first? |
| 17:28.49 | nel | I'm using zaptel and wcfxs module |
| 17:28.50 | epoch | Zeeek: in CVS or in 1.0.7? |
| 17:28.56 | Zeeek | for the TDM |
| 17:29.10 | Zeeek | ah, I see - ya don't know if it's CVS only |
| 17:29.12 | nel | Zeeek: do I have to modprobe that module? |
| 17:29.14 | *** part/#asterisk embo (~embo@ss-dsl-153.southslope.net) |
| 17:29.20 | epoch | I'm sure that's CVS only |
| 17:29.28 | nel | I always do modprobe zaptel, modprobe wcfxs |
| 17:29.33 | Zeeek | ok, I bow to your superior knowledge :) |
| 17:29.35 | nel | I have a TDM400 with 4 fxo |
| 17:29.52 | SpaceBass | anyone want to help troubleshoot my dial comand? |
| 17:29.55 | epoch | also, why in the hell would you compile a kernel that doesn't allow module unloading? |
| 17:29.55 | SpaceBass | lol |
| 17:29.59 | epoch | that's just ridiculous |
| 17:29.59 | Zeeek | NO WE DON't! |
| 17:30.06 | nel | epoch: lol don't know |
| 17:30.10 | epoch | heh |
| 17:30.11 | nel | I forgot to check that option |
| 17:30.16 | SpaceBass | Zeeek of course you do... come on,it will be fun |
| 17:30.17 | epoch | well I guess you've learned your lesson :) |
| 17:30.23 | Zeeek | ok, ok shoot |
| 17:30.33 | Zeeek | I'm too scared to say anything now |
| 17:30.49 | *** join/#asterisk konrad120_ (~po@o5.pl) |
| 17:30.55 | nel | so, do I need to load any other module besides zaptel and wcfxs? |
| 17:31.00 | Zeeek | no |
| 17:31.25 | SpaceBass | Zeeek working on getting this t option to work... I have (basically) dial(Zap/g0/<called number>/tmr) |
| 17:31.33 | SpaceBass | (through the m in to see if it was working |
| 17:31.43 | Zeeek | simple: no / |
| 17:31.50 | Zeeek | Show Application Dial |
| 17:31.58 | SpaceBass | oops.. I used a pipe |
| 17:32.03 | Zeeek | Dial(Tech/num,time,options) |
| 17:32.21 | SpaceBass | Zeeek thats where I'm confused, what If I dont want a time |
| 17:32.22 | Zeeek | missing the time to ring |
| 17:32.26 | SpaceBass | have to have it? |
| 17:32.37 | Zeeek | I guess at least ,, or || |
| 17:32.41 | Zeeek | just a guess |
| 17:32.44 | SpaceBass | will try ,, |
| 17:32.53 | Zeeek | with no options tho, no you don't need a time |
| 17:33.03 | AgiNamu | Hey, im looking for a full solution that can set up someone to be a "Vonage" like compnay |
| 17:33.10 | Zeeek | as in ZAP/1 |
| 17:33.11 | AgiNamu | anyone here know someone who's offering a decent system like that? |
| 17:33.15 | *** join/#asterisk husher (~andrew@68.143.92.130.nw.nuvox.net) |
| 17:33.17 | *** join/#asterisk MattH (~matth@noc-wireless.chilitech.net) |
| 17:33.48 | MattH | Hello... does anyone know... has anyone written an implimentation to do BUSY CALL BACK? |
| 17:34.39 | Zeeek | as in dial 1 to call back? |
| 17:35.02 | Zeeek | or timed? |
| 17:35.13 | MattH | no.. as in.. <dial number> beep beep beep beep (line is busy).. hang up and dial *xx and the system will keep trying the number... when it rings... it will call you back at your extension and wala.. you are conected with the party who was busy |
| 17:35.32 | Zeeek | it wouldbe fairly easy to do |
| 17:35.47 | Zeeek | especially with a .call file |
| 17:35.57 | Zeeek | which has retries and delay between parameters built in |
| 17:36.03 | MattH | hrmm true |
| 17:36.17 | MattH | where would you pull the previous dialed number from? |
| 17:36.24 | SpaceBass | Zeeek your the man! |
| 17:36.28 | Zeeek | you'd have to save it in every dial |
| 17:36.43 | Zeeek | SpaceBass I try to solve one problem a day |
| 17:37.08 | SpaceBass | Zeeek using asterisk@home which uses AMP which makes everything a pain in the ass... hard to tell what macro is doing what |
| 17:37.08 | Zeeek | MattH so in the dial command, you'd stick the number in the db |
| 17:37.10 | *** join/#asterisk _Sam-- (sam@ns2.kneedraggers.com) |
| 17:37.18 | MattH | yeah... I guess that wouldn't be horribly hard to do |
| 17:37.25 | Zeeek | SpaceBass which is why I hate the idea of all those gadgets |
| 17:37.31 | SpaceBass | Zeeek but i have a dial-out macro that seems to work for all outgoing calls |
| 17:37.33 | nel | I'm using AMP lately, I don't know if I like it |
| 17:37.37 | SpaceBass | Zeeek beginging to feel the same way |
| 17:37.48 | nel | is there something in the middle where you could get to the config files and change things by yourself? |
| 17:37.49 | Zeeek | MattH trivial as long as you can stick the stuff in a dial macro without ripping apart the entire dialplan |
| 17:37.59 | _Sam-- | hi just wondering if anyone has any advice...have a pri trying to set outgoing caller id using exten => _9XXXXXXX.,2,SetCIDNum(302xxxxxxx); i see it being set on CLI, but phone company says their side is not seeing it |
| 17:38.00 | Zeeek | nel yes it's called make |
| 17:38.13 | _Sam-- | they setup a d channel monitor |
| 17:38.17 | ^HeLL^ | bye people. :) |
| 17:38.20 | _Sam-- | our side shows it sending, at least it looks like |
| 17:38.22 | nel | make install |
| 17:38.28 | *** part/#asterisk mass_murderer (murderer@200-171-226-51.customer.telesp.net.br) |
| 17:38.32 | MattH | what format is asterik's database? |
| 17:38.48 | MattH | is it it's own internal? |
| 17:38.49 | Zeeek | dunno |
| 17:38.51 | Zeeek | ya |
| 17:38.54 | nestAr | man.. now i need a test bed |
| 17:38.59 | Zeeek | just four simple commands |
| 17:39.00 | MattH | Zeeek: yeah... perhaps I shall try that |
| 17:39.03 | nestAr | i keep wanting to make changes... damn users. |
| 17:39.23 | *** join/#asterisk kraeMit (~chatzilla@p54890C22.dip0.t-ipconnect.de) |
| 17:39.28 | Zeeek | MattH I always save the last incoming callerid in the db |
| 17:39.31 | nel | Zeek: are you kidding me? |
| 17:39.43 | Zeeek | nel no, you can tweak all you want |
| 17:39.44 | nel | Zeeek |
| 17:39.58 | nel | yes I know, but I need a gui interface for the telephone guy |
| 17:40.06 | nel | I prefer the text files too |
| 17:40.15 | SpaceBass | Zeeek with the # thing working, could I use that do to some kind of soft "flash" for call waiting? |
| 17:40.34 | Zeeek | SpaceBass I'm not sure but look up Flash and see what it does |
| 17:40.54 | Zeeek | in fact, show applications, print that and look at the list - there's some interesting shit there! |
| 17:41.01 | Zeeek | like "Flash" |
| 17:41.13 | SpaceBass | never knew that command in the cli... its great, thanks |
| 17:41.41 | Zeeek | funing gui bullshit interfaces |
| 17:41.53 | SpaceBass | HA! |
| 17:42.08 | Zeeek | SpaceBass of course the # will stop you from ever using # on a call |
| 17:42.17 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
| 17:42.17 | Zeeek | i.e., talking to your bank |
| 17:42.28 | SpaceBass | Zeeek yeah, thought about that |
| 17:42.31 | Zeeek | "please enter your ss number ending with the pound key" |
| 17:42.41 | Zeeek | so I limlit it to incoming |
| 17:42.47 | SpaceBass | Zeeek but I dont call my bank... they do tend to call me though.... hummm |
| 17:43.01 | SpaceBass | Zeeek how do you limit it? |
| 17:43.02 | Zeeek | anyway, why doesn('t the softwarez know ss numbers are fixed length? |
| 17:43.36 | Zeeek | if your bank calls you and you need to input #, you'll have to divert them using callerid (which they prolly have turned off) |
| 17:43.49 | SpaceBass | Zeeek I've never understood that about any IVR prompt that asks for fixed lenght info enter your 4 ddigit pen followed by the # key |
| 17:43.59 | Zeeek | better still: |
| 17:44.00 | _Sam-- | thats another pet peeve of mine...on ecommerce sites, why do they ask what type of card you have <visa,mc,ae,etc>...its obvious by the first digit |
| 17:44.08 | SpaceBass | Zeeek how do you limit the t switch to incoming calls only? |
| 17:44.15 | Zeeek | why do I enter it, then when the op answers she askls me again? |
| 17:44.29 | SpaceBass | I HATE THAT! |
| 17:44.30 | Zeeek | SpaceBass read the application description |
| 17:44.32 | SpaceBass | HATE IT |
| 17:44.40 | Zeeek | it's ten lines long |
| 17:44.46 | SpaceBass | makes me see red... "hello what is your account number" I JUST ENTERED IT! |
| 17:44.46 | Zeeek | and has all the answers |
| 17:45.36 | jontow | weeeiiirdd.. my cdr_mysql stopped logging uniqueids |
| 17:45.41 | *** join/#asterisk cyberkoa (~cyberkoa@65.90.93.3) |
| 17:46.24 | cyberkoa | sorry just now disconnected , anybody heard "bellster" or fwdOut b4 ? how do they configure the Asterisk to do that ? |
| 17:46.39 | Zeeek | see their site |
| 17:47.39 | *** join/#asterisk r0d3nt|m (nobody@wsip-24-234-241-84.lv.lv.cox.net) |
| 17:48.23 | cyberkoa | yes , I hv browsed to their site , but wht I hope to know is , how can I do the same thing as them , but I just need to connect my Asterisk box to another asterisk box to call out |
| 17:48.47 | *** join/#asterisk Egonis (~chultay@69.194.211.129) |
| 17:49.21 | cyberkoa | i mean both asterisk box are mine , but at 2 different places |
| 17:54.04 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@l01m-20-133.d4.club-internet.fr) |
| 17:54.59 | *** join/#asterisk ejakupov (~ejakupov@zektor.gpcc.itd.umich.edu) |
| 17:55.28 | ejakupov | Hi all, I have a quick question ,how do I setup asterisk to route incoming calls to SIP extensions, I'm using asterisk@home with broadvoice |
| 17:55.33 | ejakupov | any docs I can be refered too? |
| 17:55.48 | Zeeek | Starter tutorial: |
| 17:55.48 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html |
| 17:55.48 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html |
| 17:55.48 | Zeeek | http://www.automated.it/guidetoasterisk.htm |
| 17:55.48 | Zeeek | THE reference of the moment: |
| 17:55.49 | Zeeek | http://www.asteriskdocs.org |
| 17:56.05 | marlowe | If you cant get ASterisk@home to work... wow. |
| 17:56.17 | ejakupov | I got it to work |
| 17:56.29 | marlowe | Doesnt it work out of the box? |
| 17:56.33 | marlowe | CD, installation |
| 17:56.33 | marlowe | rather |
| 17:56.37 | ejakupov | It's just when I call the broadvoice number it does not route the call to my sip phone |
| 17:56.46 | ejakupov | marlowe: it does work out of the box |
| 17:57.11 | marlowe | Well I dont support Asterisk@home - Sorry. |
| 17:57.17 | marlowe | I support asterisk only. |
| 17:57.28 | ejakupov | well it's asterisk |
| 17:57.40 | *** part/#asterisk _Sam-- (sam@ns2.kneedraggers.com) |
| 17:57.42 | Zeeek | there is an @home mailing list |
| 17:57.55 | ejakupov | <PROTECTED> |
| 17:57.55 | ejakupov | <PROTECTED> |
| 17:58.01 | Zeeek | and bv has many issues (see asterisk-users) |
| 17:58.05 | ejakupov | that's what I get, my extension is 301 |
| 17:58.36 | marlowe | Why the hell does Asterisk@Home use an AGI script to dial a sip extension |
| 17:58.42 | marlowe | Who made that crap anyhow |
| 17:58.55 | ejakupov | man the config file is freaking huge and complicated |
| 17:59.09 | ejakupov | but it was easy to get going and me being a novice it seemed like a good idea |
| 17:59.13 | marlowe | I find asterisk the easiest program I've ever had to learn to install and configure |
| 17:59.19 | marlowe | W/o using @Home |
| 17:59.20 | Zeeek | LOL |
| 17:59.31 | marlowe | I think more people have more problems with Asterisk@Home then Asterisk |
| 17:59.52 | bonez39 | !lastseen shido6 |
| 18:00.10 | marlowe | shido6 was last seen on my bed 2h 1m 3s ago. |
| 18:00.17 | bonez39 | on your bed, eh? |
| 18:00.22 | ejakupov | marlowe: can you at least point me to files I might need to check to get this working? |
| 18:00.29 | marlowe | ejakupov: extensions.conf |
| 18:00.31 | marlowe | ejakupov: sip.conf |
| 18:00.45 | marlowe | It |
| 18:00.57 | marlowe | It's using an AGI script - For whatever reason (dumb I think)... and I dont know what that script dodes. |
| 18:01.05 | marlowe | And I'm not going to look at it either |
| 18:01.13 | ejakupov | I understand |
| 18:01.35 | marlowe | Asterisk itself works out of the box |
| 18:01.39 | marlowe | WWhy do people need @Home |
| 18:01.52 | ejakupov | no idea :) |
| 18:01.59 | ejakupov | is this an AGI script stuff exten => s,4,SetVar(intype=${INCOMING}) |
| 18:01.59 | ejakupov | exten => s,5,Cut(intype=intype,-,1) |
| 18:02.16 | Egonis | in the 10 minute guide, it says that asterisk will do a demo on ext1000, where are the sound files? the install of asterisk didn't look that big! |
| 18:02.16 | SpaceBass | marlowe I'm using it... for a newbie it a starting place... i could make changes in AMP and see what they were doing by looking at the conf filesw |
| 18:02.33 | SpaceBass | now, I feel like amp is a pain in the ass and causes more problems than it fixes |
| 18:02.47 | marlowe | I agree with you fully on the second statement |
| 18:02.51 | SpaceBass | : |
| 18:02.52 | SpaceBass | "_ |
| 18:02.56 | marlowe | ejakupov: no |
| 18:02.59 | SpaceBass | well, i cannot type today |
| 18:03.24 | marlowe | Agi would be s,1,AGI(...) |
| 18:03.25 | marlowe | :) |
| 18:03.31 | marlowe | and omfg |
| 18:03.41 | marlowe | I'm on server # 38 out of 297 |
| 18:03.46 | marlowe | to install Service Pack 1 |
| 18:03.48 | marlowe | for Server 2003 |
| 18:03.53 | marlowe | Im dying |
| 18:03.56 | ejakupov | marlowe use SUS |
| 18:04.02 | ejakupov | let them update themselves |
| 18:04.02 | Zeeek | didn't they just release #2? |
| 18:04.02 | marlowe | I dont want to |
| 18:04.07 | marlowe | I do it myself |
| 18:04.12 | ejakupov | well good luck :) |
| 18:04.13 | marlowe | SUS has been evil in the past |
| 18:04.17 | MattH | marlowe: shouldn't you be doing that at night? |
| 18:04.19 | jakepdev | he's a union emyloyee |
| 18:04.20 | marlowe | I do everything manually |
| 18:04.21 | ejakupov | It works really good now |
| 18:04.33 | marlowe | MattH: I'm doing the non important servers now.. |
| 18:04.34 | SpaceBass | marlowe i truly feel for you |
| 18:04.40 | ejakupov | and that WSUS thing is even better |
| 18:04.53 | MattH | :) |
| 18:04.58 | marlowe | Main site, exchange server, etc. |
| 18:04.58 | *** join/#asterisk mmostafa (~mmostafa@81.10.62.101) |
| 18:05.01 | marlowe | Will be done @ 3am |
| 18:05.09 | marlowe | Shit, I dont care about my customers |
| 18:05.14 | marlowe | I are enough to do it manually for them |
| 18:05.17 | marlowe | s/are/care |
| 18:06.11 | marlowe | Anywayyyy |
| 18:06.25 | marlowe | Someone talk |
| 18:06.40 | denon | no |
| 18:06.51 | Zeeek | Why would an IAX call disconect afetr the exact delay given in dial when the person answered? |
| 18:06.51 | marlowe | please? |
| 18:07.01 | Zeeek | ZAP->IAX2 |
| 18:07.22 | Zeeek | Dial(IAX2/someone,15,t) |
| 18:07.24 | marlowe | Zeeek: As in you set yuour timeout to 30s and you picked up exactly at 30s? |
| 18:07.31 | Zeeek | ringing.... |
| 18:07.34 | *** join/#asterisk anonymous^ (~strace@ADSL-F49-S197-critical-coi.nortenet.pt) |
| 18:07.37 | Zeeek | codec is ulaw... blah |
| 18:07.40 | Zeeek | "hello" |
| 18:07.41 | anonymous^ | how do I let sip clients register anonymously? |
| 18:07.44 | marlowe | Or you pick up after 2s, and it disconnects @ 15s? |
| 18:07.54 | Zeeek | we talk and at exactly 15 the dial goes to "the person at..." |
| 18:08.05 | marlowe | haha that's great |
| 18:08.13 | marlowe | I've never seen that one before |
| 18:08.18 | Zeeek | pickup and talking at say, 5 secs - this happened ONLY from ZAP |
| 18:08.26 | marlowe | hmm |
| 18:08.27 | Zeeek | yeah it's kinda funny |
| 18:08.38 | anonymous^ | funky more like |
| 18:08.38 | anonymous^ | :> |
| 18:08.40 | Zeeek | then I had him call me at same ZAP and it didn't happen |
| 18:08.40 | marlowe | Something isn't acknowledging the call was picked up |
| 18:08.49 | marlowe | Even though it was |
| 18:08.52 | marlowe | Even though you can talk |
| 18:08.53 | Zeeek | could a codec thing do this? |
| 18:08.58 | marlowe | So when it hits 15s it goes to VM |
| 18:08.59 | marlowe | It could |
| 18:09.07 | marlowe | I've seen a codec problem cause stranger issues |
| 18:09.11 | Zeeek | cause first I see codec is ulaw |
| 18:09.15 | Zeeek | (from ZAP) |
| 18:09.23 | *** join/#asterisk LoRez_ (lorez@lorez.staff.freenode) |
| 18:09.27 | Zeeek | but when we spoke, codec was actually iLBC |
| 18:09.27 | marlowe | Im gonna barf |
| 18:09.37 | marlowe | Why you using iLBC? |
| 18:09.37 | Zeeek | "please format c: ?" |
| 18:09.40 | inspired | does anyone know an open source e-commerce suite for PostgreSQL? |
| 18:09.43 | Zeeek | his client was I assume |
| 18:09.54 | Zeeek | since I'm on ZAP I didn't pay attention |
| 18:10.00 | Zeeek | but that is my only clue |
| 18:10.02 | marlowe | Oh true that might be the problem |
| 18:10.04 | ejakupov | well this does work out of the box, my phone was off :( |
| 18:10.18 | ejakupov | sorry |
| 18:10.23 | Zeeek | the iron won't work if it ain't plugged in sweetheart :) |
| 18:10.31 | marlowe | You see - You figured it out all yourself.. So that's good |
| 18:10.38 | ejakupov | I need to be hit over the head |
| 18:10.48 | Zeeek | My brother did that to me wuith his new scanner: 30 minutes of USB debugging before he realized it was off |
| 18:10.50 | ejakupov | thank you |
| 18:11.04 | Zeeek | "oh, shit, wait a sec..." |
| 18:11.11 | anonymous^ | how do I let sip clients register anonymously? |
| 18:11.18 | marlowe | anonymous^: you asked once |
| 18:11.23 | ejakupov | I usually check the cables first, but it's been long day |
| 18:11.25 | marlowe | anonymous^: don't ask twice in such a short period of time |
| 18:11.31 | Zeeek | or read this: |
| 18:11.32 | machinehd | I have some phones all on the same switch that do not work when calling EX to EX. The phones ring, but there's no sound on either side. Besides codec problems, what else would cause this? |
| 18:11.32 | Zeeek | Starter tutorial: |
| 18:11.32 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html |
| 18:11.32 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html |
| 18:11.32 | Zeeek | http://www.automated.it/guidetoasterisk.htm |
| 18:11.33 | Zeeek | THE reference of the moment: |
| 18:11.33 | Zeeek | http://www.asteriskdocs.org |
| 18:11.37 | Zeeek | First article tells |
| 18:11.50 | marlowe | Zeeek: You should uhh make that less of a paste |
| 18:11.53 | nestAr | man, i've gotten spoiled compiling on my Xeon |
| 18:12.01 | nestAr | this test box is setup is sloooowwwww |
| 18:12.04 | Zeeek | not that busy atm |
| 18:12.12 | marlowe | nestAr: all my machines are now Dual Xeon's |
| 18:12.15 | marlowe | Except my workstation |
| 18:12.18 | marlowe | But |
| 18:12.25 | marlowe | In 2 weeks |
| 18:12.31 | marlowe | Nice Dual Xeon 2.8ghz workstation |
| 18:12.32 | marlowe | hell yeah |
| 18:12.35 | Egonis | nestAr: how quick does yours compile? mine takes 28 seconds |
| 18:12.39 | ejakupov | how about this dual xeon 1.5GHz vs P4 3.0Ghz, which one should I run as workstation |
| 18:12.42 | *** join/#asterisk toddf (~toddf@adsl-66-141-66-223.dsl.okcyok.swbell.net) |
| 18:12.46 | marlowe | Dual Xeon 1.5 |
| 18:12.57 | Egonis | marlowe: Dual Xeon 2.8 |
| 18:13.04 | *** join/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk) |
| 18:13.04 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 18:13.05 | marlowe | Egonis: i was talking to ejakupov |
| 18:13.20 | marlowe | Hey Brian |
| 18:13.35 | Egonis | marlowe: Sorry... lol |
| 18:13.44 | ejakupov | thank you everyone for help, time to get back to work |
| 18:13.44 | marlowe | 'its ok :) |
| 18:13.51 | marlowe | ejakupov: Go work and do some SUS |
| 18:14.01 | *** join/#asterisk cyberkoa (~cyberkoa@65.90.93.3) |
| 18:14.02 | ejakupov | marlowe: it's done for me :) |
| 18:14.06 | nestAr | Egonis: i dunno.. i didn't count.. fast enough i don't worry about it.. |
| 18:14.31 | cyberkoa | sorry , i got disconnected again , anybody already answered my question ? |
| 18:14.34 | nestAr | EWWW... the cvs version won't build. |
| 18:15.17 | marlowe | cypromis: who are you? what was the question |
| 18:15.37 | nestAr | anyone built CVS-HEAD today? |
| 18:15.53 | marlowe | I got some head |
| 18:15.57 | nestAr | lol |
| 18:16.00 | marlowe | i did yesterday |
| 18:16.01 | marlowe | not todays |
| 18:16.03 | nestAr | walked into that one |
| 18:16.30 | nestAr | this build bombs when it gets to app_queue |
| 18:18.17 | *** join/#asterisk W1thdraw (~Withdraw@ip70-181-96-254.oc.oc.cox.net) |
| 18:18.21 | VirTERM | anyone using realtime with iax trunks/peers? |
| 18:19.46 | *** join/#asterisk KristinH (~KristinH@muppet.geekgirls.us) |
| 18:20.18 | KristinH | hi. a quick question for anyone that can answer |
| 18:20.52 | KristinH | I set up a iax type=friend and I can get the two asterisk servers to talk |
| 18:21.08 | *** join/#asterisk cjk (~cjk@80.92.75.232) |
| 18:21.21 | *** join/#asterisk _GiGi_ (gigi@jabber.szczecin.pl) |
| 18:21.26 | KristinH | i am passing 333101 in the dialstring |
| 18:21.35 | _GiGi_ | hello |
| 18:21.49 | machinehd | do you really need ports 10000:20000 opened on the box? or can that be slimmed down? Currently there's only 5 calls max happening at any one time. |
| 18:21.56 | _GiGi_ | asterisk have chan_skype ? for connect to skype ? :) |
| 18:22.36 | KristinH | since there will be different extensions than just 333101, what would I wantto yse on the extensions side of the remote server |
| 18:24.28 | *** join/#asterisk topping (~topping@cpe-24-210-82-196.columbus.res.rr.com) |
| 18:24.43 | KristinH | or better yet, how can i enable dumping the progress of extensions processing |
| 18:24.48 | *** part/#asterisk Egonis (~chultay@69.194.211.129) |
| 18:25.24 | MattH | Is there any kind of documentation (I have the asterisk documentation) to walk a newish user through the setup (I have the file telling how to get from the cvs).. and get say an FXO card to work? |
| 18:25.31 | Zeeek | machinehd I use 10000-10100 |
| 18:25.37 | bkw_ | HOLY SHIT .. I feel like i'm in Iraq |
| 18:25.44 | anonymous^ | how do I let sip clients register anonymously? |
| 18:26.34 | Zeeek | MattH ever seen this? http://www.automated.it/guidetoasterisk.htm |
| 18:26.40 | Zeeek | it does all that |
| 18:26.54 | bkw_ | anonymous^, you don't |
| 18:26.56 | bkw_ | oh wait |
| 18:27.00 | bkw_ | autocreatepeer=yes |
| 18:27.08 | MattH | Zeeek: there we go :) |
| 18:27.12 | anonymous^ | bkw_: on sip.conf? |
| 18:27.14 | Zeeek | MattH and http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html too |
| 18:27.20 | bkw_ | anonymous^, check the sip.conf.sample |
| 18:27.37 | anonymous^ | bkw_: thanks |
| 18:27.40 | Zeeek | bkw_ can't you use the default context to accept calls? |
| 18:28.03 | Zeeek | oh register to call them... |
| 18:28.06 | Zeeek | duh |
| 18:28.10 | *** join/#asterisk Beirdo (~gjhurlbu@beirdo.user) |
| 18:32.41 | denon | ~seen jerjer |
| 18:32.45 | jbot | jerjer <~JerJer@dsl-106-170.che.centurytel.net> was last seen on IRC in channel #asterisk, 32d 2h 13m 50s ago, saying: 'mrgoby: sure'. |
| 18:32.45 | bkw_ | Zeeek, :P |
| 18:32.47 | denon | ~jerjer |
| 18:32.48 | jbot | it has been said that jerjer is the guy who runs nufone |
| 18:32.48 | denon | ah |
| 18:32.52 | denon | ~seen jerjer[mobile] |
| 18:32.53 | jbot | jerjer[mobile] <~nonyobizn@65.173.197.174> was last seen on IRC in channel #asterisk, 2h 19m 53s ago, saying: 'then override the internal with SetCIDNum for sendng out to the pstn'. |
| 18:34.18 | jakepdev | I'm looking for a minimal linux install (NO GUI) that works well with *. Suggestions? |
| 18:35.16 | *** join/#asterisk blitzrage (~blitzrage@blitzrage.documenter.extraordinaire.of.asterisk) |
| 18:35.19 | *** join/#asterisk ManxPower (~eric@dsl-209-16-67-160.datasync.com) |
| 18:36.26 | bjohnson | jakepdev: debian, gentoo, centos, etc, etc, etc |
| 18:36.50 | ManxPower | MattH: BellSouth is blaming the loss of dialtone and DSL problem on "inside wireing", but were absolutily NOT interested in testing the line after I disconnected the entire house from the line at the dmark |
| 18:37.01 | jakepdev | bj - debian is 2 ISOs |
| 18:37.02 | MattH | lol |
| 18:37.19 | bjohnson | jakepdev: not if you install over net |
| 18:37.35 | jakepdev | ok |
| 18:37.46 | bjohnson | or use one of the minimal deb install systems |
| 18:38.09 | bjohnson | btw .. you didn't specify the size of the install image in your request |
| 18:39.02 | jakepdev | bj - found the minimal instll network - that'll work - tnx |
| 18:39.44 | bjohnson | if you want really small you could look at openwrt |
| 18:39.57 | jakepdev | has anyone tried it with *? |
| 18:40.30 | jakepdev | k - lol - I found it |
| 18:40.53 | bjohnson | jerjer runs it I beleive |
| 18:41.04 | jakepdev | that's pretty amazing |
| 18:41.06 | denon | it can be done, yes |
| 18:41.09 | denon | its not pretty :) |
| 18:41.14 | bjohnson | I intend to but ran into some hardware version problems |
| 18:41.37 | bjohnson | well .. I've been told up to 4 concurrent gsm iax sessions have run on it |
| 18:42.02 | denon | gsm even.. |
| 18:42.09 | denon | wonder how many ulaw |
| 18:42.27 | ManxPower | MattH: The repair person got frustrated because I didn't know any local phone numbers. |
| 18:42.39 | bjohnson | it isn't appropriate for business use but it suitable for home without vm |
| 18:43.00 | ManxPower | The only time I make a local call is to order pizza or a cab |
| 18:43.00 | bjohnson | ManxPower: didn't they? |
| 18:43.27 | ManxPower | bjohnson: She sounded like she thought I was a moron when I asked her what number she wanted me to call. |
| 18:43.40 | ManxPower | Of course, my dialtone AND DSL suddenly, macially started working. |
| 18:44.00 | *** join/#asterisk Shido6 (~greg@d57-87-253.home.cgocable.net) |
| 18:44.13 | ManxPower | I had to go to the 'frige to get a pizza advertizement. |
| 18:44.13 | Zeeek | it's called "provider in denial" |
| 18:44.22 | bjohnson | first line of defence |
| 18:44.30 | Shido6 | stupid wind |
| 18:44.31 | *** join/#asterisk Achillies (~nick@165.252.43.77) |
| 18:44.34 | *** join/#asterisk PoWeRKiLL (~PoWeRKiLL@l01m-20-133.d4.club-internet.fr) |
| 18:44.36 | MattH | ManxPower: ha! I've seen repair people get frustraited! |
| 18:44.41 | Zeeek | I once returned a light bulb that blew out when turned on |
| 18:44.47 | MattH | ManxPower: I have voip at home over a wireless link... I asked to have the d-mark removed from my house |
| 18:44.48 | ManxPower | what really gets me is when I offered to disconnect the house AT THE DEMARC they refused to test the line again. |
| 18:44.51 | Zeeek | the guy said "it's your power installation"! |
| 18:44.55 | MattH | ManxPower: "what is the number there?" me: "ummm thre is none" |
| 18:45.04 | Achillies | can anyone help me with the defualt password for Asterisk@Home |
| 18:45.09 | MattH | ManxPower: "so you want to order service?" me: uhhh no |
| 18:45.13 | MattH | Achillies: which one? |
| 18:45.18 | nestAr | lol @ ManxPower and the HTML procmail filter |
| 18:45.23 | ManxPower | ~google site:lists.digium.com default password home |
| 18:45.23 | Achillies | web admin |
| 18:45.32 | ManxPower | nestAr: *grin* |
| 18:45.43 | MattH | Achillies: u:maint p:password |
| 18:46.03 | Achillies | ok the manual is wrong then ;) |
| 18:46.07 | Bacon | nestAr: Howdy. |
| 18:46.13 | MattH | Achillies: eh? you mean the maintence password right? |
| 18:46.14 | KristinH | how can i enable dumping the progress of extensions processing |
| 18:46.24 | ManxPower | MattH: My DSL ISP is the same as my largest customer. My largest customer is the ISP/TELCO's second largest customer. When I call for help they do ANYTHING for me. |
| 18:46.29 | Achillies | Enter the web interface (user: admin, pass: password) |
| 18:46.42 | ManxPower | But they don't offer residential service |
| 18:47.06 | Achillies | MattH: that worked thank you |
| 18:47.33 | ManxPower | nestAr: imagine of 50 or 100 people set up that filter and automated message. |
| 18:48.13 | `Sauron | Manx: what filter? |
| 18:48.54 | nestAr | ManxPower: haha, yeah, the html users might get the point if they were flooded |
| 18:49.00 | nestAr | hi Bacon |
| 18:49.03 | nestAr | long time no chat |
| 18:50.09 | ManxPower | nestAr: exactly |
| 18:50.25 | ManxPower | nestAr: The best way would be for the mailing list software to just reject the messages |
| 18:50.28 | Bacon | nestAr: Rolled out to what end, just internal phone system? |
| 18:50.53 | _GiGi_ | asterisk have chan_skype ? for connect to skype ? :) |
| 18:51.10 | Shido6 | heeh |
| 18:51.21 | ManxPower | Shido6: do you know how to use patlooptest? |
| 18:51.44 | Shido6 | that sounds like a vaginal medical test |
| 18:51.49 | Shido6 | not sure I wanna know what that is |
| 18:53.03 | ManxPower | Shido6: tests PCI-Digium interface for corrupted data |
| 18:53.16 | ManxPower | mark told me to try it, but didn't give any instructions. |
| 18:53.36 | *** join/#asterisk mkhan (~mkhan@ip66-105-190-122.z190-105-66.customer.algx.net) |
| 18:53.40 | mkhan | hello... |
| 18:53.53 | *** join/#asterisk KristinH (~KristinH@muppet.geekgirls.us) |
| 18:54.02 | mkhan | I have a one cents question !!.. is * a PSTN gateway? |
| 18:54.09 | *** part/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
| 18:54.25 | Nugget | it can be. |
| 18:55.02 | KristinH | i have a stupid qursion.. is this the only irc #asterisk group are there any other networks |
| 18:55.02 | Shido6 | hold on |
| 18:55.07 | Shido6 | let me get you a visual |
| 18:55.09 | sivana | lol |
| 18:55.15 | sivana | ppl are amazing |
| 18:55.37 | *** join/#asterisk tessier (~treed@210.245.97.125) |
| 18:55.54 | ariel_ | KristinH, there is #asterisk-stable the #asterisk-dev and if your using Amp there is #amportal |
| 18:55.54 | sivana | I love how open source ppl want it both ways |
| 18:56.11 | Shido6 | http://shido.gotdns.org/takealook313/Presentation/img6.html |
| 18:56.46 | ManxPower | Do people actually have seperate servers for M0H |
| 18:56.52 | Wonka | yeah. i want the newest devel snapshot, but it should be stable, dammit! ;) |
| 18:56.54 | KristinH | ariel I specifically an #asterisk cchannel, say on efnet or somethhing? |
| 18:56.55 | Shido6 | its a rough colorful example of digiums original presentation |
| 18:56.58 | sivana | KristinH: There may be channels on other network, but this is the official one |
| 18:57.16 | Wonka | _and_ mISDN, chan_misdn and chan_capi must compile and run with it, too |
| 18:57.18 | sivana | KristinH: * developers and core supporters are here |
| 18:58.24 | ManxPower | KristinH: If there is, there very few people will be there. |
| 18:58.24 | ariel_ | KristinH, here is where most of the people come anyway. |
| 18:59.15 | KristinH | ariel_, thank you |
| 18:59.16 | *** join/#asterisk Defraz (~t0tal@dhcp161-pool2.wireless.fiber.dcdi.net) |
| 18:59.36 | KristinH | i guess i'll take a number and wait for my turn |
| 18:59.45 | sivana | KristinH: what's up? |
| 19:00.07 | KristinH | I just want to watch the debug of a call |
| 19:00.19 | sivana | sip call? |
| 19:00.20 | KristinH | how extensions is processed |
| 19:00.27 | KristinH | iax2 |
| 19:01.00 | *** part/#asterisk mkhan (~mkhan@ip66-105-190-122.z190-105-66.customer.algx.net) |
| 19:01.33 | sivana | iax2 set debug on ? |
| 19:01.34 | KristinH | i have the fun task of building a network to support about 10,000 uesers |
| 19:01.52 | ManxPower | KristinH: "fun" is not the first word that comes to mind. |
| 19:01.57 | KristinH | and the powers that be want it done strictly in asterisk |
| 19:02.15 | KristinH | no kidding |
| 19:02.40 | ManxPower | I'll be doing a 75 phone or so install in a week or so |
| 19:03.07 | Hmmhesays | how does one make asterisk support that many users |
| 19:03.09 | KristinH | so all I am trying to do now is figure out what is passed during a call setup |
| 19:03.13 | ManxPower | I'm just finishing up a 12 phone + PRI install |
| 19:03.20 | denon | Hmmhesays: lots of iron |
| 19:03.25 | KristinH | ie, if i dial 12345 |
| 19:03.28 | ManxPower | KristinH: Put in a development/test server. Have a few people use it. |
| 19:03.51 | marlowe | |WoLf| |
| 19:04.12 | KristinH | what is the way to use variables in extensions so I do not have 10,000 bloddy extensions that all do the same thing |
| 19:04.24 | Shido6 | macro's |
| 19:04.37 | Shido6 | or dialplan logic |
| 19:04.42 | KristinH | i'm usinng realtime config |
| 19:04.47 | ManxPower | KristinH: you'll need 10,000 exten lines. |
| 19:04.48 | nestAr | macros = win |
| 19:05.02 | nestAr | but instead of 30,000 lines, only 10,000 |
| 19:05.09 | nestAr | <3 macros |
| 19:05.19 | Shido6 | no |
| 19:05.47 | nestAr | or patern match |
| 19:05.56 | ManxPower | Users will want things that are custom to that user. |
| 19:05.59 | KristinH | it seems to me that the right way is to have the phones register with SER and then use asterisk for class five features |
| 19:06.04 | *** join/#asterisk ikey (ikey@220.226.40.182) |
| 19:06.05 | nestAr | Users get nothing! |
| 19:06.09 | ManxPower | Like forward to cell phone on no answer or something like that. |
| 19:06.10 | nestAr | and they like it! |
| 19:06.18 | ikey | did any one have experiance on |
| 19:06.24 | ikey | MFC r2 with asterisk |
| 19:06.39 | nestAr | i want the Nugget-style bluetooth prox forwarding |
| 19:06.47 | nestAr | but i have no bluetooth cell phone |
| 19:07.03 | *** join/#asterisk jets (~brian@guardian.pmt.org) |
| 19:07.11 | jets | I wonder if anyone is in the support queue at digium :) |
| 19:07.46 | KristinH | is that probably the case about using ser + asterisk? |
| 19:08.42 | Shido6 | heh |
| 19:08.44 | Shido6 | whats up jets? |
| 19:10.16 | jets | Hellllllo |
| 19:10.19 | `Sauron | ... when you're a jet, you're a jet all the way ... |
| 19:10.46 | ManxPower | The people that want to move the mailing lists to a web forum need to be bitchsmacked. |
| 19:11.04 | `Sauron | yes |
| 19:11.07 | `Sauron | SMACK |
| 19:11.19 | Nugget | I concur |
| 19:11.29 | ManxPower | I'm just happy that it will never happen |
| 19:11.38 | *** join/#asterisk Moc____ (~mochouina@h66-201-214-109.gtconnect.net) |
| 19:11.44 | Nugget | web forums are the cesspools of the internet. |
| 19:12.01 | `Sauron | and they're busy-work |
| 19:12.05 | `Sauron | as in you have to poll them |
| 19:12.10 | ManxPower | Nugget: Well nobody would use the forum that actually KNOWS anything |
| 19:12.15 | `Sauron | instead of event driven (the email just shows up) |
| 19:12.16 | Nugget | yep |
| 19:12.27 | KristinH | are there any variables that get setup on a call tthat can be used in call plans |
| 19:12.29 | chap | heheheheh |
| 19:12.50 | ManxPower | KristinH: README.variables |
| 19:12.50 | ManxPower | ~docs |
| 19:12.51 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 19:12.51 | ManxPower | ~mailinglist |
| 19:12.52 | jbot | rumour has it, mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ |
| 19:12.52 | KristinH | k |
| 19:12.59 | Nugget | but think how great it would be to be able to photoshop up a nice 512 x 128 gradient graphic signature gif that detailed the scope of your asterisk network. |
| 19:13.09 | jets | OMG i am STILL on hold |
| 19:13.20 | Nugget | animated gif. |
| 19:13.24 | *** join/#asterisk W1thdraw (~Withdraw@ip70-181-96-254.oc.oc.cox.net) |
| 19:13.27 | Shido6 | "/usr/src/asterisk/configs" "/usr/src/asterisk/doc/" |
| 19:13.28 | ManxPower | jets: be sure to time your time on hold and then post the info to the mailing lists. |
| 19:13.39 | sivana | nothing like trying to get to Ford's IT dept... 3 phone numbers later |
| 19:13.47 | Shido6 | HAHA |
| 19:13.53 | Shido6 | I used to work for Ford's IT dept |
| 19:13.58 | *** join/#asterisk viLeR (1000@ip-47-252.telesat.com.co) |
| 19:14.09 | Shido6 | and Ford Credit |
| 19:14.16 | Nugget | what do you expect? "quality" is job one. where do you think that leaves "answering the phone"? |
| 19:14.18 | sivana | hehe |
| 19:14.34 | Shido6 | where are you sivana? |
| 19:14.41 | AgiNamu | somewhere really cold |
| 19:14.49 | Shido6 | want a tour of Fords football field data center? |
| 19:15.01 | sivana | I'm in Ontario |
| 19:15.04 | Shido6 | there are at least 3 huge ones in Dearborn |
| 19:15.09 | ikey | did any one have experiance on MFC r2 protocol woth asterisk |
| 19:15.16 | *** join/#asterisk Nukemizer (~Nuke@66.237.85.58.ptr.us.xo.net) |
| 19:16.13 | sivana | ya |
| 19:16.31 | Shido6 | opencall |
| 19:16.33 | Shido6 | spandsp.org |
| 19:16.55 | Shido6 | http://www.spandsp.org/unicall/installing-mfcr2.html ? |
| 19:17.00 | *** join/#asterisk r0d3nt|m (nobody@wsip-24-234-241-84.lv.lv.cox.net) |
| 19:17.19 | ikey | yeah unicall |
| 19:17.30 | ikey | i tried comipling and i installed it |
| 19:17.49 | Nukemizer | Does anyone know if there a way to have an external phone be an ACD agnt ? Like a cell phone or Zap device ? |
| 19:18.04 | ikey | only issue is the line is getting blocking signal and when i start compiling asterisk i should get unicall module in there |
| 19:18.16 | *** join/#asterisk sandnigg0r (~crunkuser@66-55-197-254.gwi.net) |
| 19:18.27 | Shido6 | yes |
| 19:18.35 | Shido6 | for the longest |
| 19:18.39 | KristinH | I guess I cannot setup a dialplan like is: exten => ${EXTEN},1,Dial(${EXTEN},20,r) |
| 19:18.46 | Shido6 | I had a phone call into the nufone pbx |
| 19:18.50 | Shido6 | and when a sales call came in |
| 19:18.53 | Shido6 | it would ring my cell |
| 19:18.58 | Shido6 | and my crisco 7960 |
| 19:19.14 | *** join/#asterisk mw` (id0383@p5480C701.dip.t-dialin.net) |
| 19:19.19 | Shido6 | you can have an ip phone, analog phone or celly be in the queue |
| 19:19.24 | Shido6 | or act as an agent |
| 19:19.30 | mw` | hi |
| 19:19.38 | Shido6 | KristinH, lord |
| 19:19.41 | Shido6 | here..... |
| 19:19.57 | ariel_ | KristinH, that will not work to well. |
| 19:20.13 | KristinH | so i am finding out |
| 19:21.07 | mw` | does zaptel (with hfc) ignore * as prefix? if i dial *9xxx i get into context s (instead of *9.), without * i get into the right context |
| 19:21.20 | Nukemizer | Shido, Do you have login to let the sell phone start taking calls ? or just let the cell phone be in the ACD group ? |
| 19:21.59 | *** join/#asterisk IronHelix (~irc@ool-182c3fe9.dyn.optonline.net) |
| 19:22.37 | KristinH | and can anyone tell me which is best in a large scale installation (multiple states, offices) asterisk or asterisk + ser |
| 19:23.54 | Shido6 | http://pastebin.ca/8557 |
| 19:23.57 | Shido6 | there ya go KristinH |
| 19:24.04 | Shido6 | thats a nice start |
| 19:24.56 | *** join/#asterisk Lee__ (~lee@ool-44c26ebc.dyn.optonline.net) |
| 19:25.32 | *** join/#asterisk Cheng29 (~cheng29@d57-87-253.home.cgocable.net) |
| 19:25.57 | *** join/#asterisk sangee (~rkuru@207.188.77.82) |
| 19:26.04 | bjohnson | KristinH: you could do exten => ${PHONE1:4},1,Macro(stdexten,${PHONE1VM},${PHONE1}) |
| 19:26.29 | bjohnson | where PHONE1=SIP/2201 |
| 19:27.25 | KristinH | i'm really starting to believe that ser+asterisk is the best way to tackle this |
| 19:28.15 | *** join/#asterisk Mike (~mike@201.135.48.158) |
| 19:28.32 | Mike | hey guys something wrong with g729? im trying to connect to a carrier using it and it seems its not there |
| 19:28.45 | bjohnson | I would think that * + iax would be the best way |
| 19:28.56 | bjohnson | Mike: did you buy it? |
| 19:29.30 | KristinH | problem is all the client phones are sip |
| 19:29.46 | KristinH | and i am using TNT's for the PSTN gateway |
| 19:30.41 | Mike | bjohnson: its pass-trou |
| 19:31.25 | KristinH | and some of the bigger fish want their phones to ring whether they are at location a or location z |
| 19:31.38 | KristinH | without having to do anything aaside turn on their phone |
| 19:31.43 | ariel_ | KristinH, then you should use something more like exten => 786X.,1,Dail(Sip/${EXTEN}) or use a macro to determine other things. But this asumes that the did you give them is the same as there sip device. |
| 19:32.01 | *** join/#asterisk pr0m (~pr0metheu@ip-wv-68-187-250-031.charterwv.net) |
| 19:32.12 | *** join/#asterisk abatista (~Ariel@dsl-20-177.cofs.net) |
| 19:33.00 | KristinH | ariel_, i'm talking about users that pick up the phone, get a dialtone and want it to work like a pots |
| 19:33.53 | KristinH | back later |
| 19:33.57 | KristinH | thanks everyone |
| 19:34.05 | *** part/#asterisk KristinH (~KristinH@muppet.geekgirls.us) |
| 19:34.18 | *** join/#asterisk rrk (~chatzilla@rrcs-67-53-9-175.west.biz.rr.com) |
| 19:34.20 | abatista | if a user picks up a phone and wants to dial your still going to have to make some rules. like send there call to a dial out context like exten => _X.,Dial(Zap/g1/${EXTEN}) |
| 19:35.14 | ManxPower | "I don't believe Digium is interested in using a web based forum just because a few people don't understand how to manage e-mail. " --Me |
| 19:35.39 | epoch | e...mail? |
| 19:35.51 | epoch | do they have that on the interweb yet? |
| 19:37.21 | chap | Like Homer Simpson said.. "The Internet. Is that thing still around?" |
| 19:37.35 | epoch | hahaha |
| 19:37.55 | Nugget | "the internet? they have that on computers now?" |
| 19:38.03 | cjk | is there a variable i can use in the dialplan containing the username? |
| 19:38.18 | ManxPower | cjk: all variables are documented in README.variables |
| 19:39.11 | ddum | Someone using the Dlink DVG-1120S? Is the FXO port ONLY available when the switch in the back is in "PSTN" mode? As in, is it not usable as a FXO for Asterisk? |
| 19:41.52 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 19:42.36 | *** join/#asterisk hajekd (~hajekd@21.208.65.212.contactel.net) |
| 19:44.15 | hajekd | I have issues with changing system ID of TE110P....? |
| 19:44.49 | *** join/#asterisk harryvv (~leonardo@S010600055d210201.vs.shawcable.net) |
| 19:44.50 | *** join/#asterisk Tili (~Tili@202-133-65-154-dialup.sat.net.pk) |
| 19:48.23 | *** join/#asterisk JerJer[mobile] (~nonyobizn@65.173.197.174) |
| 19:48.56 | *** join/#asterisk whiplash (~whiplash@deepthought.armory.com) |
| 19:48.57 | *** join/#asterisk loick (~loick@APuteaux-151-1-46-102.w82-124.abo.wanadoo.fr) |
| 19:58.38 | bjohnson | ddum: that is my understanding .. it's an emergency pass through |
| 20:03.36 | VirTERM | anyone using realtime with iax trunks/peers? I need some tips |
| 20:05.29 | eKo1 | About what? |
| 20:08.21 | *** join/#asterisk mogorman (~mogorman@207.111.174.1) |
| 20:09.43 | ManxPower | Is there anyone here familiar with Brussels? |
| 20:09.58 | VirTERM | table structure |
| 20:10.01 | VirTERM | ..working |
| 20:10.52 | *** part/#asterisk Defraz (~t0tal@dhcp161-pool2.wireless.fiber.dcdi.net) |
| 20:10.56 | Matth_Away | where is the asterisk cvs? cvs.digium.com doesn't seem to resolve |
| 20:11.10 | *** join/#asterisk chrislwade (~clwade@river104.bigriver.net) |
| 20:12.27 | ManxPower | ~astcvs |
| 20:12.28 | jbot | rumour has it, astcvs is echo "CVS-HEAD:"; cvs co asterisk asterisk-sounds libpri zaptel; echo "CVS 1.0.x:"; cvs co -r v1-0 asterisk asterisk-sounds libpri zaptel; echo "Anyone that uses CVS and is not on asterisk-cvs mailing list is an idiot! See http://lists.digium.com/mailman/listinfo/asterisk-cvs" |
| 20:12.59 | *** join/#asterisk doughecka (~dheckaman@doughecka.user) |
| 20:17.53 | *** join/#asterisk DeadDawg (~DeadDawg@CPE000d5659d23f-CM000a739a1839.cpe.net.cable.rogers.com) |
| 20:17.56 | *** join/#asterisk _Sam-- (sam@ns2.kneedraggers.com) |
| 20:18.09 | _Sam-- | does anyone have any recommendations for a USB style headset for use with softclients? |
| 20:18.21 | *** join/#asterisk Carp1 (carp_xigon@ip-204-97-151-114.modem.logical.net) |
| 20:18.39 | MattH | ~jbot |
| 20:18.40 | jbot | rumour has it, jbot is the shipboard computer, but you may call me eddie if it helps you relax |
| 20:18.52 | nestAr | EDDIE! |
| 20:18.54 | nestAr | :) |
| 20:19.43 | MattH | ~help |
| 20:19.58 | nestAr | _Sam--: plantronics makes a USB adapter for their headsets |
| 20:20.03 | nestAr | it's kinda pricey though |
| 20:20.10 | DeadDawg | i find plantronics headsets are nice |
| 20:20.15 | Carp1 | ~seen JerJer |
| 20:20.18 | jbot | jerjer <~JerJer@dsl-106-170.che.centurytel.net> was last seen on IRC in channel #asterisk, 32d 4h 1m 23s ago, saying: 'mrgoby: sure'. |
| 20:20.26 | jets | WAKE UP DIGIUMMM |
| 20:20.27 | _Sam-- | i saw that adapter...its as much as most nice USB headsets |
| 20:20.33 | *** join/#asterisk mgth (~chatzilla@dsl093-001-038.det1.dsl.speakeasy.net) |
| 20:20.34 | nestAr | yea |
| 20:20.34 | _Sam-- | but i am needing USB w/ a quick disconnect |
| 20:21.02 | Carp1 | JerJer has been on his mobile for 32 days? |
| 20:21.13 | _Sam-- | i only found one company that has a quick-disconnect type USB headset |
| 20:21.26 | nestAr | we're just doing quick disconnects into Polycom phones.. Softphones = lose IMHO |
| 20:21.31 | MattH | ~seen MattH |
| 20:21.33 | jbot | matth is currently on #asterisk (2h 48m 16s). Has said a total of 23 messages. Is idling for 2s |
| 20:21.47 | *** join/#asterisk ckruetze (~nospam@i3ED65A4E.versanet.de) |
| 20:22.11 | jaiger | nestAr, what headset do you recommend for polycom? I was just looking for some |
| 20:22.58 | nestAr | jaiger: we're using GN Netcom's |
| 20:23.03 | nestAr | hold and i'll give you a model |
| 20:23.15 | bjohnson | ManxPower: just the sprouts |
| 20:23.17 | *** join/#asterisk moy (~kvirc@201.137.244.172) |
| 20:23.44 | Majestik | I'm trying to get a sip connection from a real IP through my NAT to my asterisk box.. but the packets don't seem to be getting past the firewall box. I'm running OpenBSD.. |
| 20:24.03 | bjohnson | why not cordless headsets? |
| 20:24.23 | bjohnson | Majestik: switch to iax2 if possible |
| 20:24.33 | _Sam-- | i would like some cordless USB ones...they are only about 250 each |
| 20:24.48 | moy | Hi everybody. Is the info in /proc/zaptel realiable to know wich zaptel cards are working and the channels assigned to them? |
| 20:24.49 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
| 20:24.52 | bjohnson | cordless bluetooth for approx $60 USD |
| 20:24.53 | nestAr | we're uning the 2200 series over the head type headsets. |
| 20:24.54 | _Sam-- | or if osmeone would make an over-the head bluetooth that would be perfect |
| 20:24.55 | Majestik | bjohnson: I'm trying to get xlite to connect remotely.. don't think I can do IAX2, unless I install asterisk on my laptop... |
| 20:25.10 | _Sam-- | show me an over the head type bluetooth one, please! |
| 20:25.12 | bjohnson | Majestik: use an iax2 softphone? |
| 20:25.34 | bjohnson | _Sam--: picky |
| 20:25.45 | _Sam-- | i dont know that you can get 8 hours of talk time out of a behind the ear blue tooth set |
| 20:25.46 | Majestik | bjohnson: good idea... is there a decent one for Mac? |
| 20:25.54 | _Sam-- | my guys are on the phone at least 6 to 7 hours out of 8 |
| 20:26.09 | bjohnson | no idea .. I run linux so you might have luck with iaxcomm |
| 20:26.33 | bjohnson | _Sam--: no .. most I've seen is about 4 hours |
| 20:27.21 | bjohnson | _Sam--: although I saw one that hung around your neck and the mic pinned to your shirt .. Sony I think |
| 20:27.29 | PTG1234 | yah i have an 8 hour bluetooth |
| 20:27.32 | bjohnson | it might have had better battery |
| 20:27.56 | _Sam-- | any USB headset will work with any softclient under windows? |
| 20:28.06 | _Sam-- | er i know that is a big statement... |
| 20:28.08 | _Sam-- | but generally? |
| 20:28.10 | bjohnson | no idea |
| 20:28.18 | PTG1234 | sam: just about all of them, i recommend the plantronics or if you wanna spend more and get the best senheiser |
| 20:28.26 | bjohnson | most headsets I use plug into the sound card |
| 20:28.56 | _Sam-- | bjohnson : that is the type ive been using, but i cant find any of them either with a quick disconnect so if an agent wants to go to the bathroom they leave the set on |
| 20:28.58 | Shido6 | senheiser? |
| 20:29.32 | _Sam-- | sennheiser makes bluetooth ones |
| 20:30.26 | *** join/#asterisk zno (~chatzilla@ip-160-79-174-101.autorev.intellispace.net) |
| 20:30.29 | *** join/#asterisk mjdyer (~mjdyer@dsl-20-105.cofs.net) |
| 20:30.48 | _Sam-- | er im not finding sennheiser bluetooth |
| 20:30.58 | zno | why does Allison rock socks? |
| 20:31.15 | zno | did she give away some ivr stuff? |
| 20:32.22 | *** join/#asterisk zotz (~zotz@24.231.32.191) |
| 20:32.27 | epoch | maybe bkw_ has the hots for her |
| 20:32.36 | eKo1 | that's impossible. |
| 20:32.39 | epoch | though, that'd be pretty disturbing |
| 20:32.40 | eKo1 | unless he's bi. |
| 20:32.47 | epoch | eKo1: or unless she's a he |
| 20:32.48 | epoch | ! |
| 20:32.56 | *** join/#asterisk emacsen (~serge@pcp09295616pcs.arlngt01.va.comcast.net) |
| 20:32.59 | eKo1 | true that |
| 20:33.08 | eKo1 | so bi or tranny |
| 20:33.16 | Nugget | allison puts the mint in all our julips. |
| 20:33.20 | epoch | or both? |
| 20:33.23 | _Sam-- | bjohnson : there wouldnt be any problem having 15 people in the same area all using bluetooth headsets? |
| 20:33.35 | _Sam-- | i mean no interference or issues to worry about |
| 20:34.06 | Nugget | bluetooth and 802.11b|g can be a problem. |
| 20:34.21 | emacsen | Hi. I installed the hardware (the knockoff FXS card) and the asterisk ubuntu package. Now I don't know what to do. |
| 20:34.49 | Nugget | we don't know what you want to do. |
| 20:35.12 | Carp1 | whats the latest version of mIRC? |
| 20:35.18 | jets | What's mIRC? |
| 20:35.20 | bjohnson | _Sam--: mileage will vary |
| 20:35.24 | emacsen | Nugget: well 1) Know it's all working. 2) "Answering Machine" 3) Make SIP calls, once I get some headset to do soft phone |
| 20:35.27 | bjohnson | jets: an IRC client |
| 20:35.36 | Nugget | emacsen: 0) http://voip-info.org/ and read. |
| 20:35.40 | emacsen | jets: a proprietary Windows one |
| 20:35.42 | bjohnson | ~docs |
| 20:35.43 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 20:35.47 | jaiger | can anyone recommend good hold music? stuff that won't piss off customers? |
| 20:35.59 | bjohnson | jaiger: Black Sabbath |
| 20:36.04 | emacsen | jaiger: Rammstein |
| 20:36.13 | *** join/#asterisk mikeh720 (~mh720@c-24-0-113-5.client.comcast.net) |
| 20:36.18 | jets | hey anthm whatssssss up :) |
| 20:36.27 | mikeh720 | greetings all |
| 20:36.41 | jets | I did put the PO in but I had forgotten about it |
| 20:36.45 | bjohnson | jaiger: David Hasselhoff |
| 20:36.46 | Carp1 | Whats the best IRC client? lol |
| 20:36.47 | lyoungz_ | emacsen: I installed the asterisk@home package |
| 20:37.02 | emacsen | lyoungz_: that's not a package, that's a disro |
| 20:37.03 | bjohnson | Carp1: X-chat of course |
| 20:37.07 | jets | Our callcenter projects got pushed back a fe wmonths but i'll go ask my bosssssssss if he signed that |
| 20:37.22 | emacsen | Nugget: that's a lot of docs. Is there a howto? |
| 20:37.23 | lyoungz_ | emacsen: it comes with a lot of pre-configuration done, with AMP, |
| 20:37.32 | emacsen | lyoungz_: AMP? |
| 20:37.44 | lyoungz_ | a web based management console |
| 20:38.02 | mikeh720 | asterisk management portal, not bad |
| 20:38.03 | lyoungz_ | gives you simplified (possibly) configuration options |
| 20:38.04 | bjohnson | good if it does exactly what you want it to do |
| 20:38.19 | lyoungz_ | actually it has given me some grief already. |
| 20:38.32 | moy | Does anybody know if the info in /proc/zaptel is realiable to know wich zaptel cards are working and the channels assigned to them? |
| 20:38.39 | lyoungz_ | tends to obscure things, and it wouldn't let me use multiple 'allow' directives on a trunk. |
| 20:38.48 | Shido6 | cat /proc/zaptel/(number here) |
| 20:38.51 | bjohnson | if it doesn't .. it's like falling into a lagoon of man-eating sharks before learning to swim |
| 20:39.02 | lyoungz_ | when I tried, it erased all of my config for that trunk! |
| 20:39.03 | mikeh720 | lol |
| 20:39.11 | mw` | does zaptel (with hfc) ignore * as prefix? if i dial *9xxx i get into context s (instead of *9.), without * i get into the right context |
| 20:39.46 | Shido6 | Try exten => _*9... |
| 20:39.48 | lyoungz_ | asterisk@home comes with an extensive extensions configuration pre-built. |
| 20:39.57 | lyoungz_ | I don't know if that is a good thing or a bad thing :-) |
| 20:40.12 | mw` | Shido6: ops i have _*9. of course |
| 20:40.17 | Shido6 | Ive scheduled more sessions thanks to asterisk@home than I ever have in the past year |
| 20:40.18 | bjohnson | good if it does exactly what you want it to do |
| 20:40.30 | Shido6 | its confused the shit out of people :) thank you asterisk@home. |
| 20:40.36 | mw` | Shido6: if i press call and dial after that, everything works |
| 20:40.37 | bjohnson | Shido6: sessions with your shrink? |
| 20:40.41 | Shido6 | hehe |
| 20:40.47 | Shido6 | Asterisk Training sessions |
| 20:41.19 | mikeh720 | they helped your * support business pick up? |
| 20:41.23 | Shido6 | yes |
| 20:41.27 | *** join/#asterisk Source707 (Source@196.46.64.9) |
| 20:41.57 | mikeh720 | anyone know if realtime * voicemail uses the same "options" format as voicemail.conf? i.e. vertical bars between options? |
| 20:42.03 | lyoungz_ | The lack of a how-to document has really made asterisk@home look very attractive. |
| 20:42.04 | moy | thanks SHido |
| 20:42.16 | mikeh720 | tiki doesn't know and I can't extract an answer from the source |
| 20:42.18 | lyoungz_ | unfortunately it -also- lacks a how-to document, and it needs one. |
| 20:42.26 | DeadDawg | So after scouring google looking for * to act as a MGCP client, I have come to the conclusion that I will have to stick with my hardware MGCP gateway, and connect its FXS interface to my FXO card in the * box. |
| 20:42.32 | zno | I'm thinking of using asterisk@home, but will you be restricted to using the gui to configure asterisk? |
| 20:42.52 | DeadDawg | zno: no you can always do manual editing |
| 20:42.54 | lyoungz_ | you can do the config files directly. |
| 20:43.02 | Shido6 | no you will not be restricted to the GUI |
| 20:43.06 | zno | DeadDawg: but won't that screw up the gui? |
| 20:43.10 | Shido6 | you will ALWAYS have access to the .conf files directly |
| 20:43.13 | lyoungz_ | but then again if you then touch them with the gui it might overwrite them :-) |
| 20:43.15 | Shido6 | you will always have CLI access |
| 20:43.18 | mikeh720 | but once you manually edit does the gui not like it? |
| 20:43.23 | *** join/#asterisk leonardox (~leonardox@dsl-200-78-97-55.prod-infinitum.com.mx) |
| 20:43.24 | zno | yeah |
| 20:43.25 | Shido6 | the gui is there to help you... *gulp* |
| 20:43.28 | *** part/#asterisk leonardox (~leonardox@dsl-200-78-97-55.prod-infinitum.com.mx) |
| 20:43.29 | DeadDawg | i believe the GUI just parses the conf files each time you load it. |
| 20:43.38 | eKo1 | gui? wtf?! |
| 20:43.44 | mikeh720 | web gui |
| 20:43.48 | mikeh720 | AMP |
| 20:43.48 | DeadDawg | although I have never used AMP or A@H |
| 20:43.52 | lyoungz_ | the GUI stores its config in a database |
| 20:43.53 | eKo1 | noooooo |
| 20:44.06 | zno | I'm gonna give it a try and see if I can replicate my conf files with just the gui |
| 20:44.06 | bjohnson | lyoungz_: there are hundreds of howto documents |
| 20:44.10 | zno | if so that'll be nice |
| 20:44.12 | Shido6 | um |
| 20:44.13 | Shido6 | zno |
| 20:44.13 | bjohnson | lyoungz_: try google |
| 20:44.16 | Shido6 | backup your configs |
| 20:44.18 | Shido6 | first |
| 20:44.21 | emacsen | no amp for ubuntu? |
| 20:44.22 | Shido6 | before trying amp |
| 20:44.24 | *** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net) |
| 20:44.30 | lyoungz_ | yeah, mazes of twisty little passages all slightly different |
| 20:44.36 | Shido6 | i wanna run ubuntu so bad I can taste it |
| 20:44.37 | zno | Shido6: it's all in cvs :) |
| 20:44.39 | lyoungz_ | if you are old enough to remember 'adventure' |
| 20:44.58 | bjohnson | zno: forget it |
| 20:44.59 | jaiger | Shido6, what's it taste like? |
| 20:45.13 | *** join/#asterisk Source707 (Source@196.46.64.9) |
| 20:45.17 | bjohnson | zno: AMP uses specialized conf file with variables all over the place |
| 20:45.22 | zno | bjohnson: have you tried? is it a waste of time? |
| 20:45.26 | bjohnson | zno: choose one system and stick to it |
| 20:45.31 | ManxPower | Revolution OS is now available on NewFlix |
| 20:45.37 | bjohnson | it's good if it does exactly what you want it to do |
| 20:45.38 | ManxPower | and NetFlix too! |
| 20:45.44 | bjohnson | if it doesn't .. it's like falling into a lagoon of man-eating sharks before learning to swim |
| 20:46.12 | lyoungz_ | wow, a google on asterisk howto -does- show a lot of stuff. |
| 20:46.15 | *** part/#asterisk moy (~kvirc@201.137.244.172) |
| 20:47.03 | DeadDawg | If anybody knows if support for MGCP UserAgent is being worked on, I would be interested to know. Perhaps I should search the mailing lists. |
| 20:47.50 | zno | right now i'm the only guy who knows how to configure the phone, so I want others to be able to do it too. I hate being the "phone" guy |
| 20:48.08 | DeadDawg | being in demand is usually a good thing. |
| 20:48.28 | eKo1 | eh, not when everybody is bitching that their phone doesn't work.ç |
| 20:49.11 | DeadDawg | hehe that is true. |
| 20:49.28 | emacsen | no one knows of an amp deb? |
| 20:49.29 | eKo1 | unless the person bitching are the chicks for accounting. |
| 20:49.57 | zno | well my core competency is software dev, not phone admin |
| 20:51.14 | Majestik | Is there a limit to the size of a password for iaxcomm? |
| 20:51.29 | *** join/#asterisk nvrswork (RUR@cwn7.ads.uwaterloo.ca) |
| 20:51.35 | *** join/#asterisk angler_ (~angler@suid.digium.com) |
| 20:51.47 | *** part/#asterisk chrislwade (~clwade@river104.bigriver.net) |
| 20:51.54 | Majestik | nevermind.. I must have pasted it wrong |
| 20:51.55 | *** join/#asterisk bah (048830696@ACA8323F.ipt.aol.com) |
| 20:52.18 | nvrswork | Hi im trying to re route a call to my office extention. Is it possible for asterisk to dial the university where I work, wait a couple seconds then phone my extention number? |
| 20:52.18 | *** join/#asterisk NewSole (david@i216-58-44-245.avalonworks.net) |
| 20:52.21 | *** join/#asterisk Ro[b]ert (~acidburnn@cust.7.204.adsl.cistron.nl) |
| 20:52.27 | Ro[b]ert | ehlo ppl... |
| 20:53.06 | Ro[b]ert | how can i Playback more .gsm files?? at once..?? comma seperated? |
| 20:53.13 | anthm | see D(digits) in app_dial |
| 20:53.18 | Ro[b]ert | Playback(today,your,head,is, etc |
| 20:54.40 | bjohnson | nvrswork: yes |
| 20:54.41 | Ro[b]ert | ok.. like Playback(test/your/asterisk/server) |
| 20:54.43 | Ro[b]ert | ?? |
| 20:54.47 | *** join/#asterisk dsfr (~dsfr@207.111.174.1) |
| 20:54.53 | ManxPower | Ro[b]ert: Use multiple playback statements |
| 20:55.10 | Ro[b]ert | ManxPower: mmmmm how? |
| 20:55.17 | bjohnson | hehe |
| 20:55.28 | *** join/#asterisk johan_hammy (~mhammett@c-24-14-114-227.client.comcast.net) |
| 20:55.30 | nvrswork | bjohnson, what is the symbol for pausing |
| 20:55.38 | bjohnson | I think it is a comma |
| 20:55.47 | anthm | w |
| 20:55.49 | *** join/#asterisk bannerman (~bannerman@209.216.176.42) |
| 20:55.52 | ManxPower | Ro[b]ert: That's such a basic question, it's obvious you've read nothing. |
| 20:55.54 | bannerman | Yay, basic QOS is easy |
| 20:55.54 | ManxPower | ~docs |
| 20:55.55 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
| 20:55.57 | anthm | D(123www1234) |
| 20:55.59 | johan_hammy | Could anyone tell me how to get this Asterisk module for webmin setup? |
| 20:56.03 | anthm | w = 500 ms |
| 20:56.19 | bjohnson | nvrswork: you could also do 2 dial statements with a wait in between |
| 20:56.34 | PTG1234 | anthm: you got a sec? Do you know of a good example of how to implement a cli command? |
| 20:56.35 | *** join/#asterisk jhiver (~jhiver@AStDenis-103-1-14-65.w81-248.abo.wanadoo.fr) |
| 20:56.37 | nvrswork | i have something like Dial(SIP/5195554567,,,3266) |
| 20:56.50 | Ro[b]ert | ManxPower: im reading the docs now... just want to speed up things... |
| 20:56.52 | bjohnson | nvrswork: replace the commas with w's |
| 20:56.58 | nvrswork | ok thanks |
| 20:57.05 | ManxPower | nvrswork: "w" only works on ANALOG ports. |
| 20:57.06 | bjohnson | nvrswork: you weren't listening to anthm |
| 20:57.11 | jhiver | guys I'm wondering what's the typical number of users / number of max. simultaneous calls ratio for VoIP services such as NuFone or VoIPJet |
| 20:57.13 | ManxPower | You have to listen to anthm |
| 20:57.36 | PTG1234 | there should be a max jhlver |
| 20:57.37 | anthm | cli command ? |
| 20:57.40 | PTG1234 | you pay per minute |
| 20:57.42 | harryvv | jhiver well with a lag of over 2,000 ms from voipjet not much :) |
| 20:57.44 | PTG1234 | anthm: yah in a c module |
| 20:58.14 | bjohnson | jhiver: I'd guess about 1 in 10 |
| 20:58.29 | johan_hammy | Could anyone tell me how to get this Asterisk module for webmin setup? |
| 20:58.30 | harryvv | bj, I was told that by a company also |
| 20:58.43 | epoch | ManxPower: does D() work at all on non-analog ports? |
| 20:58.47 | anthm | not much to em you build a registration struct with a func pointer to a func to be called when the cli is executed and its done a lot like int main in c except it has a fd you use with ast_cli to print output |
| 20:59.12 | anthm | D() works on anything , I made it myself |
| 20:59.16 | harryvv | day accounts for Local bussinesses night accounts for home users and a late evening for mabey dating service accounts. :) |
| 20:59.21 | PTG1234 | whats the name of the struct? |
| 20:59.27 | harryvv | 10 to 1 should work |
| 20:59.30 | epoch | I remember having problems with D() and a PRI card |
| 20:59.39 | ManxPower | epoch: D() should work on all port types |
| 20:59.51 | epoch | maybe it was the w thing that was busted |
| 21:00.11 | anthm | look in most apps for a cli exampler |
| 21:00.52 | anthm | like app_realtime |
| 21:00.54 | anthm | for instance |
| 21:01.14 | ManxPower | Dial(Zap/1/1234www5678) won't work on PRI |
| 21:01.26 | ManxPower | Since a PRI is not analog. |
| 21:01.29 | jontow | hmm |
| 21:01.58 | jontow | what about Dial(Zap/g1/1NXXNXXXXXX|60|D(wwwwXXXXXwwwNXXXXXX)) |
| 21:01.59 | jontow | ? |
| 21:02.06 | ManxPower | That should work |
| 21:02.12 | jontow | ok.. i've got it mostly working |
| 21:02.15 | ManxPower | Since D works with non-analog ports as well as analog ports. |
| 21:02.30 | mjdyer | Anyone had experience with iax.cc as a provider? |
| 21:02.34 | Mother_ | I had something weird happen today, two zap channels froze, you could dial out, but incoming was not detected - a reboot cured it |
| 21:02.37 | jontow | whats happening though, its dialing to a pager service and leaving XXXXX as the pager #, then NXXXXXX as the callback number.. |
| 21:02.45 | bjohnson | mjdyer: a little |
| 21:02.52 | Mother_ | however, does this say much for reliability? these people couldn't really have noticed |
| 21:02.54 | jontow | there is no terminating '#' or anything like that it has to dial, but there is a confirmation tone and you have to wait until it stops beeping to hangup |
| 21:02.57 | mjdyer | bjohnson: how did it go? |
| 21:03.00 | epoch | ManxPower: why wouldn't w work on PRI, anyways? |
| 21:03.04 | bjohnson | fine so far |
| 21:03.09 | jontow | * is hanging up before i tell it to, and so it isn't waiting for the end of the tone |
| 21:03.16 | ManxPower | epoch: Because w as part of the number only works on analog ports. |
| 21:03.18 | bjohnson | other than waiting forever for one of my two DIDs |
| 21:03.22 | jontow | even if i do Dial(${TRUNK}/.....|60|D(..)) |
| 21:03.27 | epoch | ManxPower: that's not a terribly good explanation ;P |
| 21:03.27 | jontow | and on the next priority Wait(10) |
| 21:03.32 | jontow | .. it still does the same thing |
| 21:03.34 | epoch | "why doesn't it work" "because it doesn't" ;P |
| 21:03.58 | ManxPower | epoch: Because "w" in the dial is only supported by wcfxo and wcfxs? |
| 21:04.01 | PTG1234 | i just need to do a cli_register |
| 21:04.01 | bjohnson | and yet it is perfectly clear |
| 21:04.04 | PTG1234 | with that structure? |
| 21:04.06 | anthm | wait after dial is irrelivant |
| 21:04.13 | epoch | ManxPower: ah, I see |
| 21:04.16 | anthm | , PTG1234, yep |
| 21:04.16 | jontow | hmm |
| 21:04.19 | ManxPower | jontow: put w's after you finish sending the DTMF |
| 21:04.22 | PTG1234 | simple enough |
| 21:04.24 | jontow | that does the same thing |
| 21:04.28 | anthm | Dial keeps control of the channel until the call is over |
| 21:04.38 | anthm | you never ever get to the next pri till its over |
| 21:04.46 | Mother_ | so any ideas as to why a zap channel suddenly stops detecting rings inbound? |
| 21:04.54 | anthm | use more w |
| 21:05.10 | anthm | and zapscan it to spy and hear if the timing is wrong |
| 21:05.19 | bjohnson | they're like little mountains |
| 21:05.37 | anthm | I wanted W to be a full second but mark wanted it simple ! |
| 21:05.40 | *** join/#asterisk syslod (~yurplsl@65.114.15.70) |
| 21:05.41 | bannerman | So OpenBSD is the way to go if you want real QoS on a software router, huh |
| 21:05.47 | syslod | Hello. |
| 21:05.57 | PTG1234 | bannerman: or freebsd :) since they use the same thing now |
| 21:06.11 | bannerman | Oh. Which is better (translated easier) to get into? |
| 21:06.22 | syslod | Anyone know how to make record work on a handup rather than pressing #? |
| 21:06.24 | PTG1234 | which OS? |
| 21:06.28 | bannerman | yeah |
| 21:06.32 | PTG1234 | freebsd defintely |
| 21:06.36 | PTG1234 | much wider user base |
| 21:06.36 | bannerman | ok |
| 21:06.39 | bannerman | ok, cool |
| 21:06.40 | Nugget | yay freebsd. |
| 21:06.49 | ManxPower | syslod: Record is not supposed to record the # |
| 21:07.00 | Nugget | openbsd still rules for QoS, though. altq is pretty raw in freebsd |
| 21:07.02 | bannerman | I always thought iptables was the way to go for routing |
| 21:07.08 | Nugget | linux is poo. |
| 21:07.16 | PTG1234 | Nugget: supposably its now the same |
| 21:07.20 | VirTERM | still looking for someone who actually runs * with realtime (iax) |
| 21:07.22 | PTG1234 | Nugget: since the 5.3 tree |
| 21:07.34 | syslod | ManxPower: It doesn't the problem I have is I have built intercom but they have to push the # to end recording. I want it to end on 3 sec of silence (which works) or a hangup. |
| 21:07.45 | ManxPower | VirTERM: Your mailing list search didn't turn up anything? |
| 21:07.50 | VirTERM | nope |
| 21:07.58 | jontow | ok.. so its dialing correctly and just failing to finish correctly |
| 21:08.02 | jontow | like something in the middle is hanging up :( |
| 21:08.07 | ManxPower | VirTERM: Since it's realtime, I can't help you. |
| 21:08.15 | VirTERM | nothing real on wiki nor through google... |
| 21:08.21 | johan_hammy | Could anyone tell me how to get this Asterisk module for webmin setup? |
| 21:08.29 | VirTERM | hmm |
| 21:08.29 | Lee__ | would anyone have a list of all the available configuration options in sip.conf? |
| 21:08.38 | VirTERM | you are here in realtime :) |
| 21:08.42 | ManxPower | Lee__: That would be sip.conf.sample. |
| 21:08.44 | syslod | If I wait 3 secs it work or if I hit # to finish it work. If I hangup it doesn't seem to do anything. |
| 21:08.57 | ManxPower | VirTERM: No, there's a 5 second delay imposed by the censors. |
| 21:09.06 | Nugget | heh |
| 21:09.06 | VirTERM | almost realtime |
| 21:09.14 | Lee__ | ManxPower: that looks more like an annotated sample, rather then a full list but I'll take your word for it... |
| 21:09.58 | ManxPower | I don't use CVS-HEAD and realtime, |
| 21:11.07 | ManxPower | I use 1.0.x stable and realtime is not available in that production version of Asterisk |
| 21:11.22 | ManxPower | Lee__: If it doesn't have every option listed then it's a bug. |
| 21:11.33 | VirTERM | well, then you can't really help me :) |
| 21:11.36 | ManxPower | Lee__: Just be sure to look at the sip.conf.sample from your CURRENT source. |
| 21:11.45 | ManxPower | VirTERM: that's what I said in the first place. |
| 21:12.37 | *** part/#asterisk iq (~iq@70-57-182-28.omah.qwest.net) |
| 21:12.58 | ManxPower | VirTERM: if you don't get an answer here then ask on the mailing lists. |
| 21:13.02 | *** join/#asterisk iq (~iq@70-57-182-28.omah.qwest.net) |
| 21:13.27 | VirTERM | i did |
| 21:13.42 | VirTERM | this is my resource... |
| 21:13.52 | syslod | The only thing records says about hangup is it returns a -1. I want it to do the same thing it does if you hit #. Is that an option? |
| 21:14.26 | VirTERM | my last resource |
| 21:14.52 | ManxPower | Darwin35: But you LIKE pain. |
| 21:15.06 | jontow | nevermind.. looking like they shut off my pager's service. |
| 21:15.34 | *** join/#asterisk florz (nobody@2001:1a50:503c:0:0:0:0:1) |
| 21:18.48 | Darwin35 | Manx just a little |
| 21:21.09 | Darwin35 | ahh its the friking zap channel |
| 21:21.22 | Darwin35 | 1.0.6 |
| 21:21.31 | Darwin35 | has a zap channel issue |
| 21:21.39 | Darwin35 | frick |
| 21:21.53 | ManxPower | Darwin35: I assumed it was, since Asterisk runs in userspace and userspace apps should not be able to panic the kernel. |
| 21:22.28 | *** join/#asterisk vidia22 ([U2FsdGVkX@vidiamob4.vidiacom.com) |
| 21:22.31 | johan_hammy | Could anyone tell me how to get this Asterisk module for webmin setup? |
| 21:22.38 | Darwin35 | no on fbsd it is being started as root |
| 21:22.46 | *** join/#asterisk fugitivo (~ajf@201.255.109.28) |
| 21:22.48 | Darwin35 | but not now |
| 21:23.12 | vidia22 | Could anyone help me with echo on sip phones - but just for the first few seconds.. |
| 21:23.15 | mikeh720 | does anyone have realtime voicemail options working from a database, e.g. saycid=yes|attach=yes ? Mine seems to ignore all options |
| 21:23.19 | ManxPower | johan_hammy: I don't think anyone here uses that. |
| 21:23.34 | VirTERM | yes, I have voicemail working |
| 21:23.38 | VirTERM | from realtime |
| 21:23.51 | *** join/#asterisk Wazb (Wazb@207.245.215.111) |
| 21:23.53 | VirTERM | will exchange info for a dumpa of iax table from realtime :) |
| 21:23.56 | mikeh720 | vir - as do i, but options seem to be ignored |
| 21:23.59 | VirTERM | dump |
| 21:24.06 | JerJer[mobile] | realtime is a joke |
| 21:24.11 | *** join/#asterisk kjcsb (~ext_news@60.234.196.21) |
| 21:24.24 | mikeh720 | is that really JerJer saying that? |
| 21:24.31 | JerJer[mobile] | yes |
| 21:24.47 | Wazb | hi all |
| 21:24.53 | mikeh720 | do you mean realtime isn't primetime yet? |
| 21:25.01 | mikeh720 | howdy |
| 21:25.07 | JerJer[mobile] | not implemented properly into asterisk |
| 21:25.12 | JerJer[mobile] | bloat |
| 21:25.31 | Milligan | Is there anyone here that is willing and capable of being hired to assist me in setting up a production implementation of asterisk.... To replace an existing system? |
| 21:25.40 | mikeh720 | it's in a state of flux, but is already better for some people's needs than flatfiles |
| 21:25.40 | Wazb | <PROTECTED> |
| 21:25.47 | VirTERM | you have to start somewhere.... |
| 21:26.04 | jontow | milligan; where're you located? |
| 21:26.04 | JerJer[mobile] | its crap |
| 21:26.10 | mikeh720 | waz are you running more than 1 asterisk on that box? |
| 21:26.16 | Milligan | jontow: Arkansas.... |
| 21:26.24 | JerJer[mobile] | mikeh720: we don't run flatfiles |
| 21:26.29 | JerJer[mobile] | nor do we have realtiime |
| 21:26.35 | ManxPower | Milligan: I will, but not using CVS-HEAD |
| 21:26.38 | JerJer[mobile] | use the power of astersk, do not side-step it |
| 21:26.55 | Wazb | no , actaully i got this error when i tried to install asterisk/channels/h323 |
| 21:26.55 | JerJer[mobile] | cvs -head is far superior code |
| 21:27.07 | JerJer[mobile] | a warning is NOT an error |
| 21:27.10 | mikeh720 | waz: grep bindaddr in the etc/asterisk directory and make sure you have your correct IP that is on the box listed |
| 21:27.11 | *** join/#asterisk aminorex (~tony@c66.191.69.132.dul.mn.charter.com) |
| 21:27.36 | ManxPower | hardwire: 2008. |
| 21:27.42 | hardwire | heh |
| 21:27.51 | chap | thats nailing it down. :) |
| 21:28.20 | mikeh720 | vir: do your options work for vm users in realtime? |
| 21:28.52 | bannerman | So FreeBSD+Dummynet |
| 21:28.53 | VirTERM | let me check if I actually have any on the box running realtime |
| 21:29.36 | mikeh720 | wazb: not familiar with h323, but it probably has a configuration file that goes into etc/asterisk, make sure it has the right bindaddr line |
| 21:29.47 | VirTERM | no options there, but I can put something as this is just my home box |
| 21:29.52 | JerJer[mobile] | A WARNING IS NOT AN ERROR |
| 21:29.55 | VirTERM | what would you like me to test? |
| 21:29.56 | JerJer[mobile] | pay attention |
| 21:30.01 | ManxPower | Wazb: 90% of H323 problems are you not using the required version of the OpenH323/PWlib libs |
| 21:30.09 | mw` | what is meant by shared irq (for the hfc card) the irq number or the physical interrupt line? |
| 21:30.27 | mikeh720 | vir, I appreciate it, i've tested on cvshead 3/23 and 3/31 and both read in the options, but don't seem to act on them |
| 21:31.00 | fugitivo | one question, i'm trying to make a sip call to my friend, we're both using asterisk and a softphone, but when we answer, we can't listen anything, asterisk log: Maximum retries exceeded on call .......@PRIVATE_ADDRESS for sqno 102 |
| 21:31.23 | VirTERM | my cvs is appx. 5 days old, if you want I can actually update it to the mar/31 and try some options |
| 21:31.25 | fugitivo | i think the problem is that asterisk is trying to communicate with the internal address of my friend |
| 21:31.31 | VirTERM | or I can add options now.... |
| 21:31.34 | mikeh720 | vir: you could just try attach=yes and see if your email gets the voicemail as an attachment (assuming that you have attachments turned off by default) |
| 21:31.35 | VirTERM | your choice :) |
| 21:31.38 | Wazb | i am using the correct veriosn as they have specified in their README file in Asterisk directory |
| 21:31.51 | florz | mw`: That's the same. |
| 21:31.59 | VirTERM | ok, I will turn it off in global and set as an option, 5 mins |
| 21:32.06 | *** join/#asterisk meanphil (~pmurray@222-152-246-166.jetstream.xtra.co.nz) |
| 21:32.06 | mikeh720 | ty |
| 21:32.15 | ManxPower | Wazb: you confirmed that there are no other openh323 or pwlib libs on the system? |
| 21:32.40 | JerJer[mobile] | grrrr |
| 21:33.00 | JerJer[mobile] | wazb: that message you get is NOT NOT NOT NOT NOT NOT NOT a fming error |
| 21:33.05 | JerJer[mobile] | pay attention |
| 21:33.17 | tzanger | JerJer[mobile]: it sounds like you've you've you've got a framing error |
| 21:33.19 | ManxPower | JerJer[mobile]: Why don't you make your Makefile just download and install the correct libs and cut the user out of the loop? |
| 21:33.31 | tzanger | ManxPower: he prefers education :-) |
| 21:33.41 | mikeh720 | wazb, stop asterisk, run 'netstat -an | grep LISTEN' and you can see what ports your machine is already listening on, sounds like h323 can't bind because the ip is not on the machine, or the port is already in use |
| 21:33.59 | ManxPower | tzanger: users are not smart enough to be educated. |
| 21:34.04 | mw` | florz: are you sure? arent there only 4 interrupt lines for all pci slots? (INTA..D) |
| 21:34.13 | tzanger | ManxPower: that's where the entertainment comes in |
| 21:34.17 | tzanger | mw`: yup |
| 21:34.23 | mikeh720 | heh |
| 21:34.38 | VirTERM | testing.... |
| 21:34.56 | florz | mw`: Well, how should the PCI controller distinguish different devices pulling the same line? |
| 21:35.23 | tzanger | florz: it doesn't |
| 21:35.29 | tzanger | florz: the interrupt handler does that |
| 21:35.37 | mw` | polling the int register, maybe :) |
| 21:35.37 | bannerman | dummynet doesn't seem to do a lot more than you can do with tc |
| 21:35.39 | tzanger | by polling each device connected to that interrupt |
| 21:36.00 | tzanger | that is why PCI interrupts are supposed to be level and not edge triggered |
| 21:36.20 | nestAr | i use a laser level when hanging my interupts. |
| 21:36.25 | mw` | so it should be the best to connect the hfc card to a not shared interrupt line |
| 21:36.30 | nestAr | bob villa showed me the way. |
| 21:36.38 | VirTERM | ok, I set in globals; attach=no then set option in SQL to attach=yes |
| 21:36.42 | VirTERM | it didn't work |
| 21:37.05 | mikeh720 | right, so looks confirmed |
| 21:37.08 | VirTERM | now I will remove the attach statement from globals to see if it superseeds the sql |
| 21:37.17 | VirTERM | hold on |
| 21:37.49 | mikeh720 | good test, by design individual vm options should override globals, maybe that isnt happening |
| 21:38.08 | *** join/#asterisk Muxen (~mux@bonin.phillips.com) |
| 21:38.25 | mikeh720 | actually saycid=yes doesn't work either, and I don't globally set that |
| 21:38.32 | florz | tzanger: oh, really? =:-) |
| 21:39.18 | tzanger | florz: yes. when two devices pull hte same IRQ the handler finds the first, handles it and reenables interrupts. the line is still low so it gets triggered again and the second device gets its turn. you can't (easily) do that with edge-triggered IRQs |
| 21:40.15 | florz | tzanger: Gee, what I wrote was the explanation for wm`, why different devices on the same IRQ line can't trigger different IRQs =:-) |
| 21:40.35 | VirTERM | testing.. |
| 21:40.51 | VirTERM | did yout iax with realtime? |
| 21:41.30 | florz | mw`: Yes, at least the one providing the timing if you are using my patch. The others don't matter then ... |
| 21:41.36 | tzanger | florz: but the PCI controller isn't responsible for determining which slot requested the interrupt. That's done in software. shitty drivers can hold the interrupt a long time whihc is why digium's cards don't particularly like sharing interrupts. |
| 21:42.03 | tzanger | now sangoma's cards handle sharing just fine, it's a driver issue but it is going to take a lot of rework to get digium's drivers to a similar state |
| 21:42.09 | ManxPower | And since the Digium cards dont have any buffer on them, any delay in servicing an interrupt will cause corrupted data |
| 21:42.29 | tzanger | the Digium way appears to be "it works for 75% of people, that's good enough" and for most cases (75% in fact) it works just fine |
| 21:42.35 | tzanger | ManxPower: bingo |
| 21:42.38 | kjcsb | Does anyone have experience of integrating SER and Asterisk? Specifically I am interested in thoughts on how to best keep track of users. Presumably use realtime and have asterisk and ser looking at the same tables? |
| 21:42.40 | *** join/#asterisk Blissex (~Blissex@82-69-39-138.dsl.in-addr.zen.co.uk) |
| 21:43.31 | *** join/#asterisk kram (~mark@kram.digium.sponsor.pdpc) |
| 21:43.31 | *** mode/#asterisk [+o kram] by ChanServ |
| 21:43.31 | mw` | florz: ah, i just wanted to ask, should i use your patch, even if ive got only one hfc? |
| 21:43.35 | *** join/#asterisk zetho (~zetho___@dku135.neoplus.adsl.tpnet.pl) |
| 21:43.35 | VirTERM | actually, I just got the attachment |
| 21:43.54 | VirTERM | so , no entry for attach under [general] but option in sql |
| 21:43.58 | tzanger | kram: did you get my messages? |
| 21:44.05 | VirTERM | no, we could have attach=yes as default... |
| 21:44.07 | VirTERM | dunno |
| 21:44.16 | zetho | hello, did any body managed to run SER with Asterisk ? |
| 21:44.32 | mikeh720 | vir, i'll check that out, so looks like that flatfile option overrides realtime voicemail options |
| 21:44.41 | Hmmhesays | lots of people |
| 21:44.50 | VirTERM | or, the default is "yes" |
| 21:45.26 | VirTERM | now, do you have iax peers/trunks running with realtime :)? |
| 21:45.34 | VirTERM | i know, that i must be annoying... |
| 21:45.36 | zetho | i just can't find any good howto to SER.. asterisk is great but SER has strange configuration file.. |
| 21:45.40 | mw` | florz: but maybe it arent the interrupts that cause bad sound quality, because isdn <-> isdn just works fine, only sip <-> isdn is bad |
| 21:45.46 | jhiver | zetho: I agree |
| 21:46.05 | tzanger | mw`: you can also have poor audio from shitty network cards / switches / networks |
| 21:46.10 | jhiver | a good, simple howto of Asterisk + SER + nathelper would be _so cool_ |
| 21:46.26 | zetho | maybe because SER was ("is") made for Fokus for their own purposes |
| 21:46.36 | zetho | you're right jhiver |
| 21:46.37 | Blissex | jhiver: that cannot exist -- it is not a simple matter. |
| 21:46.58 | Blissex | jhiver: just explaining the _terminology_ to a n00b can take a long time. |
| 21:47.00 | *** join/#asterisk Maveric (maveric@ip68-3-248-136.ph.ph.cox.net) |
| 21:47.01 | zetho | Blissex, but a step-after-step howto would be good |
| 21:47.01 | florz | mw`: Probably, at least I'd prefer it over the original one, since older versions of the original driver have some bugs that are fixed in newer versions and the newest version requires RTAI which makes use a bit complicated ... |
| 21:47.04 | mw` | tzanger: with a sip client everything is ok |
| 21:47.10 | AgiNamu | Screw SER |
| 21:47.15 | VirTERM | heh |
| 21:47.21 | AgiNamu | A good solution would be to just work on some Asterisk scalability / clusterint. |
| 21:47.24 | zetho | more or less it's not a very rare case (SER + nathlpr + ASTRSK) |
| 21:47.29 | VirTERM | yes, unless you want to play with sms... |
| 21:47.39 | mikeh720 | vir: i have sip users configured in RT |
| 21:47.50 | VirTERM | no problems? |
| 21:47.52 | zetho | sms works FINE just for me... really.. |
| 21:47.57 | VirTERM | i didn't pass iax yet :) |
| 21:47.58 | AgiNamu | That SER is a flirt. all redirects. no media handling. |
| 21:48.14 | *** join/#asterisk goldenolden (~goldenold@65.171.196.23) |
| 21:48.14 | AgiNamu | sipteast. |
| 21:48.16 | VirTERM | it all depends what you want to do with sms |
| 21:48.19 | AgiNamu | siptease. |
| 21:48.27 | zetho | hmm.. ivr + voicemail |
| 21:48.36 | mw` | florz: ill give it a try, i think rtai breaks most of my kernel patches |
| 21:48.37 | goldenolden | hi |
| 21:48.39 | zetho | asterisk has one tousand times better docs than SER.. :( |
| 21:48.47 | VirTERM | mikeh720: did you just use the table layout like on wiki? |
| 21:48.49 | ManxPower | Blissex: You've not been here in the evening USA time, have you? |
| 21:48.50 | zetho | but asterisk has no registrar :( |
| 21:48.55 | AgiNamu | wait... there's a project with worse docs then asterisk? |
| 21:49.01 | goldenolden | has anyone successfully bought anything from thevoipconnection.com? they charged my card two weeks ago and never delivered the goods and do not return phone calls or emails. |
| 21:49.04 | AgiNamu | man, now I'm DEFINATELY staying away from SER |
| 21:49.09 | Blissex | ManxPower: sometimes, why :-) |
| 21:49.24 | zetho | AgiNamu, so what registrar do you use ? |
| 21:49.26 | florz | mw`: And it still doesn't solve the problem, which my patch does =:-) |
| 21:49.34 | AgiNamu | Zetho, asterisk all the way |
| 21:49.38 | AgiNamu | I use IAX as well |
| 21:49.39 | AgiNamu | fuck SIP |
| 21:49.40 | *** join/#asterisk lilwookie (~zoidmeste@modemcable215.87-81-70.mc.videotron.ca) |
| 21:49.40 | Shido6 | you dont need SER |
| 21:49.44 | AgiNamu | SIP is such a piece of shit |
| 21:49.47 | Shido6 | u have Asterisk |
| 21:49.58 | VirTERM | gonna switch machines; be back in 15 |
| 21:50.02 | mikeh720 | vir: yes I used the sip table on the wiki |
| 21:50.04 | PTG1234 | sip rocks |
| 21:50.04 | AgiNamu | what I did was write a semi softswitch in .NET, exposed it via SOAP |
| 21:50.08 | Blissex | AgiNamu: but you just said to fuck it -- ever more strange sex preferences :-) |
| 21:50.08 | mikeh720 | works fine |
| 21:50.09 | AgiNamu | and then im gonna hook asterisk up |
| 21:50.12 | zetho | But Asterisk is for PC's with some strange E1/T1/ISDN cards right ? |
| 21:50.23 | AgiNamu | blissex, think klingon sex. |
| 21:50.24 | zetho | i Have a gateway that is standalone |
| 21:50.28 | Darwin35 | no |
| 21:50.33 | AgiNamu | zetho.... um, no |
| 21:50.37 | AgiNamu | it's not for PCs with strange cards. |
| 21:50.40 | Darwin35 | asterisk is for sip/voip also |
| 21:50.50 | AgiNamu | and IAX and MGCP and H323 and whatever else |
| 21:50.54 | AgiNamu | yes, obviously :) |
| 21:50.56 | Darwin35 | no need for weird cards |
| 21:50.58 | zetho | Darwin35,.... really :) |
| 21:51.11 | eKo1 | Why would anyone call an e1 card weird? |
| 21:51.13 | ManxPower | "<AgiNamu> what I did was write a semi softswitch in .NET, exposed it via SOAP" <-- now THERE'S something that's SICK. |
| 21:51.14 | AgiNamu | i dont have any cards in 2 of my asterisk server. |
| 21:51.25 | AgiNamu | manx, only to the unitiated |
| 21:51.27 | zetho | So, a simple case - LAN with sip phones + ASterisk + Voip Gateway connected to E1 - is Asterisk good ? |
| 21:51.28 | Darwin35 | you need to read the asteriskpage again |
| 21:51.33 | mikeh720 | vir: the attach option works now, thanks for clearing that up, setcid doesn't work but that's an advanced voicemail option that was added to cvshead according to bugtracker, but doesn't work for me via realtime, I;ll try if from flat and see if it works |
| 21:51.39 | AgiNamu | www.asterisk.org start there |
| 21:51.48 | Darwin35 | I use it for home service over cable |
| 21:51.48 | Blissex | zetho: YES, thats about the obvious case. |
| 21:51.51 | AgiNamu | or call Digium. Ask for Mark. |
| 21:51.54 | Maveric | yes zetho |
| 21:52.01 | AgiNamu | he'll guide you through all the features and explain what it does. |
| 21:52.06 | zetho | i have had asterisk once but i could not register phones in it :( |
| 21:52.12 | zetho | hehehe |
| 21:52.30 | Darwin35 | you did not setup sip right |
| 21:52.34 | zetho | but i did not come deeper to understand it.. |
| 21:52.35 | Blissex | zetho: I am writing a sample config, and a nice intro, previsely for that case. |
| 21:52.39 | Darwin35 | and you did not read the wiki |
| 21:52.42 | Blissex | zetho: wait some weeks :-) |
| 21:52.56 | ManxPower | Anyone have ideas on what I can present at Astricon? |
| 21:52.57 | zetho | oh Blissex , there's no howto like this ? |
| 21:53.07 | Darwin35 | Kram is god |
| 21:53.18 | Darwin35 | kram =mark at digium |
| 21:53.19 | Blissex | zetho: there are dozens of bits and pieces spread everywhere... And not that well done |
| 21:53.25 | AgiNamu | just dont talk about how GCC initializes variables |
| 21:53.32 | kjcsb | I'm following the SER conversation. My understanding was that if you want scalability, resiliancy and have SIP users, SER was the way to go |
| 21:53.40 | AgiNamu | kjcsb, that |
| 21:53.43 | AgiNamu | is the common way |
| 21:53.49 | Blissex | zetho: it can be done, and there are things like the Asterisk Live CD distro and the Asterisk at home project... |
| 21:53.51 | AgiNamu | or, build your own little softswitch |
| 21:53.54 | AgiNamu | thats what i want to do |
| 21:53.57 | ManxPower | kjcsb: That really depends on a lot. |
| 21:53.58 | kjcsb | Use Asterisk as the media gateway |
| 21:54.02 | Blissex | zetho: but I think I can do better... |
| 21:54.23 | zetho | okay... so let me think... if Asterisk is so good and popular ,,, why do people user SER ? the configuration is one thousand times more difficult (in SER) |
| 21:54.30 | Blissex | zetho: however, if you are not fussy on how clean the config is, a quick'n'dirty is not hard to do. |
| 21:54.34 | ManxPower | kjcsb: SER is really only useful for Asterisk people if you want to have phones that are behind NAT at different sites send their audio directly between each other and not via Asterisk |
| 21:54.51 | AgiNamu | zetho, Asterisk has some SERIOUS perf issues |
| 21:55.03 | mikeh720 | vir: actually this looks like a bug, i removed attach=yes from RT SQL options, and still get the attachment. It looks like if you comment out attach= in the voicemail.conf, the default is to send an attachment |
| 21:55.06 | AgiNamu | for instance, in a stock build, you can't even LOAD a million SIP users. |
| 21:55.07 | zno | zetho: the wiki says that you need SER to scale |
| 21:55.13 | Blissex | AgiNamu: but they are largely irrelevant for small setupd. |
| 21:55.17 | Blissex | AgiNamu: but they are largely irrelevant for small setups. |
| 21:55.20 | hajekd | can I use same voipjet account from two asterisk systems symultaneously? |
| 21:55.30 | AgiNamu | yea, but as soon as you get to about 20,000 IAX/SIP peers... things go slow as shit |
| 21:55.35 | zetho | right.. i'll use ser for now.... |
| 21:55.37 | AgiNamu | im just talking LOADING |
| 21:55.37 | ManxPower | Asterisk does not really support that many users. You want multiple asterisk servers |
| 21:55.45 | AgiNamu | let alone handling registrations |
| 21:55.47 | AgiNamu | and there's no reason it shuldn.t |
| 21:55.53 | Blissex | zetho: but you said it is for a small site with two phones... |
| 21:55.56 | *** join/#asterisk caesar2 (caesar@p5497AFA0.dip.t-dialin.net) |
| 21:56.04 | AgiNamu | as far as i can tell, it's just some poor data type selections. i.e., linked lists instead of a hashtable. |
| 21:56.11 | AgiNamu | But kpfleming is working on something along those lines |
| 21:56.14 | AgiNamu | to makie it kick ass |
| 21:56.17 | zetho | no - i did not said that, i only mentioned a case. My network is rather BIG |
| 21:56.17 | mikeh720 | sweet |
| 21:56.22 | ManxPower | zetho: Asterisk has PBX features, where other things like SER do not. |
| 21:56.22 | zetho | based on h323 :P |
| 21:56.44 | Blissex | zetho: well, if you are using H323 you are already doomed :-) |
| 21:56.46 | zetho | what PBX features mean for you ? ManxPower |
| 21:56.48 | *** join/#asterisk terrapen (~cjs@fw-01.satx.bikeworld.net) |
| 21:56.52 | terrapen | hrmmm |
| 21:56.54 | ManxPower | For the types of Asterisk systems I set up for clients I'll never ever need SER |
| 21:57.00 | terrapen | any nufone users around? |
| 21:57.04 | ManxPower | zetho: Try setting up voicemail only using SER. |
| 21:57.05 | tzanger | terrapen: yes |
| 21:57.11 | zno | nufone user here |
| 21:57.12 | zetho | no problem ManxPower |
| 21:57.18 | AgiNamu | if you only have a couple hundred users, its not an issue |
| 21:57.21 | terrapen | tz, can you dial toll-free numbers with your nufone account? |
| 21:57.28 | AgiNamu | In fact, i think you could hit maybe 1000 on a beefy machine. |
| 21:57.32 | ManxPower | Or connect to the PSTN using SER |
| 21:57.40 | terrapen | <PROTECTED> |
| 21:57.42 | tzanger | terrapen: never tried. I have a PRI I send that stuff through |
| 21:57.42 | terrapen | that's failing |
| 21:57.45 | terrapen | fast busy |
| 21:57.47 | zetho | voicemail + sending recoded mail for email and so on and so on ManxPower |
| 21:57.50 | zno | ManxPower: how many users are there on average for your clients? |
| 21:57.56 | *** join/#asterisk iCEBrkr (icebrkr@chrome.cyberdyne.org) |
| 21:57.58 | ManxPower | terracon: try not sending a leading 1 |
| 21:58.07 | AgiNamu | im designing a system that needs to handle say, a million users. |
| 21:58.20 | tzanger | ManxPower: odd, nufone wants leading one for all other NA numbers |
| 21:58.21 | ManxPower | zno: less than 300 and none or very few phones that are not on the local LAN |
| 21:58.22 | terrapen | manx, if i send a leading 1 on a non-toll-free Dial() it works just fine |
| 21:58.25 | iCEBrkr | ONE MILLLLLLLLLLLLLION users |
| 21:58.26 | zno | terrapen: I had some problems earlier this afternoon |
| 21:58.26 | tzanger | and 011 for international |
| 21:58.27 | AgiNamu | in reality, i'd only expect say, 10K or 20K |
| 21:58.35 | ManxPower | tzanger: Really? *shrug* I've been known to be wrong before. |
| 21:58.36 | terrapen | tz: i agree |
| 21:58.37 | AgiNamu | iCEbrkr, yea, design for a million |
| 21:58.41 | tzanger | iCEBrkr: who's got one million users |
| 21:58.44 | AgiNamu | gets you thinking on how you're gonna do that. |
| 21:58.46 | terrapen | someone with nufone, try ccalling this please: |
| 21:58.54 | terrapen | 1-877-816-8243 |
| 21:58.58 | zno | ManxPower: are they voip users or traditional pbx users? just curious |
| 21:59.00 | eKo1 | AgiNamu is trying to take over Guatemala's telephone market. |
| 21:59.05 | AgiNamu | hehe yea bitch |
| 21:59.07 | terrapen | i can call it from my cellphone just fine |
| 21:59.16 | kjcsb | So one day Asterisk might be scalable for thousands (and more) of users but at the moment, use SER. Is that fair? |
| 21:59.18 | AgiNamu | the problem is if you design a solution for a couple thousand |
| 21:59.22 | terrapen | its almost like the NuFone folks don't want me calling toll-frees |
| 21:59.25 | AgiNamu | and it doesnt scale up, you can get screwed |
| 21:59.28 | kjcsb | I mean SER AND Asterisk |
| 21:59.31 | terrapen | perhaps because they don't make any money on them? |
| 21:59.33 | AgiNamu | if you design for 1 million, you wont run into issues |
| 21:59.42 | zetho | kjcsb, in my opinion - you're right.. |
| 21:59.46 | ManxPower | Right now VoX/Net has maybe 100 users at most scattered across 6 locations |
| 21:59.50 | bannerman | terrapin: I just recently had trouble getting toll free numbers to work on liveVoip |
| 21:59.59 | terrapen | that's kind of crappy, i think |
| 22:00.01 | bannerman | terrapin: my problem was in the callerid I was sending. had to change it to a local number. |
| 22:00.12 | lilwookie | is SER linux based? |
| 22:00.13 | Blissex | but wait a sec: I have no need to try a large scale Asterisk config, but shouldn't that be handled by the MySQL backend? |
| 22:00.16 | AgiNamu | For instance, using gSOAP making calls to Windows /ASP.NET, I could do 470request/sec. That includes the entire call from linux to windows (over a lan) hitting an MSSQL db, etc. |
| 22:00.22 | ManxPower | terracon: nufone's trunks for toll frees are different from their non-toll-free trunks, so maybe they are having another toll-free outage. |
| 22:00.23 | AgiNamu | BLissex, MySQL is shit |
| 22:00.30 | zetho | lilwookie, hehe :) ofcourse |
| 22:00.40 | Blissex | AgiNamu: but it scaled a lot better than '.conf' files. |
| 22:00.41 | AgiNamu | its like "Hey, for version *5* we're gonna have stored procedures!!! Wowzers!" |
| 22:00.45 | Blissex | AgiNamu: but it scales a lot better than '.conf' files. |
| 22:00.49 | lilwookie | I never heard if SER before.. cool I will check it out |
| 22:00.57 | Shido6 | what kind of problems are you having when you reach 1k users with sip, I have yet to encounter these problems |
| 22:00.59 | AgiNamu | that doesnt mean mysql doesnt suck |
| 22:01.05 | zetho | lilwookie, iptel.org |
| 22:01.12 | ManxPower | I'm looking forward to the next major release of Asterisk for it's directory features. |
| 22:01.15 | Blissex | lilwookie: SIP Express Registrar IIRC |
| 22:01.17 | lilwookie | Well Asterisk can scale fine, just horizontally |
| 22:01.23 | AgiNamu | Shido6, register 1,000 IAX2 customers on a single Asterisk box |
| 22:01.28 | AgiNamu | it brings it to its knees in my tests. |
| 22:01.28 | lilwookie | Thanks zetho & blackjack |
| 22:01.31 | lilwookie | er Blissex |
| 22:01.34 | AgiNamu | CPU gets nailed. |
| 22:01.40 | eKo1 | AgiNamu: How are you testing it? |
| 22:01.40 | Shido6 | and if you're having 8xx number issues call me at 248-724-VoIP |
| 22:01.40 | AgiNamu | this was on a 2.8GHz Celeron |
| 22:01.44 | Shido6 | so I can resolve them |
| 22:01.51 | AgiNamu | eKo1, registering 1000 IAX2 peers. |
| 22:01.58 | Shido6 | Celeron? |
| 22:01.59 | zetho | maybe the keyword - Celeron AgiNamu :) |
| 22:02.00 | Shido6 | are you kidding me? |
| 22:02.00 | zno | AgiNamu: was it all the transcoding you think? |
| 22:02.05 | eKo1 | Yes, what are you using to generate those peers? |
| 22:02.06 | AgiNamu | the celeron is a test machine |
| 22:02.07 | Shido6 | you dont run 1k users on a Celeron |
| 22:02.09 | zetho | how much RAM AgiNamu ? |
| 22:02.09 | AgiNamu | zno, no transcoding |
| 22:02.16 | Shido6 | thats your problem |
| 22:02.18 | AgiNamu | 512MB. but asterisk only used 40MB |
| 22:02.23 | Shido6 | doesnt matter |
| 22:02.29 | Shido6 | you dont use a celeron for that amount of load |
| 22:02.30 | AgiNamu | at any rate, 1K users is nothing |
| 22:02.32 | zetho | so how much cache memeory does celeron have ?? AgiNamu |
| 22:02.33 | ManxPower | AgiNamu: one would expect a dual or quad(?) Xeon would be required for that. |
| 22:02.43 | AgiNamu | Manx: Then it's poor code. |
| 22:02.48 | ManxPower | And one would also not expect all 1,000 users to register in a very short time. |
| 22:02.51 | lilwookie | I use a Duak right now for about 1k users |
| 22:02.54 | *** join/#asterisk Derkommissar (~alberto@66.64.215.7.nw.nuvox.net) |
| 22:02.54 | ManxPower | AgiNamu: so fix it. |
| 22:02.57 | AgiNamu | registration isn't exactly a huge computational task. |
| 22:03.07 | AgiNamu | yea, kpfleming is working on a new object model |
| 22:03.25 | Derkommissar | i have a te100 and i keep getting this message when i run ztcfg |
| 22:03.26 | Derkommissar | line 0: Unable to open master device '/dev/zap/ctl' |
| 22:03.29 | Derkommissar | :-/ |
| 22:03.38 | Blissex | Derkommissar: drivers? |
| 22:03.40 | AgiNamu | I bet a lot of it is the same reason Asterisk can't parse a large .conf file. |
| 22:03.44 | ManxPower | Derkommissar: running kernel 2.6 |
| 22:03.47 | AgiNamu | for IAX2 peers, it's an N*N operation |
| 22:03.50 | fugitivo | Derkommissar: ztcfg |
| 22:03.55 | Blissex | Derkommissar: have you actually created those device filed? |
| 22:04.13 | fugitivo | Derkommissar: run ztcfg |
| 22:04.44 | *** join/#asterisk goatmilk (~goatmilk@130-127-45-50.chouse.resnet.clemson.edu) |
| 22:04.46 | Derkommissar | i did |
| 22:04.46 | lilwookie | Actually what would be a decent channel bank(s) for 1K user load (zap channels) |
| 22:05.04 | eKo1 | are you nuts. |
| 22:05.08 | fugitivo | Derkommissar: did you load the modules? |
| 22:05.09 | Derkommissar | [root@prueba zaptel]# ztcfg |
| 22:05.09 | Derkommissar | Notice: Configuration file is /etc/zaptel.conf |
| 22:05.09 | Derkommissar | line 0: Unable to open master device '/dev/zap/ctl' |
| 22:05.11 | Derkommissar | yes |
| 22:05.25 | lilwookie | eKo1, heheh me? |
| 22:05.28 | fugitivo | Derkommissar: try wcfxo? |
| 22:05.36 | eKo1 | yes |
| 22:05.41 | Derkommissar | wcte11xp 27936 0 |
| 22:05.42 | Derkommissar | zaptel 207492 1 wcte11xp |
| 22:05.54 | Derkommissar | ? |
| 22:06.02 | kjcsb | lilwookie: I've read good things about ADIT600 |
| 22:06.09 | Derkommissar | fugitivo, i have a wcete11xp |
| 22:06.14 | lilwookie | lol why? I am in the process of replacing the local telco in a condo of 1004 units |
| 22:06.55 | lilwookie | Yeah if I am forced to I was going to use AT600's but was looking for something in a larger qty |
| 22:07.38 | JunK-W | 1000 zap will be a large deploy. |
| 22:08.36 | lilwookie | Well my firm is doing triple play in the building, so its pretty large project, IPTV/VDSL/VOICE services |
| 22:09.28 | jhiver | 1k user load != 1k zap channels |
| 22:09.37 | eKo1 | will everything go through vdsl? |
| 22:09.48 | Shido6 | its not that large, you could do with 5 dual xeon systems with 2 quad span cards each |
| 22:10.03 | lilwookie | up yeah I need to provide a FXS for each apt |
| 22:10.07 | fugitivo | Derkommissar: are you using udev? |
| 22:10.13 | Derkommissar | udev ? |
| 22:10.19 | Derkommissar | im sorry for my ignorance |
| 22:10.39 | JunK-W | Shido6: this is what i call large. :) |
| 22:11.09 | Shido6 | doing voicemail,too lilwookie ? |
| 22:11.15 | lilwookie | Shido6, yeah thats sort the route I am going now, but 4 duals really, but was looking for a larger bank to trim down |
| 22:11.39 | Derkommissar | fugitivo, what is udev ? |
| 22:11.58 | lilwookie | Shido6, not at the condo site, we have a softswitch from Hauwei being installed soon to handle the services themselves |
| 22:12.33 | lilwookie | Basically using * as a mini switch for the banks and a small interface to the door and garage systems |
| 22:13.16 | lilwookie | plue I love trunking IAX over fibre :) mouahaha |
| 22:13.30 | *** join/#asterisk Carp1 (carp_xigon@ip-204-97-151-178.modem.logical.net) |
| 22:14.29 | Shido6 | do it! |
| 22:14.49 | harryvv | lil, using dtmf to control the garage and doors? |
| 22:15.17 | harryvv | you mean a dtmf decoder/relay board ;) |
| 22:15.22 | hardwire | lilwookie: in hawaii? |
| 22:15.29 | lilwookie | no.. In Canada |
| 22:15.41 | hardwire | heh |
| 22:15.42 | hardwire | ok |
| 22:15.43 | mmostafa | hi all, I have a problem with getting asterisk running. When I try to run asterist i get the following error: [cdr_custom.so] => (Customizable Comma Separated Values CDR Backend) |
| 22:15.43 | mmostafa | <PROTECTED> |
| 22:15.44 | mmostafa | Apr 1 00:16:36 WARNING[22660]: loader.c:388 __load_resource: cdr_custom.so: load_modulefailed, returning -1 |
| 22:15.44 | mmostafa | <PROTECTED> |
| 22:15.44 | mmostafa | Apr 1 00:16:36 WARNING[22660]: loader.c:509 load_modules: Loading module cdr_custom.so failed! |
| 22:15.46 | Derkommissar | Is there anyone from Digium here? can i get some help with this card ? |
| 22:15.46 | hardwire | whats hauwei? |
| 22:15.59 | lilwookie | the company who's switch were looking to get is from china.. huawei (pronouced WOWWEE) |
| 22:16.03 | Derkommissar | Huawei is a chinese vertion of everything cisco does |
| 22:16.16 | hardwire | heh |
| 22:16.23 | Derkommissar | :-) |
| 22:16.25 | zetho | Huawei is really good.. i have some stuff made by huawei |
| 22:16.27 | lilwookie | Yep.. they are in court with cisco I belive over stolen software convepts even |
| 22:16.36 | Derkommissar | I have a SoftX3000 |
| 22:16.42 | Derkommissar | i configured it in ecuador. |
| 22:16.50 | harryvv | lilwookie I heard about triple play but for a building? |
| 22:17.03 | lilwookie | Derkommissarm heheh thats what we are getting a SoftX3000 |
| 22:17.16 | Derkommissar | They cant or couldnt sell their producst in US |
| 22:17.29 | Derkommissar | lilwookie, its not a big thing... |
| 22:17.36 | Derkommissar | lots of problems with H323 |
| 22:17.41 | lilwookie | tripleplay only seems to make real sense in communities like Building |
| 22:17.47 | evolutionxtinct | Is it better to install asterisk and all its components from source or is it just as good from an RPM, if ok to use RPM anyone know where to find a FC3 RPM for Asterisk and such? |
| 22:17.48 | Derkommissar | the only thing that works rigth is MGCP |
| 22:17.59 | lilwookie | Derkommissar, yeah and I am waiting for them to finish the 911 support |
| 22:18.47 | Derkommissar | lilwookie, you are in for a lot of problems, their support sucks |
| 22:18.50 | Maveric | evolutionxtinct i would only use source |
| 22:18.54 | Derkommissar | MAN |
| 22:19.00 | Derkommissar | i keep getting this |
| 22:19.01 | Derkommissar | line 0: Unable to open master device '/dev/zap/ctl' |
| 22:19.10 | Derkommissar | when i try to run ztcfg |
| 22:19.22 | Derkommissar | dam it.... :-( i was used to the old cards |
| 22:19.28 | lilwookie | really? damnit... I've been seening some spotty support on thier VDSL but they said it was because its a new line.. |
| 22:19.59 | harryvv | what did the softswitch cost you |
| 22:20.09 | Derkommissar | it cost 2.5 mill |
| 22:20.15 | Derkommissar | witouth support |
| 22:20.23 | Derkommissar | and with 12 e'1 mgw |
| 22:20.29 | harryvv | mm obviosly backbone equipment. |
| 22:20.34 | fugitivo | Derkommissar: sorry |
| 22:20.48 | fugitivo | Derkommissar: check if you have /etc/udev/ |
| 22:21.01 | Derkommissar | yup |
| 22:21.03 | Shido6 | 2.5 mill |
| 22:21.04 | Shido6 | wow |
| 22:21.07 | harryvv | man |
| 22:21.07 | Shido6 | for 12 e1's |
| 22:21.08 | lilwookie | Derkommissar, yeah I was quoted 750K us for 1million user cap |
| 22:21.14 | Shido6 | thats what $10k in Asterisk land |
| 22:21.23 | harryvv | ohhhh |
| 22:21.35 | Derkommissar | i would rather use asterisk than huawei |
| 22:21.35 | harryvv | yea 2.5 million what denomination? |
| 22:21.36 | fugitivo | Derkommissar: /etc/udev/rules.d/50-udev.rules something like that? |
| 22:21.41 | Shido6 | pesos |
| 22:21.41 | Derkommissar | USD |
| 22:21.47 | harryvv | what? |
| 22:21.59 | harryvv | only the big boys can afford that! |
| 22:21.59 | Derkommissar | do i edit that file ? |
| 22:22.00 | hardwire | heh |
| 22:22.12 | fugitivo | Derkommissar: yep |
| 22:22.13 | hardwire | I just got "Mad Scientist" put next to my name in the company directory. |
| 22:22.23 | file[laptop] | lol |
| 22:22.31 | Derkommissar | ok |
| 22:22.40 | harryvv | What kind of bandwith on it? |
| 22:23.01 | lilwookie | I am thinking of going with * instead even sppoke to the Digium and the firm they recommended to help build it.. just its a horozontal platform |
| 22:23.21 | hardwire | what does horizontal platform mean? |
| 22:23.23 | Maveric | anybody have the 7960 random reboot problem during established rtp |
| 22:23.30 | Shido6 | random reboot |
| 22:23.33 | Shido6 | ouch |
| 22:23.38 | Shido6 | what firmware? |
| 22:23.39 | hardwire | Maveric: monkeys plugging/unplugging the power? |
| 22:23.47 | Maveric | 7.2 |
| 22:23.52 | Derkommissar | fugitivo, what am i looking for in this file ? |
| 22:23.53 | Shido6 | i run 7.1 |
| 22:24.06 | Shido6 | time to step down :) |
| 22:24.20 | Maveric | i know of there used to be an issue when the the rtp counters rolled over or some such |
| 22:24.26 | Derkommissar | Shido6, people like brand names.... i would bet anything that asterisk is better than the softx3000 |
| 22:24.28 | Maveric | but i thought that was fixed |
| 22:24.44 | Maveric | needless to say the ceo's phone just rebooted randomly on an important call lol wasn't happy |
| 22:24.45 | lilwookie | hardwire, basically you add more CPU's Servers to the solution as you need more capacity.. to what I understand |
| 22:24.58 | hardwire | spiffay |
| 22:25.01 | harryvv | ohh mavric thats not good. |
| 22:25.36 | harryvv | mav, just switch phones if you can. if you dont have one switch it with the lowest peon in the company. |
| 22:25.37 | Maveric | it doesn't happen hardly at all |
| 22:25.43 | Derkommissar | fugitivo, what should i do next |
| 22:25.51 | Maveric | harry its not just his phone |
| 22:25.54 | Maveric | it hardly ever happens |
| 22:26.00 | Maveric | like i said its just very random |
| 22:26.14 | harryvv | that is generally not good enough for a ceo unless he does not mind |
| 22:26.41 | harryvv | some client on the other end might thing he got hung up on in a heated argument :) |
| 22:27.31 | *** join/#asterisk MultiF (john@66-220-72-1.gogenesis.com) |
| 22:28.35 | MultiF | I was wondering if someone here could help me with a problem |
| 22:29.13 | *** join/#asterisk invi_ (~invi_@64.128.35.234) |
| 22:29.39 | Maveric | harryvv he understands its new |
| 22:29.41 | *** join/#asterisk jesster (jesster@jesster.org) |
| 22:29.42 | invi_ | hi there |
| 22:29.54 | harryvv | yea bugs will be worked out |
| 22:29.54 | Maveric | and i've done a very good job of getting it implemented with very little hassle |
| 22:30.07 | Maveric | we've been running on it for nearly a year now i tihnk |
| 22:30.08 | Maveric | think |
| 22:30.41 | invi_ | can anybody tell me what do i have to do so * will work with Cisco's QoS |
| 22:31.14 | ManxPower | invi_: set the tos to 0x8b |
| 22:31.20 | *** join/#asterisk bjohnson (~bjohnson@ip193-172.dsl.istop.com) |
| 22:31.38 | ManxPower | that should make it look like a EF DSCP |
| 22:32.58 | Maveric | ManxPower you have any ideas on the 7960 reboot problem? |
| 22:33.28 | Maveric | i wish it was easy to recreate that way i could track down whats going on |
| 22:33.35 | Shido6 | bring it down to 7.1 |
| 22:33.59 | *** join/#asterisk [Pi] (pi@dsl81-214-19370.adsl.ttnet.net.tr) |
| 22:34.13 | MultiF | I have an IVR setup and option 2 dials a sip number, if that number is unavailible I send it back to asterisk with a diversion header for the number, however asterisk sends it to voicemail as 2 ( like option 2) |
| 22:34.40 | MultiF | is there a way to change this behavior... |
| 22:35.19 | `Sauron | Yeah |
| 22:35.28 | `Sauron | look into +102 or something |
| 22:35.58 | MultiF | ? |
| 22:36.48 | `Sauron | read the docs about priority in extensions.conf |
| 22:36.49 | Shido6 | pastebin.ca your dialplan |
| 22:37.29 | lilwookie | MultiF, * will follow the order of your dial plan, Busy condition it jumps the next line + 102 for example |
| 22:37.42 | *** join/#asterisk r0d3nt|m (nobody@wsip-24-234-241-84.lv.lv.cox.net) |
| 22:38.03 | `Sauron | busy is +101 |
| 22:38.10 | `Sauron | I think no answer is +1, or +102 |
| 22:38.43 | invi_ | <PROTECTED> |
| 22:38.54 | Blissex | lilwookie: that '+ 101' thing makes me laugh every time... I just can't imagine why Asterisk has not got TCL or a Scheme based proper embedded languaged instead of its current mad syntax. |
| 22:39.42 | invi_ | <PROTECTED> |
| 22:40.11 | MultiF | hmm, I am sending this call to a sip extension on another box and it call forwards na back to * with a diversion header |
| 22:40.14 | lilwookie | Blissex, well Its different, but I got used to it. and its easy to teach to secrataties |
| 22:40.24 | MultiF | should not asterisk use info in this header |
| 22:40.50 | harryvv | what ata does vonage hand out to its customers? |
| 22:40.54 | lilwookie | I have 3 clients where the girl at the front does all the dial plan changes... and hasent pooched it yet |
| 22:41.45 | invi_ | <PROTECTED> |
| 22:42.06 | evolutionxtinct | Does anyone know of a decent howto site or newbie site other then the voip wiki site? |
| 22:42.45 | *** join/#asterisk zoa (~k@213.91.216.136) |
| 22:43.10 | *** part/#asterisk _Sam-- (sam@ns2.kneedraggers.com) |
| 22:43.17 | *** join/#asterisk ta[i]nted (~ta_i_nted@65-60-70-243-cust.telepacific.net) |
| 22:44.14 | *** join/#asterisk Achillies (~nick@165.252.43.77) |
| 22:44.28 | Blissex | evolutionxtinct: AsterixDocs.org perhaps |
| 22:44.38 | Achillies | anyone know a good guid or site that covers using cisco VOIP Routers with Asterisk |
| 22:46.06 | MultiF | achillies: How do you want to use a cisco voice router with asterisk? |
| 22:47.25 | ManxPower | invi_: that is correct |
| 22:48.08 | invi_ | ManxPower: 0xB8? |
| 22:48.22 | ManxPower | yes. |
| 22:48.45 | Achillies | i found it |
| 22:49.58 | invi_ | ManxPower: it does not work with cisco. im trying it on 3 diff routers w/o any results |
| 22:50.22 | ManxPower | invi_: The Cisco has to be configured for it of course. |
| 22:51.35 | invi_ | ManxPower: got QoS configured for EF. calls made with xten get priority but * does not |
| 22:52.05 | ManxPower | invi_: time to break out tcpdump |
| 22:53.16 | invi_ | ManxPower: did it last night. tos is set on packets |
| 22:54.02 | *** join/#asterisk Inv_Arp (junya@adsl-3-255-42.mia.bellsouth.net) |
| 22:54.03 | *** join/#asterisk johnnyb (~johnnyb@sdsl-38-17-139.tulsaconnect.com) |
| 22:54.44 | Inv_Arp | Hello World! |
| 22:54.52 | invi_ | ManxPower: im running out of ideas. been sitting on it for the last week |
| 22:55.24 | *** join/#asterisk W1thdraw (~Withdraw@ip70-181-96-254.oc.oc.cox.net) |
| 22:55.28 | Inv_Arp | hey who setup fwd to dial 800 numbers... lets compare configs |
| 22:55.53 | Maveric | wtf |
| 22:55.58 | Inv_Arp | invi_: whats the prob? |
| 22:56.22 | husher | so, when i plug in a TE110P, what's supposed to happen? |
| 22:56.42 | invi_ | Inv_Arp: cisco does not do QoS on * packets |
| 22:57.23 | _asr | yawn |
| 22:57.23 | bannerman | what makes some calls sound crappy and some calls sound nice? |
| 22:57.30 | Inv_Arp | your doing QOS by port or by protocol? |
| 22:57.36 | CoaxD | PROCLARUSH TEONAS |
| 22:57.39 | _asr | anybody get cisco as5300/5400 correctly passing caller id name to asterisk? |
| 22:57.48 | _asr | i've tried tinkering around with RPID to no avail |
| 22:57.48 | Inv_Arp | bannerman: bandwidth |
| 22:58.12 | invi_ | Inv_Arp: DSCP |
| 22:58.24 | bannerman | Inv_Arp: In my case it's definitely not a bandwidth thing. I can have a call going that sounds just horrible, crackly and stuff, and I can make an additional call and the new one (at the same time) sounds crystal clear |
| 22:58.29 | invi_ | Inv_Arp: tos=0xB8 |
| 22:58.36 | *** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net) |
| 22:59.45 | bannerman | at least, I don't see how it could be a bandwidth issue |
| 22:59.54 | *** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net) |
| 22:59.55 | Inv_Arp | bannerman: what codec? |
| 22:59.58 | bannerman | ulaw |
| 23:00.22 | invi_ | at this point im not sure if this is * or there is some discrimination going in cisco's QoS |
| 23:01.45 | *** join/#asterisk Faithful (~Faithful@202-6-145-116.ip.adam.com.au) |
| 23:02.01 | invi_ | packets r tagged with 0xb8 & cisco still puts it as Trivial traffic |
| 23:02.06 | mikeh720 | asr_, https://puck.nether.net/pipermail/cisco-voip/2005-February/002977.html , maybe isdn incoming display ?? |
| 23:02.15 | *** join/#asterisk VirTERM (~VirTERM@204.225.113.90) |
| 23:02.44 | bannerman | ulaw is the preferred codec for quality, isn't it? |
| 23:02.51 | Inv_Arp | bannerman: weird.. hmm old machine good power spply etc..? |
| 23:03.01 | MultiF | seems like you might have a dsp issue on the cisco |
| 23:03.10 | MultiF | what router is it? |
| 23:03.22 | invi_ | 1800 series |
| 23:03.22 | MultiF | what ios ver? |
| 23:03.26 | *** join/#asterisk helvetico (~helvetico@84-72-189-191.dclient.hispeed.ch) |
| 23:03.37 | bannerman | Inv_Arp: It's a decent machine. Athlon 2500, 512mb ram. |
| 23:03.45 | invi_ | ios 12.3(8)T5 |
| 23:04.24 | bannerman | I don't think it can be a resource issue, because while I have one bad call going on I can initiate a new call and it sounds great. |
| 23:04.34 | MultiF | on the same dsp? |
| 23:05.47 | VirTERM | I need some help with iax from realtime; anyone? |
| 23:06.42 | mikeh720 | vir, i'm guessing you tried the sip table for iax, as suggested on the wiki, and it doesnt work? |
| 23:06.56 | MultiF | We had some problem with certian ios vers. you might want to upgrade to 12.3(11)T |
| 23:07.05 | VirTERM | correct |
| 23:07.47 | *** join/#asterisk fugitivo (~ajf@201.255.109.28) |
| 23:07.56 | VirTERM | your sip table, did you just use the one from wiki? |
| 23:08.25 | invi_ | MultiF: "on the same dsp?" > can u elaborate |
| 23:08.30 | _asr | mikeh720: i set that on my serial interface (s3/0:23 in this case), no dice |
| 23:08.30 | mikeh720 | yes, but im only doing sip, not iax |
| 23:08.50 | VirTERM | I haven't tried sip yet... |
| 23:08.58 | MultiF | invi_: show voice dsp voice to show all dsp's |
| 23:09.01 | VirTERM | I will try sip then |
| 23:09.02 | mikeh720 | asr_ bummer, just keep googling, maybe an IOS issue too |
| 23:09.04 | *** join/#asterisk rvhi (~rv@66.175.65.89) |
| 23:09.08 | _asr | heh |
| 23:09.09 | mikeh720 | sip works fine via RT |
| 23:09.13 | _asr | trust me, i have |
| 23:09.14 | VirTERM | anything tricky? |
| 23:09.34 | MultiF | invi_: you also can do a "show voice dsp active" to show what dsp is in use on your calls |
| 23:09.36 | rvhi | i am using stable version, would like some suggestion of using mysql as the backend |
| 23:09.45 | VirTERM | I bet my problem is related to [general]...just like in voicemail |
| 23:09.57 | VirTERM | I will just remove the whole iax.conf |
| 23:10.13 | rvhi | if i want to query db directy, i'd like to use mysql() |
| 23:10.33 | rvhi | use res_config to load the extension config files from db |
| 23:10.51 | rvhi | use cdr-mysql to save the call log |
| 23:10.51 | VirTERM | rvhi:switch to HEAD; get it working; then send me tips on how to get iax working with realtime |
| 23:11.12 | rvhi | i don't want to use head, |
| 23:11.14 | invi_ | MultiF: no "show voice". this rt w/o voip |
| 23:11.38 | rvhi | i have been using stable for 6 months, no problem |
| 23:11.39 | mikeh720 | rvhi, we are talking about http://www.voip-info.org/wiki-Asterisk+RealTime |
| 23:11.43 | rvhi | don't want to mess it up |
| 23:11.51 | MultiF | invi_: oh, I think I misunderstood, you are just passing ip traffic through this router |
| 23:11.51 | rvhi | realtime is head only, right? |
| 23:12.09 | invi_ | MultiF: yep |
| 23:12.21 | mikeh720 | right, its not in 1.0.7 |
| 23:12.40 | MultiF | invi_: so is it the ame numbers you call that 1 is good and 1 is bad? |
| 23:12.46 | mikeh720 | im using realtime for voicemail and sip, havent done extensions yet |
| 23:12.52 | MultiF | invi_: ^same |
| 23:13.26 | mikeh720 | ran into an issue today where voicemail "options" such as attach=yes (send .wav email attachment of voicemail) aren't being picked up by realtime |
| 23:14.03 | mikeh720 | i still need to go search bugtracker to make sure it's not an open bug before i submit |
| 23:14.04 | invi_ | MultiF: what is ame? |
| 23:14.19 | invi_ | got it |
| 23:15.48 | helvetico | Can I throw a question in the middle? When using * as a media gateway only, with what equipment can I expect it to work? Does anyone use it with Cisco or any other vendor mediagateways? |
| 23:16.55 | husher | i just installed a te110p card into my * box, and now dialing in gets me a busy signal and dialing out gets 'cannot create zap channel'. any ideas what's wrong? |
| 23:17.35 | AmaDEE0_ | How do you connect asterisk to a PRI (Canada)? T1 card? |
| 23:17.51 | JunK-C | AmaDEE0_: with a t1 card. |
| 23:17.52 | BoRiS | T1 |
| 23:17.56 | AmaDEE0_ | Thanks. |
| 23:18.28 | JunK-C | TE110P is the newest t1 card from digium. |
| 23:19.10 | *** join/#asterisk jalsot (~tamas@195.56.44.83) |
| 23:19.33 | AmaDEE0_ | perfect. |
| 23:21.03 | mikeh720 | you definately want a digium card if you use asterisk, accept no imitators (tm). |
| 23:21.58 | AmaDEE0_ | JunK-C: Can a PRI be split into in/out lines? 1/2 in and the other 1/2 out? Or does each chanel do both? |
| 23:22.23 | mikeh720 | sure |
| 23:22.32 | JunK-C | each channel do both |
| 23:22.48 | mikeh720 | you will need your telco to not deliver calls on the channels that you only want to use for outbound tho |
| 23:22.49 | JunK-C | depends of ur RTPC from ur telco. |
| 23:24.32 | helvetico | T1 discussion: Digium deserves support for supporting the asterisk project. We have to be clear that the chipset on Digium or imitator producs is worth no more than 50$ but the service Digium provides is great and they always know more than we do since they feel the pulse from the chip makers and the * users. |
| 23:25.05 | PTG1234 | who invented the chip? |
| 23:25.30 | mikeh720 | exactly. digium cards are known to work with asterisk, and will get you the best support if they dont |
| 23:26.58 | helvetico | This chip is made at some OEM premise in asia for sure. Digium is at most designing the board. And the chipsets we talk about here are not very intelligent. The DSP is emulated in the CPU. |
| 23:27.02 | mikeh720 | chip manufacturers aside, it's the drivers that you should be most concerned with |
| 23:27.38 | BoRiS | ugh :-p |
| 23:28.31 | mikeh720 | right, the chips just generally need to be able to put on and take off 64Kb audio from the channel, asterisk does the codec stuff |
| 23:28.32 | helvetico | An Intel or Eicon board with E1 capacity costs 10 times as much but at the end of the day does the same with carrier grade reliability and support. |
| 23:28.54 | mikeh720 | helve, maybe less cpu overhead? |
| 23:30.29 | helvetico | Yes but an average * user has no multiple E1s and wannabe telecoms should be fine with it "as is". 1E1 with G729 to G711 conversion with a dual Xeon 2,8k used only 10% of my CPU when I ran a test. |
| 23:30.59 | timecop | helvetico: with how many channels in use? |
| 23:31.03 | timecop | full? |
| 23:31.34 | mikeh720 | even an (intel) dsp card would probably still require the same cpu usage with *, because asterisk would still send the same type of data to the PCI card |
| 23:32.05 | helvetico | There was a post earlier where someone stated to have "bad calls". I also had this phonomenon that a call can get distorted for a couple seconds. I had this mainly in conference mode. |
| 23:32.26 | mikeh720 | sounds like a jitter buffer problem |
| 23:33.32 | helvetico | From the sound I would tell that it is not an IP transmission problem. We basically on a LAN with 10 Megs capacity. |
| 23:34.04 | helvetico | It seems really a tilt of an isolated call. BUT it DID come back to normal after a few seconds. |
| 23:34.07 | mikeh720 | which would be an * configuration option, not necessarily a problem with the PCI card, but a debug of the call might tell |
| 23:34.13 | *** join/#asterisk jalsot (~tamas@abacus.eworldcom.hu) |
| 23:34.15 | invi_ | what does xten do to its packets to get priority in DSCP... * is missing it; its not tos=0xb8 |
| 23:34.55 | mikeh720 | does it set the DSCP bit to 1 in the ip header? |
| 23:35.03 | ManxPower | invi_: remember many providers reset the tos/dscp bits when packets enter their network. Bellsouth does this. |
| 23:35.21 | helvetico | invi_: very good point. |
| 23:35.26 | *** join/#asterisk Nebukadneza (~daddel9@i3ED6E539.versanet.de) |
| 23:35.30 | Nebukadneza | hi dudes ... |
| 23:35.40 | Nebukadneza | i need some help testing my iax ... |
| 23:35.54 | Nebukadneza | can someone please call phone1@nebuk.homelinux.org? :) |
| 23:35.57 | mikeh720 | nm, i'm being on crack today |
| 23:36.49 | invi_ | ManxPower: this my cisco & my ISP (VSAT) is largerly on QoS/VOIP network |
| 23:37.15 | helvetico | invi_: but I always stand on the point that backbone capacity is never an issue. It's like italian restaurant. You get pizza at so many places that you don't need to eat twice at the same place if you get a bad one. |
| 23:38.56 | helvetico | If your upstream provider gets crowded, warn him or eat somewhere else. And if you are of too small size to talk to the backbone directly, have your ISP do it for you. You are the customer and can influence his peering policy. The monopoly is history... |
| 23:40.10 | invi_ | helvetico: xten works like charm on this connection |
| 23:41.51 | helvetico | The xten lite i tried with * sucked beautifully. The debugging was full of RTP codec messages. Besides that configuration is another nightmare for people with less knowledge than an admin. Firefly open is up to now the best client I know. |
| 23:42.38 | invi_ | helvetico: pls, no pissing contests |
| 23:43.50 | helvetico | invi_: sorry. I give an opinion. Laws are made elsewhere. |
| 23:44.39 | helvetico | The real QoS crackers are 99$ CPE routers. |
| 23:44.50 | blitzrage | hey... someone know real quick what I have to set in my <mac>.cnf file for a 7960 to change the default gateway? |
| 23:44.58 | invi_ | helvetico: its about *'s packets not getting right priority on cisco routers |
| 23:44.59 | *** join/#asterisk MrBelvedr (~tt@ip68-227-209-110.dc.dc.cox.net) |
| 23:45.48 | helvetico | invi_: Are you talking from your site to the backbone then and not on the long-haul network? |
| 23:46.12 | *** join/#asterisk lesouvage (~lesouvage@cc341200-a.assen1.dr.home.nl) |
| 23:47.08 | husher | does anyone know what a "no d-channels avalible" error means? |
| 23:47.23 | invi_ | tried it on 3 diff cisco on 3 diff isp |
| 23:48.02 | ariel_ | husher, means no d-channel on a pri is active. |
| 23:48.32 | Nebukadneza | argh ... damned iax -.- |
| 23:48.34 | ariel_ | husher, it's either configuration problem or the telco is having issue. Is this a new setup? |
| 23:48.44 | lesouvage | Mailing to remote domains not supported My question is: how can I get the mail (for fax atttachement in e-mail) to not local domains supported. |
| 23:49.02 | lesouvage | First sentence is the message I got on screen |
| 23:49.09 | ariel_ | Nebukadneza, what are you trying to do? |
| 23:49.17 | husher | ariel_: yes and no, the box has been working, but i just put in the fe110p card, so that's a new setup |
| 23:49.24 | Nebukadneza | im just trying to get called by a iax client -.- |
| 23:50.06 | Nebukadneza | i thought if a friend of me got iaxcomm without a asterisk or voip provider it should work just with calling e.g. phone1@myip |
| 23:50.09 | ariel_ | husher, make sure you put the jumper on T1/pri and not E1 if your in the states, it ships with it on E1 |
| 23:50.31 | invi_ | helvetico: im talking about 3 *s sitting behind my 3 cisco's |
| 23:50.34 | ariel_ | Nebukadneza, get them an iaxtel account |
| 23:50.40 | helvetico | invi_, do you mean that the general network traffic to the default gateway (cisco 1,2,3) is eating all the bandwith and the cisco is not freeing up it quick enough if the * requires bandwith? |
| 23:50.44 | husher | ariel_: is jumper open E1 or T1? i couldn't tell |
| 23:50.46 | Nebukadneza | but ... why? |
| 23:50.53 | sneak | hey quick question - what's the lowest LD rates you guys have seen on wireline carriers? |
| 23:51.02 | ariel_ | jumped is E1 open T1 |
| 23:51.11 | Nebukadneza | it should be possible to do it just with the ip, right? |
| 23:51.17 | husher | ariel_: it's open |
| 23:51.39 | husher | ariel_: and that's what it should be, i am in the states |
| 23:51.49 | ariel_ | what are you settings? post them on http://pastebin.ca zaptel and zapata.conf for the device. |
| 23:51.58 | invi_ | helvetico: packets from * r not given right priority on cisco while packets from xten r |
| 23:52.39 | Wazb | want environment variables i need to set before i start installation of H323 |
| 23:53.16 | invi_ | helvetico: something missing in in packet from * in regards to DSCP |
| 23:53.34 | ariel_ | Nebukadneza, do you have a guest account setup on your system? |
| 23:53.47 | Nebukadneza | ariel_: no - i mean |
| 23:53.48 | invi_ | helvetico: & its not tos |
| 23:53.50 | husher | http://pastebin.ca/8568 |
| 23:53.56 | Nebukadneza | assuming you got iaxclient or something on your machine |
| 23:54.05 | Nebukadneza | you dont do any account setup |
| 23:54.08 | Nebukadneza | and call bla@myip |
| 23:54.37 | *** join/#asterisk TommyTheKid (~tommy@dsl-206-53-27-168.den.pcisys.net) |
| 23:54.41 | Wazb | what environment variables i need to set before i start installation of OpenH323 & Pwlib |
| 23:54.48 | helvetico | invi_, I'd say this would make a good bug report if you can document this properly. Probably the * developers never had bandwith constraints because it runs near the edge router and xten typically has because it runs in the middle of the LAN.y has because it runs on NATted LANS |
| 23:55.31 | *** join/#asterisk jayeola (~jayeola@dsl-80-43-20-58.access.as9105.com) |
| 23:55.39 | TommyTheKid | I have a "patch" (cvs diff -u ?) that allows asterisk to compile on Solaris 10 x86 .. is anyone interested.. it was only one very minor code change (ifdef) and lots of makefiile edits |
| 23:56.09 | ariel_ | Nebukadneza, you need to have a guest setting and you dial like guest@IP addresss/exten? |
| 23:56.27 | *** join/#asterisk mishehu (mishehu@cshells.shavedgoats.net) |
| 23:56.36 | TommyTheKid | .. ofcourse I really haven't done anything with it.. but it did make cleanly and run :) |
| 23:57.18 | Nebukadneza | im not dialin - its just a incoming call ... isnt there a client which is able to intitiate calls without registering on a server? |
| 23:57.23 | husher | ariel_: see anything wrong with my config? |
| 23:58.22 | *** join/#asterisk Achillies (~nick@165.252.43.77) |
| 23:58.45 | *** join/#asterisk gus (~gus@46.33.233.220.exetel.com.au) |
| 23:58.51 | ariel_ | husher, your set it up for an e1 channel => 1-3 ; Set this to 1-15,17-31 for E1 |
| 23:59.02 | ariel_ | you said you are in the states? |
| 23:59.16 | ariel_ | channel 1-23 are b and 24 is d |
| 23:59.19 | husher | ariel_: yes, i am |
| 23:59.23 | helvetico | invi_, and if you asked a cisco god to configure the router that all traffic from the port where *is sitting getts priority over other LAN ports. |
| 23:59.43 | gus | so does the CVS repository usually have files/directories manaully moved around? |
| 23:59.51 | husher | ariel_: i've tried lots of combinations there, i set it up that way because only 3 lines are actually active |