00:01.43 | visik7 | how can I solve this : http://pastebin.ca/1140 |
00:03.24 | file2 | old code is bad mmmk |
00:04.47 | *** join/#asterisk JRB (~jbenson@genpubad.gotadsl.co.uk) |
00:04.49 | visik7 | so it's better to leave pbx_*console.so away ? |
00:05.10 | Moc | is there a cli command to register and unregister from a sip connection |
00:05.14 | Moc | É |
00:05.16 | Moc | ? |
00:05.59 | visik7 | is the client that register a sip connection to a server |
00:06.13 | AgiNamu | can someone call a 866 number for me? |
00:06.18 | visik7 | not the opposite |
00:06.20 | AgiNamu | PM me if so :) |
00:06.23 | Moc | well * also connect to other server |
00:09.33 | visik7 | Moc understand |
00:10.32 | bkw_ | la la la la |
00:11.24 | *** part/#asterisk TEKjacob_ (~tekjacob@c7.e3bccf.client.atlantech.net) |
00:11.51 | brc_ | lala? |
00:12.05 | brc_ | znoG, no. |
00:12.08 | *** part/#asterisk ahampton78759 (~ahampton@rrcs-24-173-78-2.sw.biz.rr.com) |
00:12.33 | *** join/#asterisk m-00kie (~r3l34s3d@pcp09300018pcs.arlngt01.va.comcast.net) |
00:14.07 | ariel_ | Lets see who might be able to help me configure a Cisco 7960 that is behind a Nat to talk to an Non natted Asterisk? |
00:15.31 | habakuk | survey. anyone be interested in DID's from the San francisco bay area? I can provide IAX or sip |
00:15.50 | paulc | ariel_ I have that config working |
00:17.03 | ariel_ | Can you give me a peak at your setup files like the one from the Cicso phone Please? |
00:17.55 | pfn | habakuk is that so, pricing? |
00:18.07 | paulc | ariel_ I can email you my SIPDefault and SIPxxxx.cnf if you like? |
00:18.10 | pfn | and TOS? |
00:18.29 | ariel_ | paulc, sure arielb27@gmail.com thank you very much. |
00:18.40 | habakuk | pfn 5$ / DID |
00:19.03 | *** join/#asterisk imcdona (~imcdona@sbi-24-177-181-60.mtv.al.charter.com) |
00:19.22 | loko | Anyone here offering the 412 (Pittsburgh) DID |
00:19.39 | pfn | habakuk and what are the terms of that? unlimited incoming, etc. etc. ? |
00:19.45 | pfn | payment? website? quantity? |
00:19.46 | pfn | hmm |
00:19.54 | pfn | website as in what's your site |
00:21.13 | habakuk | pfn : email sales@fonomni.com, website: www.fonomni.com, and yes unlimited incoming |
00:22.18 | pfn | what codecs do you have? |
00:23.44 | slePP | anyone doing spa-2000 provisioning? |
00:23.46 | slePP | via https/tftp |
00:23.52 | habakuk | pfn: supporting all at this point. working on g.729 |
00:24.05 | paulc | ariel_ sent! although outlook was being wank about .cnf files so I renamed to .txt |
00:24.59 | ariel_ | thanks I got them. |
00:30.45 | file2 | mmm Enya |
00:31.12 | paulc | Enya? Really? |
00:31.15 | *** join/#asterisk WifiFred (~wififred@198.231.65.5) |
00:31.18 | file2 | good music. |
00:31.28 | paulc | Sail away sail away sail away... |
00:31.46 | paulc | or "c'est le way, c'est le way, c'est le way".. depending on if you're ripping the piss or not.. |
00:31.59 | paulc | current favourite: Blinded By The Lights by The Streets |
00:32.50 | file2 | it's soothing music |
00:32.53 | file2 | helps me think and clear my mind |
00:32.54 | *** join/#asterisk EarlGrey (~EarlGrey@earlgrey.user) |
00:34.36 | *** join/#asterisk twisted (~twisted@twisted.active.supporter.pdpc) |
00:34.36 | *** mode/#asterisk [+o twisted] by ChanServ |
00:35.13 | icebalm | "ripping the piss" |
00:35.19 | icebalm | interesting expression |
00:35.31 | twisted | uhm, sounds extremely painful... like... an std even. |
00:35.56 | icebalm | or like a kidney stone |
00:36.25 | EarlGrey | ripping the piss = uk slang = more extream version of taking the piss |
00:36.29 | ariel_ | kidney stone's argh |
00:36.37 | brc_ | ahhhhhhhhh |
00:36.44 | icebalm | taking the piss? |
00:37.06 | icebalm | why don't you just say pissing |
00:37.29 | tzanger | haha |
00:37.31 | gambolputty | You don't take shit you leave it |
00:38.13 | icebalm | yeah I don't understand this whole line of expressions, and I'm a native english speaker |
00:38.20 | file2 | Flora's secret! |
00:38.57 | *** join/#asterisk AgiNamu (~zzzs@4.79.150.34) |
00:39.08 | ariel_ | Funny when I was in the service we would say we need to drain the vain.... |
00:39.36 | icebalm | vein |
00:39.57 | twisted | i usually say i'm gonna go take a shit |
00:39.59 | PhilM | I need to drain my lizard |
00:40.02 | twisted | but i must start correcting myself |
00:40.07 | twisted | turn some heads |
00:40.30 | twisted | OOOOH |
00:40.39 | twisted | I need that samuel L jackson soundbyte for a 7960 ringer |
00:40.56 | twisted | "If you're going to shoot me, go ahead and shoot me! But i HAVE TO ANSWER THAT PHONE!" |
00:41.09 | icebalm | lol |
00:41.20 | paulc | LOL.. that's porno cheesey but kinda funny |
00:41.24 | twisted | paulc |
00:41.24 | twisted | yeah |
00:41.34 | twisted | it's a line from die hard with a vengance |
00:42.05 | *** join/#asterisk isamar (isamar@p8131-ipadfx21sasajima.aichi.ocn.ne.jp) |
00:42.22 | paulc | evening all :-) |
00:42.25 | paulc | what's shaking twisted? |
00:42.54 | file2 | Enya! |
00:43.10 | file2 | Enya Enya Enya. |
00:43.43 | Muckl | anyone in here using asterisk to register => a SIP account from sipgate.de? |
00:43.50 | paulc | she's almost as bad as Celine Dione |
00:43.50 | icebalm | "If you're going to shoot me then you go ahead and you shoot me, but I have to answer this phone." |
00:43.57 | *** join/#asterisk algorithmn (~none@ool-182f915a.dyn.optonline.net) |
00:43.57 | paulc | slePP - chuck me a Sipura and I'll let you know ;-) |
00:44.14 | file2 | no sight for you! |
00:44.30 | paulc | omg you're one evil little mofo sometimes! |
00:44.31 | ariel_ | oh no the radio is playing Shakira... got to get another station on. |
00:44.45 | slePP | paulc: heh. |
00:44.59 | file2 | SO who wants to see the latest tracking on my PAP2-NA? |
00:45.10 | *** join/#asterisk isamar (isamar@p8131-ipadfx21sasajima.aichi.ocn.ne.jp) |
00:45.11 | ariel_ | slePP, I guess no one is doing the mass setups right now. |
00:45.17 | isamar | Hi folks |
00:45.18 | *** join/#asterisk lancey (Shady@support.net1.cc) |
00:45.19 | algorithmn | why isn't $agi -> exec(dial,UN@HOST\${NUM}); working?? |
00:45.21 | slePP | ariel_: i wanna tftp boot my new pap2-na :> |
00:45.22 | file | it's not good tracking |
00:45.23 | isamar | anyboyding usingnufone here? |
00:45.32 | slePP | isamar: what's nufone? |
00:45.43 | mlh407 | I am |
00:45.47 | lancey | guys |
00:45.55 | lancey | i seem to be having trouble with H323 |
00:45.56 | file | Sep 27 2004, 1:00PM. CONCORD, ON, CA: THE PACKAGE IS DELAYED DUE TO EMERGENCY CONDITIONS BEYOND UPS' CONTROL |
00:45.57 | isamar | mlh407... |
00:46.01 | lancey | could i call someone of you? |
00:46.05 | lancey | to test it? |
00:46.08 | isamar | slePP: www.nufonet.net |
00:46.10 | isamar | ops |
00:46.14 | isamar | www.nufone.net |
00:46.18 | slePP | oh. |
00:46.18 | mlh407 | lancey: I am lost |
00:46.18 | slePP | neat. |
00:46.27 | isamar | mlh407... trying to call prefix 380 |
00:46.28 | slePP | file: nice. |
00:46.32 | isamar | ukraine... |
00:46.38 | file2 | isn't it just peachy? |
00:46.45 | isamar | not getting... which dial plan do you use for other countries but US? |
00:47.00 | mlh407 | I am only using it for incomming class |
00:47.00 | slePP | emergency conditions |
00:47.02 | mlh407 | *calls |
00:47.03 | slePP | ie: it's in florida. sorry |
00:47.08 | paulc | file: what's that all about then? emergency conditions? like.. a lack of donuts for airport freight workers? or..? |
00:47.10 | lancey | <mlh407> lancey: I am lost? |
00:47.12 | isamar | ok |
00:47.16 | lancey | * ariel_ does not use h323 |
00:47.22 | lancey | ariel_ i don't need someone with h323 |
00:47.25 | lancey | i just need a phone number |
00:47.27 | lancey | POTS |
00:47.27 | file2 | algorithmn: $AGI->exec('Dial',UN@HOST\${NUM}); |
00:47.38 | lancey | anyone willing to help me? |
00:47.38 | ariel_ | lancey, where? |
00:47.40 | file2 | paulc: nobody knows! |
00:47.45 | isamar | lancery; wasup? |
00:47.46 | lancey | ariel_ doesn't matter |
00:47.49 | file2 | I think it's a massive conspiracy |
00:47.50 | lancey | anywhere but bulgaria |
00:48.13 | paulc | lancey: talking yellow pages: +1 604 299 9000 |
00:48.18 | AgiNamu | file2, dont you need to specify IAX or SIP? |
00:48.18 | twisted | paulc, not that |
00:48.31 | lancey | paulc: i love you :) |
00:48.33 | file2 | AgiNamu: it's his AGI, I'll let him figure it out |
00:48.37 | paulc | twisted: not? |
00:48.43 | AgiNamu | heh |
00:48.43 | paulc | lancey: I'm flattered ;) |
00:48.45 | twisted | paulc, nah. |
00:48.46 | lancey | :)) |
00:48.51 | paulc | uh.. not what? |
00:48.56 | twisted | [19:42] <paulc> what's shaking twisted? |
00:49.02 | paulc | ah |
00:49.02 | *** join/#asterisk hastur (~hasturr@sobek.7g0.net) |
00:49.04 | ariel_ | lancey, get your self one from: http://www.kallfree.com/ it's free |
00:49.08 | paulc | lagged reply ;-) |
00:49.09 | file2 | people can't learn by being given things, they have to discover them on their own! |
00:49.15 | twisted | paulc, yeah, mentally lagged. |
00:49.52 | paulc | I get like that.. |
00:49.58 | paulc | I got to write a report but it's kind of a headfuck |
00:50.21 | file2 | you IRC like you talk. |
00:50.31 | paulc | I always type like I talk :-p |
00:50.32 | twisted | lol |
00:50.43 | paulc | email, MSN, IRC... s'how I am :) |
00:50.56 | Muckl | whats the IVR number again? 1800 555 tell? |
00:51.02 | algorithmn | i've tried a few ways |
00:51.12 | *** join/#asterisk TEKjacob_ (~tekjacob@c7.e3bccf.client.atlantech.net) |
00:51.29 | algorithmn | thinking about ditching perl and using c |
00:51.57 | file | exxxxxxxxxxxxcellent |
00:52.03 | algorithmn | it is creating errors in logging |
00:52.18 | AgiNamu | YES :D |
00:52.22 | AgiNamu | Use the C API :) |
00:52.34 | algorithmn | i was thinking it was a perl api specific glitch |
00:52.38 | twisted | sing |
00:52.39 | WifiFred | song |
00:52.40 | twisted | sing a song |
00:52.43 | twisted | make it loud |
00:52.45 | twisted | make it strong |
00:52.46 | twisted | sing |
00:52.47 | WifiFred | song |
00:52.48 | twisted | of good things |
00:52.48 | twisted | not bad |
00:53.01 | TEKjacob_ | Hey all... I have been messing around with the 7960 Cisco phone... Got it working (thanks to you all) Although I am having three problems. 1. Cant telnet (.cfg file seems to be set right) 2. won't display custom logo 3. wont pickup extra ringers |
00:53.05 | TEKjacob_ | any ideas? |
00:53.10 | *** join/#asterisk Vco (~Vco@S0106080020aa7650.wp.shawcable.net) |
00:53.21 | Vco | Hello all.... |
00:53.29 | twisted | TEKjacob_, sounds likt the tftp isn't working right |
00:53.36 | file | Vco: Hello. |
00:53.45 | TEKjacob_ | yeah, but I can't find any logs... |
00:54.54 | twisted | of course not |
00:55.00 | algorithmn | agi: have you ran across this problem if i may ask? |
00:55.04 | Vco | Any */mpg123/alsa/musiconhold efficianados around? |
00:55.22 | lancey | weee |
00:55.24 | file | Vco: verify you have version 0.59r if you're having MOH probs |
00:55.25 | lancey | it's working |
00:55.25 | lancey | :) |
00:55.40 | lancey | anyone knows if chan_h323 has been modified lately? |
00:55.45 | lancey | i updated 1 hour ago |
00:55.51 | lancey | and now everything seems to run just fine |
00:56.20 | Vco | yea..0.59r is on the box.. |
00:57.08 | file | h323 was last updated: September 21st |
00:57.16 | lancey | thank you, file! |
00:57.21 | lancey | JerJer's done a nice work! |
00:57.24 | lancey | i love him too :) |
00:57.35 | lancey | bkw_ what's up? |
00:57.38 | Vco | shows as running on the * console when connecting to music on hold, audio card is working under alsa...grandstream is behind a nat, voice and everything works fine...just zero audio output from moh |
00:57.40 | file | bkw_: you and your silly screaming |
00:57.42 | paulc | pull yourself together! |
00:57.56 | Vco | been scouring wiki and google and whatever else i can find. |
00:57.59 | paulc | he screams like a gurrrrl eh? ;-) |
00:58.03 | Vco | losing my mind at this point.. |
00:58.05 | file | Vco: you verified there's no rogue copies of an old mpg123 on the system? |
00:58.34 | *** join/#asterisk Vulture- (~Vulture@247.131.vbnet.net) |
00:58.38 | Vulture- | urg damn internet |
00:58.44 | lancey | Vulture- |
00:58.48 | Vco | i'll doublecheck...clear and do another make isntall to be sure.. |
00:59.00 | file | Vco: well your distro may have installed mpg123 in another location |
00:59.08 | algorithmn | bkw: i've been thinking of an idea for you |
00:59.12 | bkw_ | or mpg321 with a synlink to mpg123 |
00:59.19 | bkw_ | algorithmn and that is? |
00:59.22 | Vco | no..it is mpg123 |
00:59.23 | file | indeed |
00:59.40 | lancey | guys |
00:59.42 | algorithmn | mind if i talk in... ahhum, private? |
00:59.45 | lancey | i need to test H323 2-way |
01:00.04 | lancey | anyone willing to speak with me for just 10 seconds |
01:00.24 | file | I don't do H323. |
01:00.27 | lancey | no |
01:00.31 | bkw_ | I dont either |
01:00.33 | AgiNamu | IS it possible to send 2 updates in Mysql at once (via real_query)? |
01:00.36 | lancey | i'll call you on a POTS |
01:00.38 | bkw_ | even though its a gay protocol.. I wont do it! |
01:00.43 | lancey | need to test the gateway |
01:00.45 | file | I don't have a DID at this time anyway |
01:00.48 | Vulture- | I don't do fatchicks... I mean H323 |
01:00.58 | lancey | yeah yeah i know H323 is crappy |
01:01.04 | lancey | but one of my providers uses it |
01:01.09 | lancey | i can't do anything about that |
01:01.10 | lancey | :( |
01:01.29 | slePP | bkw_: do you have the sipura provisioning compiler? :> |
01:01.45 | bkw_ | slePP go talk to sipura they will give it to ya |
01:01.47 | bkw_ | if you ASK for it |
01:01.53 | bkw_ | if not let me know i'll smack em around |
01:02.03 | bkw_ | how long ago? |
01:02.07 | slePP | 30 seconds |
01:02.09 | lancey | :) |
01:02.16 | bkw_ | god damn impatient fucker |
01:02.18 | bkw_ | haha |
01:02.20 | Vulture- | lol |
01:02.20 | slePP | :> |
01:02.23 | slePP | yessir! |
01:02.27 | slePP | you should see me in bed, though |
01:02.33 | slePP | patient as they cum |
01:02.33 | slePP | er. come. |
01:02.38 | slePP | damn freud and his slips |
01:02.49 | *** join/#asterisk Ferrari355 (~dcox@12-221-104-25.client.insightBB.com) |
01:02.56 | slePP | bkw_: wanna provision my pretty new PAP2-NA |
01:03.06 | slePP | but you need their doodad to do so, so now i'm waiting. but i like how it can do HTTPS w/ client cert auth |
01:03.17 | Vulture- | sounds like slePP is propositioning bkw_ |
01:03.28 | slePP | so? :> |
01:03.31 | Vulture- | lol |
01:03.32 | paulc | slePP's such a flirty flirt |
01:03.45 | lancey | slePP: are the PAP2's unlocked now? |
01:04.06 | slePP | uhm |
01:04.08 | file | PAP2 is not unlocked, PAP2-NA is unlocked... it's the little things |
01:04.09 | slePP | the PAP2-NA always has been |
01:04.11 | slePP | but they're under recall |
01:04.20 | file | so if anybody asks, we don't have any |
01:04.24 | slePP | actually, not 'recall' |
01:04.34 | slePP | rather 'GIVE IT BACK AND PRETEND IT DOESN'T EXIST' lockdown |
01:06.20 | *** join/#asterisk lancey (Shady@support.net1.cc) |
01:06.22 | lancey | whoops |
01:06.25 | lancey | our telco rulez |
01:06.36 | twisted | slePP, yep |
01:06.41 | twisted | slePP, but we have a deal with staples |
01:06.44 | twisted | they keep them for us |
01:06.46 | twisted | so we can buy them |
01:06.48 | twisted | :P |
01:06.49 | twisted | brb |
01:07.29 | pfn | well, if they sell the pap2-na, they probably lose a crapload of money on it |
01:07.52 | lancey | what's it's price anyways? |
01:07.58 | pfn | $50 |
01:08.21 | mlh407 | is it worth 50 bucks? |
01:08.25 | pfn | yes |
01:08.26 | lancey | $50??? |
01:08.31 | lancey | serious? |
01:08.34 | pfn | considering a sipura 2000 costs $100 |
01:08.37 | pfn | most definitely |
01:08.41 | lancey | and an ATA costs $150 |
01:08.48 | lancey | yeah it's worth the money! |
01:08.49 | mlh407 | but does it sound as good as the sipura you get what you pay for |
01:08.50 | pfn | the sipura 2000 is an ATA |
01:08.59 | lancey | i meant Cisco |
01:09.01 | pfn | the pap2-na *is* a spa-2000 internally |
01:09.01 | lancey | ATA-186 |
01:09.09 | lancey | hah |
01:09.12 | *** join/#asterisk Enigma81 (~trillian@pcp02587377pcs.shlb1201.mi.comcast.net) |
01:09.32 | pfn | the reason behind the cheapness of the pap2 is because of provider subsidies |
01:09.44 | *** join/#asterisk Enigma81 (~trillian@pcp02587377pcs.shlb1201.mi.comcast.net) |
01:10.10 | slePP | our cost was $68 |
01:10.12 | slePP | CAD |
01:10.17 | pfn | right, $10 US |
01:10.19 | slePP | wow. sipura's response was fast |
01:10.24 | pfn | ? |
01:10.28 | slePP | :> |
01:10.35 | slePP | heh. pfn, close. |
01:10.37 | slePP | more like $40 |
01:10.39 | slePP | :> |
01:11.24 | lancey | guys |
01:11.34 | lancey | what's your opinion about GrandStream's phones |
01:11.38 | lancey | BT-100 i think it was? |
01:11.48 | pfn | in general, I think the opinion is very poor |
01:11.54 | lancey | hm |
01:11.57 | pfn | step up a little and buy a cisco 7905 or polycom ip 300 |
01:12.04 | lancey | what's the price |
01:12.11 | cypromis | in general it's a nopinion |
01:12.20 | pfn | they're around $115 or so |
01:12.27 | lancey | hm |
01:12.28 | lancey | pricey |
01:12.37 | pfn | the BT100 is already $70 |
01:12.39 | pfn | and sucks ass |
01:12.43 | lancey | mhm |
01:12.50 | pfn | I'd rather spend a little more and get some quality |
01:12.52 | paulc | twisted: you got any more PAP2's up for grabs? |
01:12.55 | lancey | but average people's salary out here is $100 |
01:13.04 | lancey | so it better be lower :) |
01:13.15 | twisted | paulc, up for grabs? |
01:13.18 | pfn | where is out here |
01:13.18 | twisted | i never had any up for grabs |
01:13.28 | lancey | pfn : Bulgaria |
01:13.29 | mlh407 | twisted: do you have any for sale |
01:13.34 | twisted | no |
01:13.46 | cypromis | hmmm |
01:13.47 | lancey | i'm currently arguing on ICQ |
01:13.51 | cypromis | bulgaria is quite close to here |
01:13.51 | cypromis | :) |
01:13.54 | paulc | misread maybe - thought you meant you had your local staples holding a few more for you |
01:14.04 | twisted | well |
01:14.04 | twisted | we do |
01:14.07 | lancey | with one man willing to make me believe SIP is bullshit and H323 is the go! |
01:14.08 | twisted | but we're not selling them |
01:14.10 | lancey | :))))))))) |
01:14.18 | cypromis | h word |
01:14.22 | pfn | heh |
01:14.25 | pfn | h323, bleh |
01:14.26 | paulc | do your Staples peeps wanna chuck one my way? |
01:14.31 | twisted | doubt it |
01:14.36 | twisted | :P |
01:14.57 | *** part/#asterisk Ferrari355 (~dcox@12-221-104-25.client.insightBB.com) |
01:15.08 | pfn | just get a vonage branded one |
01:15.09 | pfn | and hack it |
01:15.15 | Enigma81 | Anyone know of any good IAX providers (Termination AND Origination) that can provide DID's in Michigan, New York, Orlando and Las Vegas? |
01:15.27 | pfn | well, in michigan there's nufone |
01:15.34 | pfn | in those other markets, voicepulse connect probably |
01:15.47 | lancey | is VoicePulse any good? |
01:15.53 | file | they be everywhere... |
01:15.55 | VoiceLynx | is the hack for the PAP2 out on the web somewhere? |
01:16.00 | pfn | I've been very disappointed with them in the past |
01:16.12 | file | slePP: you should make the provisioning software appear over my way too |
01:16.34 | lancey | pfn : i do know so also |
01:16.47 | Enigma81 | agreed, I've heard very bad things about VoicePulse |
01:17.14 | pfn | voicelynx my understanding is that you intercept the tftp traffic |
01:17.20 | twisted | Enigma81, they've improved quite a bit |
01:17.47 | pfn | and provide your own firmware and configuration for the pap2 |
01:17.55 | *** join/#asterisk Vco- (~Vco@S0106080020aa7650.wp.shawcable.net) |
01:17.56 | VoiceLynx | pfn: presumably feed it a suitable file... is the suitable file easily available |
01:17.57 | pfn | once you do that initially, it will no longer seek out vonage on boot |
01:18.08 | pfn | probably yes |
01:18.09 | twisted | anywho |
01:18.10 | twisted | i'm off for a bit |
01:18.32 | Vco- | well...mpg123 is all fresh..still no moh joy... |
01:18.33 | slePP | file: what prov software? ;> |
01:19.14 | cypromis | mpg is overrated |
01:20.10 | cypromis | why not just use gsm/ulaw/alaw files |
01:20.15 | cypromis | and get rif of compression ? |
01:20.19 | *** join/#asterisk kuj (~kuj@c-67-165-241-16.client.comcast.net) |
01:22.23 | Vco | hmm..should mp3/mpg show up under "show file formats" ? |
01:23.00 | *** join/#asterisk ipdeman (~Gary@cpe-maint-port0.netpathway.com) |
01:23.22 | cypromis | vco: get asterisk-addons |
01:23.28 | cypromis | it contains format_mp3 |
01:23.32 | cypromis | and it will show up than |
01:23.32 | Vco | oooh |
01:23.38 | *** part/#asterisk TEKjacob_ (~tekjacob@c7.e3bccf.client.atlantech.net) |
01:24.17 | Vco | thanks..i'll give that a shot |
01:25.18 | slePP | in sip.conf |
01:25.25 | slePP | do you just append to the end: include => something.conf? |
01:25.40 | kram | hi file :) |
01:25.42 | kram | hi slepp! |
01:25.53 | slePP | hey kram :> |
01:26.56 | cypromis | slePP: #include something.conf |
01:27.25 | cypromis | incude => something.conf would be looking for [something.conf] context to inlude |
01:28.07 | slePP | k |
01:28.15 | slePP | talking about sip.conf, of course |
01:28.22 | cypromis | hmmm |
01:28.24 | cypromis | dunno |
01:28.29 | cypromis | my example is from extensions.conf |
01:28.41 | slePP | oh. okay. that's annoying. sipura's thing outputs an xml config, but it won't make it into binary... wtf.. |
01:29.33 | *** part/#asterisk kuj (~kuj@c-67-165-241-16.client.comcast.net) |
01:29.34 | *** join/#asterisk IcEyOnE (~email@67-51-232-21.dsl1.glv.ny.frontiernet.net) |
01:29.45 | *** join/#asterisk invidiaguy (~Brandon@207.93.213.102) |
01:29.51 | blankman | hey kram, just want to finish what we started yesterday. |
01:30.14 | lancey | hm |
01:30.26 | lancey | anyone knows anythung about Sybase? |
01:30.35 | IcEyOnE | could i get some help with freeworlddialup and asterisk?..i cant dialout from my x-lite phone to any freeworlddialup..i'm getting some codec errors on the console |
01:30.42 | blankman | lancey, some from a while ago .... |
01:30.49 | PatrickDK | iceyone, use ulaw |
01:30.49 | paulc | iceyone: what do the error messages say? |
01:30.55 | IcEyOnE | hold on a sec |
01:31.11 | lancey | huh |
01:31.12 | slePP | and now we wait for it actually request a config. it takes its sweet time |
01:31.12 | IcEyOnE | WARNING[143763456]: rtp.c:1392 ast_rtp_bridge: codec0 = 10 is not codec1 = 4, cannot native bridge. |
01:31.13 | lancey | 04:31 am here |
01:31.15 | lancey | i should go |
01:31.17 | lancey | home |
01:31.24 | lancey | bye guys |
01:31.31 | IcEyOnE | hmm so disallow all and allow ulaw? |
01:32.09 | cypromis | hmmm how to pay a lot for deadlocks |
01:32.11 | cypromis | sybase |
01:32.45 | PatrickDK | you don't have todisallow all |
01:32.49 | PatrickDK | but you have to allow ulaw |
01:33.05 | IcEyOnE | if i use G711u on x-lite it appears to connect but i cant hear anything or send anything..not sure if thats somethign totally different but i've been just using gsm for now |
01:33.08 | IcEyOnE | ok |
01:33.26 | PatrickDK | dunno |
01:33.31 | PatrickDK | xlite works fine on my system |
01:33.37 | PatrickDK | you behind a firewall? |
01:33.38 | PatrickDK | or nat? |
01:33.48 | IcEyOnE | well actually its just if i go out to freeworlddialup |
01:33.53 | IcEyOnE | behind smoothwall |
01:33.56 | IcEyOnE | so..yes :p |
01:34.00 | PatrickDK | that's probably the problem |
01:34.10 | PatrickDK | fix the firewall for voip |
01:34.41 | IcEyOnE | hmm it shouldnt be blocking it..unless its incoming traffic |
01:34.56 | *** join/#asterisk invidiaguy (~Brandon@207.93.213.102) |
01:35.06 | cypromis | it is incoming traffic |
01:35.45 | IcEyOnE | ah..you know the ports by any chance? :s |
01:38.01 | cypromis | for sip ? |
01:38.08 | cypromis | why not connect to fwd by iax |
01:38.13 | cypromis | will save you some trouble |
01:38.18 | cypromis | cause t will be only one port |
01:38.20 | *** part/#asterisk algorithmn (~none@ool-182f915a.dyn.optonline.net) |
01:38.43 | icebalm | whats a really good free soft phone :P |
01:38.48 | Chuji | xlite |
01:38.56 | icebalm | cool thanks |
01:39.06 | Chuji | Although I wouldn't say "really good" and soft phone in the same sentence |
01:39.24 | IcEyOnE | hmm didnt know you could connect to them via iax |
01:39.25 | icebalm | well, as good as possible for a soft phone |
01:39.26 | IcEyOnE | i'll give it a try =) |
01:39.27 | icebalm | :D |
01:39.39 | AgiNamu | Where can I get Canadian 800's |
01:39.39 | Chuji | Yeah, there you go |
01:40.12 | slePP | i should reboot and put my new soundcard in now |
01:40.22 | icebalm | is it a good sound card |
01:40.35 | slePP | Sound Blaster PCI 128 |
01:40.36 | slePP | nothing fancy |
01:40.38 | Chuji | slePP : See if it's hot swappable |
01:40.39 | slePP | just need it for my tv tuner |
01:40.39 | Chuji | haha |
01:40.44 | slePP | Chuji: been there, done that :> |
01:40.44 | paulc | popquiz on wikis (cos you're a knowledgeable bunch!) : Any favourites/recommendations for a PHP based wiki system? I've played with WakkaWiki, like it a lot - any others I should play with? |
01:41.03 | Chuji | Tiki Wiki? |
01:41.14 | pfn | I don't like php |
01:41.15 | slePP | k. shutdown time |
01:41.16 | pfn | thus I use twiki |
01:41.35 | AgiNamu | Canada can't call USA 800s eh? |
01:41.38 | paulc | tiki wiki = twiki = same as voip-info.org? |
01:41.42 | pfn | no |
01:41.44 | icebalm | AgiNamu: not all of them no |
01:41.45 | paulc | AgiNamu: They can if they're enabled for that feature |
01:41.46 | pfn | tiki wiki is voip-info |
01:41.49 | pfn | twiki is different |
01:41.55 | pfn | tiki wiki uses php |
01:41.57 | cypromis | AgiNamu: canada 1800 is expensive |
01:42.02 | cypromis | like 7.5 cents or something |
01:42.10 | paulc | but it has to be PHP5 or something I think.. |
01:42.27 | paulc | I'll check out twiki.. easy to use for newbies? (not me - I'm thinking of the people I want to give this to) |
01:42.35 | pfn | twiki is easy to us |
01:42.36 | pfn | e |
01:42.51 | pfn | although it's kinda overwhelming with the information overload |
01:42.56 | pfn | tiki wiki is probably easier |
01:42.57 | icebalm | expensive? 7.5 cents is cheap as hell |
01:43.01 | pfn | since it has all the googaw gadgets |
01:43.12 | pfn | 7.5c/min is cheap? |
01:43.19 | icebalm | yeah, for canada it is |
01:43.29 | AgiNamu | huh? |
01:43.38 | AgiNamu | oh oh for 800 |
01:43.44 | AgiNamu | 7.5 c?? WTF |
01:43.45 | AgiNamu | wow |
01:43.48 | AgiNamu | that's f*ing crazy |
01:43.54 | AgiNamu | any reason why? |
01:44.26 | paulc | pfn: cheers - I'm off to play :) |
01:44.56 | cypromis | yeah |
01:44.58 | cypromis | bell canada |
01:45.00 | cypromis | blame canada |
01:45.07 | cypromis | lol |
01:45.39 | AgiNamu | haha |
01:47.19 | pointer-gaim | np |
01:47.23 | pointer-gaim | err, wrong window |
01:47.52 | ariel_ | blame canada sound like robin williams is around someplace. |
01:50.22 | *** join/#asterisk Vco (~Vco@S0106080020aa7650.wp.shawcable.net) |
01:53.09 | ariel_ | Ok it's family time. See you all in the morning. |
01:53.37 | EarlGrey | evening all |
01:54.50 | *** join/#asterisk rollotomnasi (~trillian@delmar-209-137-161-171-dsl.cavtel.net) |
01:55.40 | cypromis | morn |
01:56.23 | EarlGrey | :) what part of the world you in now cypromis? |
01:58.24 | *** join/#asterisk slePP (~slepp@216.123.201.19) |
01:59.03 | cypromis | still poland |
01:59.09 | cypromis | saturday germany |
01:59.13 | cypromis | monday probably nl |
01:59.14 | cypromis | :) |
02:01.51 | EarlGrey | you get around a bit :) |
02:02.12 | *** join/#asterisk L|NUX (linux@202.63.192.42) |
02:02.45 | EarlGrey | got my asterisk box connecting to another asterisk box just having problems getting a phone to talk to it |
02:09.50 | *** join/#asterisk Fgravato (frankie@ool-44c02950.dyn.optonline.net) |
02:12.26 | Godsey | we're having the hardest time getting broadvoice to work :) |
02:12.40 | Godsey | we called support and said not even the web based active/x version works |
02:12.49 | Godsey | they said that requires a hardware ATA to function :) |
02:14.22 | Godsey | I think broadvoice sucks |
02:14.23 | Godsey | :P |
02:14.37 | icebalm | broadvoice, narrowbrain |
02:15.13 | bkw_ | twisted 996 |
02:15.26 | Godsey | bkw: have you used broadvoice? |
02:15.33 | bkw_ | no |
02:15.39 | bkw_ | why oh why would I ever do that? |
02:15.57 | Godsey | :P |
02:16.00 | bkw_ | why do I need broadvoice? |
02:16.00 | bkw_ | hahah |
02:16.15 | Godsey | our pris are DID only, not DIOD :) |
02:16.15 | bkw_ | I use at most 120 min on my landline |
02:16.17 | bkw_ | its really lame |
02:16.31 | Godsey | so we're kinda afraid of what our bill is going to be after dialing ou a few times :) |
02:16.36 | bkw_ | I work in phones but I hate to talk on the phone |
02:16.38 | bkw_ | figure that one out |
02:18.02 | Godsey | is there a way to set the timeout in sip.conf for register => ? |
02:18.09 | Godsey | or refresh time? |
02:18.43 | *** join/#asterisk blitzrage (~blitzrage@d141-239-17.home.cgocable.net) |
02:18.49 | bkw_ | 996 |
02:18.57 | bkw_ | blitzrage whats up eh? |
02:19.04 | Godsey | sup and shit? |
02:19.14 | blitzrage | bkw_: lol |
02:19.16 | blitzrage | eh |
02:20.42 | blitzrage | s/recompiled/recompiles |
02:27.58 | *** join/#asterisk dfusr77 (~chatzilla@66.239.43.90.ptr.us.xo.net) |
02:32.43 | dfusr77 | anyone awake out there? |
02:33.59 | *** join/#asterisk ryguillian (~ryguillia@c-24-12-96-52.client.comcast.net) |
02:34.41 | dfusr77 | heh... I'm looking for some help on tracking down an oops on a server... |
02:35.02 | dfusr77 | I have core files that I've reviewed... and I need some help with them.. |
02:35.26 | robert_wwl | "an oops" I like the way you phrased that |
02:35.27 | dfusr77 | anyone up for a challenge? ;) |
02:35.39 | robert_wwl | what kind of core files? |
02:35.42 | dfusr77 | heh... |
02:35.51 | dfusr77 | my * has been crashing of late.. |
02:36.02 | dfusr77 | ever since I upgraded to 1.0 |
02:36.07 | robert_wwl | Hmmm... |
02:36.12 | dfusr77 | I believe it has to do with parking.. |
02:36.15 | dfusr77 | ..features. |
02:36.46 | robert_wwl | Sadly, my asterisk expertise is limited to calling Queues |
02:36.48 | dfusr77 | I noticed in the core that when someone picks up a parked call, it gets an out of bounds err |
02:36.56 | robert_wwl | hmm... |
02:37.01 | robert_wwl | that's interesting |
02:37.08 | dfusr77 | k... |
02:37.18 | robert_wwl | have you check to see if there's been a bug posted about it? |
02:37.25 | dfusr77 | I'm not looking forward to rolling back... |
02:37.27 | *** join/#asterisk sudoer (~sudoer@65.75.148.190) |
02:37.32 | dfusr77 | I don't see anything in the bugtracker |
02:37.46 | robert_wwl | You should put something there |
02:37.51 | dfusr77 | I wanted to see if anyone could help via IRC before I put it up there. |
02:37.57 | robert_wwl | Smart move |
02:38.38 | dfusr77 | it's funny, the park function works fine when there's no one on the system. |
02:38.56 | *** join/#asterisk easydone (~notdone@eksel.demon.nl) |
02:38.57 | dfusr77 | but after a certain time of activity, when accessed it dumps... ARGH! |
02:39.32 | robert_wwl | what kind of error does it give when it dumps, or does it? |
02:39.41 | dfusr77 | it doesn't |
02:39.51 | dfusr77 | just seg faults.. |
02:40.02 | robert_wwl | ah, that's what I was looking for |
02:40.14 | dfusr77 | #1 0x0807728f in pbx_extension_helper (c=0x9ce7f20, |
02:40.26 | dfusr77 | <PROTECTED> |
02:40.34 | dfusr77 | that's some of the gdb info.. |
02:40.55 | dfusr77 | 701 is the first parking slot. |
02:41.26 | robert_wwl | I can't really tell you anything that's not obvious. |
02:41.36 | robert_wwl | Do you know C? |
02:41.43 | dfusr77 | barely... |
02:41.55 | robert_wwl | hmmm... |
02:41.58 | dfusr77 | I can cut & paste with the best of them ?? |
02:42.00 | dfusr77 | heheh |
02:42.07 | robert_wwl | I'll download the new source and take a look |
02:42.27 | dfusr77 | thanks... it's current as of 26th. |
02:44.45 | dfusr77 | in an another core dump, after the I find this... |
02:45.12 | dfusr77 | #0 0x40027d92 in pthread_mutex_lock () from /lib/libpthread.so.0 |
02:45.14 | dfusr77 | #1 0x4018682d in free () from /lib/libc.so.6 |
02:45.15 | dfusr77 | #2 0x0807707b in pbx_extension_helper (c=0x0, context=0x8eabbd0 "tack\n", |
02:45.17 | dfusr77 | <PROTECTED> |
02:45.18 | dfusr77 | <PROTECTED> |
02:45.20 | dfusr77 | #3 0x0807dd00 in ast_pbx_run (c=0x40677df0) at pbx.c:1879 |
02:45.22 | dfusr77 | #4 0x4051b1d8 in ss_thread (data=0x40677df0) at chan_zap.c:4865 |
02:45.23 | dfusr77 | #5 0x4002783e in pthread_start_thread () from /lib/libpthread.so.0 |
02:45.25 | dfusr77 | #6 0x401dd04a in clone () from /lib/libc.so.6 |
02:45.30 | bkw_ | dfusr77 STOP IT STOP IT STOP IT |
02:45.35 | bkw_ | its called pastebin |
02:45.48 | robert_wwl | yeah, dude you should really pastebin that to me |
02:45.48 | dfusr77 | too much, my bad. |
02:45.49 | bkw_ | ok |
02:45.50 | bkw_ | do this |
02:45.53 | bkw_ | in gdb |
02:45.53 | file2 | + |
02:46.01 | file2 | - |
02:46.04 | file2 | there, I evened it out. |
02:46.23 | bkw_ | frame 2 |
02:46.28 | bkw_ | print context |
02:46.35 | bkw_ | well |
02:46.36 | sudoer | hi all |
02:46.37 | bkw_ | just do bt full |
02:46.40 | bkw_ | and put it on pastebin |
02:47.01 | dfusr77 | sorry guys... pastebin.. what is that? |
02:47.09 | robert_wwl | pastebin.ca |
02:47.33 | dfusr77 | got it.. |
02:48.13 | drumkilla | ~pastebin |
02:48.15 | jbot | [pastebin] a place to paste all your conf/debugs/logs for other people in the chatroom to view without flooding the channel. We suggest http://pastebin.ca |
02:48.37 | sudoer | are there any actual voip providers that aere fairly large and jsut resell other companies like nufones minutes? |
02:48.47 | blitzrage | Anytime anything tries to register with SIP, I get an __sip_xmit error Bad File Descriptor. Same configs as RC2 no longer work on my 1.0.0? |
02:48.54 | robert_wwl | ~voip providers |
02:49.04 | dfusr77 | check it out at: http://pastebin.ca/1142 |
02:49.08 | robert_wwl | sudoer: nope |
02:49.35 | robert_wwl | dfuser77: cool |
02:50.08 | dfusr77 | sorry about cut & paste before... got trigger happy. |
02:50.13 | blitzrage | I looked at the code, but I still don't understand what causes that error... |
02:50.22 | sudoer | so what is the best way to be a voip provider? |
02:50.39 | robert_wwl | I'll take a look, and maybe talk about it with my supervisor tomorrow, and if I fix anything I'll submit a patch |
02:51.17 | dfusr77 | blitz-- it seems to be from parking .. not duplicable during off-hours testing (ofcourse) but during the day with activity. |
02:51.26 | sudoer | and why do companies like voicepulse anf nufone offer reseller services? |
02:51.29 | dfusr77 | thanks robert... I assume you work @ digium? |
02:51.42 | Tond | How do I tell my extention to dial an extention? I am trying to use forking to call 3 different extentions, and I would like to avoid using SIP/ , IAX/ , ZAP/ is there a way I can tell * to dial an extention nmber since I have already told that extention where the phone is. |
02:52.26 | Tond | "How do I tell my extention to dial an extention?" => How do I tell * to dial an extention? |
02:52.37 | robert_wwl | dfuser77: Actually, I work at vcch.com |
02:53.02 | robert_wwl | We're big on asterisk |
02:53.32 | dfusr77 | cool! |
02:53.41 | Tond | Robert> do you guys setup * for your clients? |
02:53.51 | Tond | oops |
02:53.55 | *** join/#asterisk ASPWorld (~info@209.91.159.221) |
02:53.56 | dfusr77 | just checked out yer site, do you guys deal with international setups? |
02:53.58 | Tond | i just got the answr to that |
02:54.00 | Tond | lol |
02:54.07 | robert_wwl | Tond: Yeah |
02:54.13 | robert_wwl | dfusr77: Yeah |
02:54.17 | Tond | Robert> How do you guys deal with Intercom? |
02:54.30 | robert_wwl | Tond: Who? |
02:54.39 | Tond | Cisco's new IOS supports that feature.. |
02:54.48 | dfusr77 | any sat engineers on board? |
02:55.08 | Tond | Robert> Intercom |
02:55.18 | robert_wwl | Tond: I wouldn't know. My supervisor handles most of the asterisk installations. |
02:55.25 | robert_wwl | I'm just an intern |
02:55.27 | Tond | Robert> like when u dial an extention and you get to page people |
02:55.29 | Tond | Oh Ok.. |
02:55.53 | Tond | dfusr77> ask ur question, if anyone knows they'll answer i guess.. |
02:55.53 | Tond | :) |
02:56.28 | *** join/#asterisk telme (~mine@c-24-8-57-124.client.comcast.net) |
02:56.55 | dfusr77 | robert... just curious yer site shows sat pics, if that's just marketing, never mind, but I'm interested in speaking to some satellite engineers for an upcoming project.. |
02:57.40 | *** part/#asterisk jim340 (jpc@207.18.139.4) |
02:57.43 | robert_wwl | No, those are just marketing. Our salesman did our website and got those pics somewhere |
02:57.50 | dfusr77 | k... |
02:57.51 | telme | so i want to play around with g729 and i'm going through the instructions at readytechnology (via the wiki) but can't seem to find the spot on the intel site where i register for a key. any else done this? |
02:58.05 | Tond | we setup a Satellite internet for one of our locations, but that is about all I know i guess, and very little more.. |
02:58.05 | Tond | lol |
02:58.17 | dfusr77 | ;) |
02:58.55 | *** join/#asterisk RolloTomnasi (~JH@delmar-209-137-161-171-dsl.cavtel.net) |
02:59.04 | Tond | dfusr> are you planing to pass voice through Sat? Or VOIP over Sat connection? |
02:59.13 | dfusr77 | my project has a req for app testing via sat links, and having some sat engineers working along will be nice.. |
02:59.42 | dfusr77 | no voice, atleast not at the beginning.. just an app (java) and then later multi-media (video streaming) |
03:00.07 | dfusr77 | from some remote areas.... |
03:00.15 | dfusr77 | therein lies the challenges. |
03:00.28 | dfusr77 | hence the req for sat eng... |
03:01.11 | Tond | well What I know is that Sat adds around 400 - 450 ms of delay, and voice traffic can go through with great quality up to max 680 to 700 ms of delay |
03:01.31 | Tond | but i guess that info is useles to u you now |
03:01.34 | Tond | :) |
03:01.43 | dfusr77 | round trip... not one way.. |
03:01.50 | Tond | Yes, round trip |
03:01.55 | dfusr77 | I'm familiar with sat ... |
03:02.11 | dfusr77 | for industrial location connections... |
03:02.21 | Tond | well many engineers told me that 680 - 700 ms is too much for voice and I won't be able to do it. but I did it and workled perfectly |
03:02.29 | RolloTomnasi | Question about MOH over X100P.. I have 0 echo on the line, but when I put callers on hold moh is a bit scratchy, it fades in and out. Does anyone have a suggestion as to how I can stream it smoothly? |
03:02.30 | dfusr77 | not for rural areas, connecting via low power devices.. |
03:02.34 | Tond | so sometimes u have to try things |
03:02.40 | *** join/#asterisk voipjet (~beorn@ottawa-hs-64-26-155-97.s-ip.magma.ca) |
03:02.48 | dfusr77 | heh... |
03:02.52 | pfn | yeah, latency on voice isn't as big a deal as people make it out to be |
03:02.59 | dfusr77 | it'll be an interesting project.. |
03:03.00 | Tond | Rollo> I am not sure, but maybe has to do something with VAD? |
03:03.28 | RolloTomnasi | Tond: VAD? Ok thanks.. I'll look into that (dunno what it is?). |
03:03.32 | Tond | ya, I even had calls going through at 800 ms |
03:03.33 | dfusr77 | rollo- check your mpg123 is version 0.59r not 0.59s |
03:03.40 | Tond | but then the voice quality started to suffer |
03:03.52 | Tond | 680 - 700 ms u wont be able to tell the difference |
03:03.52 | RolloTomnasi | dfusr77: yup .059r |
03:04.16 | Tond | Rollo> VAD > Voice Activity Detection |
03:04.37 | dfusr77 | rollo - check the mp3 file to ensure it's not variable rate files also.. |
03:04.53 | dfusr77 | tond- what equip you using? * or some hw ? |
03:05.07 | Tond | ur MOH song might be kinda low on volume or slow or something and VAD is trying to save bandwidth by cutting the stream of rtp. But it is only a guess SDonm't even knwo if VAD will interfear with MOH |
03:05.31 | Tond | dfusr> equipment for what? |
03:05.40 | dfusr77 | on the sat link.... |
03:05.58 | Tond | I can't remember, We bought all of them together.. |
03:06.08 | dfusr77 | o... |
03:06.11 | Tond | we have a Meter Dish as well as smaller ones.. |
03:06.15 | *** join/#asterisk bdeb4 (bdeb4@alb-24-195-238-207.nycap.rr.com) |
03:06.16 | Tond | the 5 meter one has tracking |
03:06.27 | bdeb4 | hi, can anyone recommend some cheap fxs adapters? |
03:06.38 | dfusr77 | O... you got the real deal.. ;) |
03:06.45 | Tond | Yep.. |
03:06.46 | Tond | lol |
03:06.59 | Tond | costed us around 250K EUs |
03:07.07 | dfusr77 | here, I'm thinkin you got a link to provider.. w/ a sat link.. :-) |
03:07.19 | Tond | Nooo.. :) |
03:07.20 | dfusr77 | ouch.. |
03:07.29 | Tond | we pass 300 Mbits of Data through that |
03:07.43 | *** join/#asterisk Ferrari355 (~dcox@12-221-104-25.client.insightBB.com) |
03:07.45 | dfusr77 | *drooling* |
03:07.51 | Tond | lool |
03:07.53 | RolloTomnasi | dfusr77 - i stripped out all extra info & everything is <= 128 , not sure how to check if variable bit rate? it happens w/ the stock MOH mp3's too though |
03:08.22 | Tond | stock > meaning the ones that came with * install? |
03:08.28 | pfn | mpg123 0.59r? |
03:08.29 | RolloTomnasi | Tond, yup |
03:08.35 | RolloTomnasi | pfn, yup |
03:08.51 | dfusr77 | there was a change recently, stripping the id3 tags... try upping to latest cvs |
03:08.56 | Tond | Hrm... And this happen on local LAN right? |
03:09.03 | pfn | shouldn't be choppy then |
03:09.16 | RolloTomnasi | not really choppy, i'd almost say scratchy |
03:09.26 | RolloTomnasi | tond - no, only when calling out on pstn |
03:09.31 | RolloTomnasi | inbound sip calls are 100% |
03:09.34 | Tond | Rollo> do u have a TDM card in the box? |
03:09.40 | RolloTomnasi | x100p |
03:09.44 | Tond | Oh.. |
03:09.46 | dfusr77 | it was nice chattin with you all... I gotta bolt... robert- thanks for your help! g'night |
03:09.55 | RolloTomnasi | thx for the help dfusr77 |
03:10.02 | Tond | dfusr> lates |
03:10.11 | dfusr77 | or morning depending on where you're at. ;) |
03:10.13 | dfusr77 | l8r |
03:10.17 | RolloTomnasi | general q.. if I update from cvs, 1.0 is default now? |
03:10.27 | Tond | Rollo> when you call out is your voice liek that too? |
03:10.36 | Tond | Rollo> almost like statics on the line, right? |
03:10.37 | RolloTomnasi | tond, nope, voice is 5x5 |
03:10.45 | RolloTomnasi | yes |
03:11.03 | Tond | well then voice must be the same.. I think I heard abotu this in the Conference.. |
03:11.12 | Tond | try changing the PCI slot of the card |
03:11.23 | Tond | and make sure the card has it's own intrupts |
03:11.28 | Tond | interupt |
03:11.31 | Tond | IRQ |
03:11.32 | RolloTomnasi | voice had a bit of echo at first, changed rx&tx (also couldn't hear MOH at all before changed gain and turned on training) |
03:11.49 | RolloTomnasi | changed that earlier ;) cat /proc/interrupts is my friend |
03:11.58 | Tond | :) |
03:12.10 | Tond | change the PCI slot # |
03:12.13 | Tond | physically |
03:12.15 | RolloTomnasi | that should be in bold at the top of the list... |
03:12.24 | Tond | I heard that when some ppl did that it wrked |
03:12.33 | RolloTomnasi | for this issue? |
03:12.41 | Tond | Ya.. for having Static on their line.. |
03:12.55 | RolloTomnasi | ok cool.. thanks :) i'm dreading it (btdt 2x today) |
03:12.57 | Tond | like static noise |
03:13.00 | RolloTomnasi | yeah |
03:13.07 | RolloTomnasi | that's exactly what it is - but only MoH |
03:13.11 | Tond | well try it.. not sure why the hardware might do that |
03:13.29 | *** join/#asterisk michael12345 (~mick@196.40.69.228) |
03:13.36 | Tond | some guy said he uused a static free bags that come with hardware and put it between his cards and the static problem was solved |
03:13.53 | RolloTomnasi | that's interesting |
03:14.00 | michael12345 | can asterisk do call intruding for helpdesk training |
03:14.19 | pfn | zapbarge |
03:14.25 | RolloTomnasi | tond - thanks for the help, thanks pfn |
03:14.28 | pfn | intruding? |
03:14.38 | pfn | listening or interacting |
03:14.40 | Tond | Rollo> sure.. |
03:14.41 | Tond | :) |
03:15.19 | Tond | my damn Ethernet card keeps on sharing the IRQ with my Zap card! lol |
03:15.25 | michael12345 | pfn: both would be great |
03:15.28 | Tond | but so far I haven't had any problem (Knock on wood) |
03:15.33 | RolloTomnasi | really? |
03:15.35 | michael12345 | but just listening for the time being |
03:15.46 | RolloTomnasi | tond - i couldn't dial out until i cleared the irq sharing |
03:15.46 | Tond | ya |
03:15.49 | *** join/#asterisk abombss (~abombss@c-67-163-3-0.client.comcast.net) |
03:16.04 | Tond | Mine is working fine.. Maybe it is my mothernoard |
03:16.21 | Tond | Liek the MBV is handeling that |
03:16.25 | pfn | zapbarge for zap channels |
03:17.16 | RolloTomnasi | yeah, my problem was that only slot 1 shares resources in BIOS, but mobo jumps on to share different interrupts... so shuffle..shuffle.. bang case..shuffle :( |
03:17.22 | Tond | Can I tell * to Dial an extention number instead of using SIP/.... ? |
03:17.48 | Tond | Hrm.. |
03:17.55 | michael12345 | pfn: is that a no |
03:18.25 | *** part/#asterisk telme (~mine@c-24-8-57-124.client.comcast.net) |
03:18.33 | Tond | Rollo> what do you use * for? |
03:18.51 | RolloTomnasi | i write ext=SIP/#### then just dial(${Ext}/${EXTEN}) |
03:19.26 | RolloTomnasi | tond - well for business, three diff partners in various locations |
03:19.31 | RolloTomnasi | I'm a complete newbie |
03:19.39 | Tond | Rollo> me 2 |
03:19.40 | Tond | :) |
03:19.45 | RolloTomnasi | heh |
03:19.56 | RolloTomnasi | US? |
03:20.16 | Tond | Canada |
03:20.38 | RolloTomnasi | so pretty much the same carriers? |
03:21.15 | RolloTomnasi | I did a lot of research trying to figure out who to use, etc etc etc. that was more frustrating than setting up * |
03:22.21 | RolloTomnasi | in the end, I had a vonage line here for voice/fax, and a couple extra virtual #'s.. so i just decided to use vonage for primary then roll over to gafachi outbound, using vonage softphone for inbound. other 2 offices are using (sp?) voicepulse connect |
03:23.06 | Tond | well we have our own Carrier access too.. Cause we are int he business of wholesale minutes |
03:23.11 | Tond | so our carriers are nicer to us |
03:23.11 | Tond | :) |
03:23.22 | RolloTomnasi | ahh |
03:25.47 | Tond | What do you guys think about using Python? |
03:26.06 | Tond | I am debating between using Perl and Python to do some developments for * |
03:26.30 | *** join/#asterisk edguy3 (~edguy@host-24-225-213-218.patmedia.net) |
03:26.35 | afrosheen | perl is good, so is python |
03:26.37 | pfn | ~seen stealth_man |
03:26.38 | jbot | stealth_man <stvpn@ool-18bc2736.dyn.optonline.net> was last seen on IRC in channel #asterisk, 1d 3h 10m 8s ago, saying: 'OEJ: I will see you shortly in Europe or let's talk after you are back to homeland'. |
03:26.43 | afrosheen | perl is probably easier to share though |
03:26.54 | sudoer | are there any actual voip providers that aere fairly large and jsut resell other companies like nufone's minutes? |
03:27.05 | *** join/#asterisk MustDie (~voip@ool-18b91fa7.dyn.optonline.net) |
03:27.06 | Tond | But Python is easier to work with and it is Object Oriented |
03:28.23 | Tond | plus I heard that the Networking protocol was written using Python |
03:28.30 | Tond | like TCP |
03:29.01 | afrosheen | sounds like you have already decided |
03:29.22 | Tond | afrosheen> well not really, but so far Python is wining |
03:29.26 | Tond | :) |
03:29.30 | sudoer | has anyone had a problem where they make a call from voip phone, but only the first second of outgoing audio is sent, then nothing else? |
03:29.59 | Tond | Don't get me wrong I think Perl is a great language, but Python's Object Orientation is very handy |
03:30.20 | sudoer | I am using several providers and the only one that doesnt work proerly is voipjet.com, the voice stops afger one second, but i want to use voipjet as they seem like the best company |
03:30.34 | Tond | sudoer> strange.. |
03:30.37 | afrosheen | sounds like they suck to me |
03:30.49 | Tond | do some packet tracing and sip debugs (if that is what you use) and see what happens |
03:30.49 | afrosheen | voip-nyet |
03:31.15 | sudoer | i have the sip debug on, but i dont know what to make of the output becuase its spitting out hunderds of lines |
03:32.00 | Tond | gotta go though them one by one |
03:32.01 | Tond | :) |
03:32.39 | afrosheen | brb |
03:34.44 | RolloTomnasi | sudoer, it shouldn't be too hard to find the appropriate messages, search by the line you're calling... |
03:34.47 | *** join/#asterisk Connor_ (~billy@198-144-165-65.knx.tn.nxs.net) |
03:35.11 | sudoer | what does this mean? answering with prefferred capibility 0x4(ULAW) then the next line: answering with non-codec capibility 0x1(g723) |
03:35.54 | sudoer | in sip config file for this phonei have disallow=all;allow=law; does that mean its useing g723 when it shouldnt be? |
03:36.05 | sudoer | allow=ulaw i mean |
03:37.16 | *** join/#asterisk PTG123 (~root@ip68-106-19-249.ph.ph.cox.net) |
03:37.57 | *** part/#asterisk PTG123 (~root@ip68-106-19-249.ph.ph.cox.net) |
03:38.01 | RolloTomnasi | sudoer, you're beyond my knowledge, but ulaw is g.711, g723.1 requires a license |
03:38.07 | RolloTomnasi | http://www.voip-info.org/wiki-Asterisk+codecs |
03:38.13 | pfn | no, it's not answering with anything |
03:38.20 | pfn | no idea what that non-codec capability shit is |
03:38.25 | Tond | sudoer> does ur phone support g723? |
03:38.30 | pfn | look at the SDP |
03:38.33 | pfn | not the sip message |
03:38.47 | sudoer | i dont thikn so its just a grandstream budgettone, what is sdp? |
03:40.07 | Tond | could be that ur provider is changin codec after the call is establioshed? |
03:40.37 | sudoer | i dont think so, their site says the stay with ulaw if possible |
03:40.48 | sudoer | also here has a similar message: http://mharc.lists.openservices.ca/archives/html/asterisk-users/2003-08/msg01540.html |
03:40.55 | Tond | one of the 2 ends might want to use g723 and the other side doesn't support it so * will ahve to convert between them , and * can't do that with g723. (JUST THEORY) :) |
03:42.07 | RolloTomnasi | sudoer, you've been trying to figure this out for a while huh? I couldn't remember why this sounded familiar but it was here on sat hehe |
03:42.14 | Tond | manually set ur codecs to Ulaw and see if it helps |
03:42.19 | bkw_ | ok linux is not for the desktop at all |
03:42.23 | bkw_ | too much crap to do |
03:42.26 | bkw_ | it should "just work" |
03:43.00 | sudoer | Tond, i have disallow=all;allow=ulaw; in many places |
03:43.17 | Tond | bkw_> I ain't stuck on the idea of only using Linux. I use all OS.. Windows, Unix, ofcoarse windoes is only good for desktop and nothing else.. |
03:43.18 | Tond | lol |
03:43.29 | bkw_ | zactly |
03:43.33 | Tond | sudoer> how about in ur phone? |
03:43.34 | sudoer | RolloTomnasi, i fixed most of the problem,s before all providers had no outgoing sound, now only one doesnt have outgoing sound |
03:44.33 | sudoer | setting now, rebooting ... |
03:45.12 | sudoer | no, didnt work |
03:45.44 | sudoer | pfn what is the SDP stuf i should look at? |
03:46.39 | cypromis | hmm |
03:46.43 | cypromis | linux is good for desktop |
03:46.50 | cypromis | if you need anything else you just start vmware |
03:46.54 | cypromis | and run anything else |
03:47.10 | Tond | sudoer> what are the errors again? |
03:47.13 | RolloTomnasi | sudoer, do you have that sip error? |
03:47.15 | RolloTomnasi | ha |
03:47.18 | pfn | sudoers nevermind |
03:47.18 | Tond | and this only happens with one provider? |
03:47.23 | pfn | sudoers the g723 is irrelevant, ignore it |
03:47.28 | RolloTomnasi | it looks like it's just saying that it's not compatible |
03:49.01 | sudoer | Tond, the error i am getting is that i only get 1 error of sound sent out from my voip phone, but this only happens on one provider, voipjet |
03:49.21 | sudoer | which sip error? |
03:49.43 | *** join/#asterisk Odie_flocon (~Odie_floc@S010600095b4c7aaa.lb.shawcable.net) |
03:50.10 | Tond | sudoer> so u said u hear somethign for 1 sec and then it drops right? |
03:50.17 | sudoer | yes |
03:50.21 | Tond | sudoer> What is it u hear? ring? |
03:50.39 | RolloTomnasi | sudoer, n/m what i asked - the error you stated earlier - non-codec capability - all it's saying is , compatible w/ abc, not compatible w/ z.. like pfn said, ignore the g723 line, it's not an error |
03:51.19 | sudoer | no, i am testing with voip phone and cell, i call my cell through one voip provider, and answer cell, then i start talking into voip phone and i hear on cell for 1 seconds then it get cut off |
03:51.26 | Tond | Sudoer> I think * is trying to convert between 2 different codecs and it can't cause one side is using g723 |
03:51.28 | sudoer | ok RolloTomnasi |
03:51.35 | Tond | Oh |
03:51.41 | Tond | Ok nm my answer too then |
03:51.58 | sudoer | I dont think there is g723 anywhere in the path.... |
03:53.12 | Tond | I don't knwo why the voice would get droped unless the prtovider is changing the codec suring the call |
03:53.16 | Tond | which is posible |
03:53.36 | Tond | It would ahve been very helpful to see the problem occurng in front of me |
03:53.48 | Tond | but it could also be missconf on ur provider side |
03:53.50 | sudoer | you mean here? |
03:53.59 | sudoer | or see sip debug? |
03:54.00 | sudoer | or...? |
03:54.04 | sudoer | hear |
03:54.40 | Tond | here |
03:54.41 | Tond | :) |
03:54.46 | Tond | like be at the comp |
03:54.56 | Tond | to be able to change config, make calls look at the logs.. |
03:54.56 | Tond | :) |
03:55.36 | m-00kie | anyone have a recommeded Sipura-like analog converter? |
03:55.47 | m-00kie | apparently the Sipuras generate too much noise |
03:55.58 | Tond | I wouldn't know |
03:56.02 | m-00kie | i need something thats as small and low-profile as possible |
03:56.09 | Tond | I yet haver to test my ATA-188 with * |
03:56.10 | Odie_flocon | hey sudoer was it you I was talking to about a cheap sip phone? |
03:56.10 | Tond | :) |
03:56.35 | sudoer | It cvould have been me |
03:56.55 | sudoer | Tond, if i test with a iax phone, will that tell me anything? |
03:57.03 | Odie_flocon | I thought that there was a cheap sip phone on digiums website. |
03:57.11 | Odie_flocon | I figured out where it was. |
03:57.23 | Odie_flocon | it's like 65.00 |
03:57.31 | Tond | try it and see if the call goes through |
03:57.31 | NeroLabs | budgetone |
03:57.53 | Odie_flocon | and it's on pulver.com |
03:58.04 | Tond | Odie> have ya used it? |
03:58.05 | sudoer | which phone, i have budgettone |
03:58.54 | Odie_flocon | no, I'm planning on getting one. |
03:59.06 | Odie_flocon | I'm sure it's the bugettone. |
04:00.46 | NeroLabs | sudoer: you like budgetone? I just ordered one (BT-101) to test it out |
04:01.56 | JerJer | the newer ones are better |
04:02.03 | JerJer | but they are still toys to me |
04:02.54 | Tond | I like the new IOS on Cisco 79xx |
04:03.15 | Tond | it allows for call forwarding feature on screen as a button and also Intercom option, etc.. |
04:03.17 | bkw_ | whats the latest? |
04:03.22 | Tond | 7.2 |
04:03.23 | RolloTomnasi | which is that? |
04:03.33 | RolloTomnasi | CFwdALL? |
04:03.33 | Tond | Cisco 79xx phones |
04:03.38 | Tond | ya! |
04:03.40 | RolloTomnasi | hahah |
04:03.47 | sudoer | i dont like it that much |
04:04.26 | RolloTomnasi | yeah, nice. should have a pingtel unit tomorrow.. but the 7960's & 70's are sooo nice |
04:04.46 | Odie_flocon | hmmm. pingtel. not too bad. |
04:04.50 | Odie_flocon | I like snom. |
04:05.05 | Odie_flocon | the snom 200 is a nice set. |
04:05.27 | Tond | Rollo> I ahve 2 pingtels, thwey are shit |
04:05.35 | Tond | they shitties phones are pingtel! |
04:05.48 | Tond | I ahve snom too.. |
04:05.49 | Tond | they are OK |
04:05.55 | Tond | nothing beats Cisco! |
04:05.56 | Tond | :) |
04:06.09 | Odie_flocon | me no like cisco. |
04:06.11 | RolloTomnasi | just price.. |
04:06.33 | Tond | well ya |
04:06.42 | Tond | But Snom ain't as reliable |
04:06.47 | Tond | Pingtel, don't even talk about it |
04:07.01 | Tond | Snom behaves wierd from time to time and also crashes too |
04:07.11 | Odie_flocon | yeah |
04:07.16 | Odie_flocon | snom like to crash. |
04:07.20 | WilliamK | Tond, which firmware release? |
04:07.43 | Tond | Will> for Cisco? 7.2 |
04:07.54 | Tond | or Snom? |
04:07.54 | WilliamK | no for the Snom |
04:08.01 | Tond | i can checkl |
04:08.17 | Tond | but the latest, cause it checks the web after every reboot and tells me if there is anyhting to update |
04:08.32 | WilliamK | 2.04g? |
04:08.38 | Vco | so, i have to compile the linux 2.6 kernel source to build zaptel? |
04:09.23 | Tond | Will> trying to find through the web interface |
04:09.33 | Vco | or am i just dense? |
04:09.51 | Odie_flocon | no you don't have to compile the 2.6 kernel. |
04:10.25 | Odie_flocon | you just need to have the sources available to it. |
04:10.43 | *** join/#asterisk sudoer (~sudoer@65.75.148.190) |
04:10.46 | sudoer | got disconnected |
04:11.01 | Vco | when try to compile zaptel (and have sources linked to /usr/src/linux-2.6) it craps itself |
04:11.15 | sudoer | Tond, is it possible for you to help me look at my sip debug, i cant figure out what my problem is |
04:11.54 | Tond | Sudoer> I ain't no expert, but I can login and have a look if u want me to |
04:11.59 | Odie_flocon | what's wrong with your sip sudoer? |
04:12.22 | sudoer | I only get one second of outgoing calls with one of the providers im trying to use |
04:12.35 | *** join/#asterisk gafachi (~gafachi@gm1.gafachi.com) |
04:12.39 | WilliamK | Tond, also which ver of the bootloader are you using on the phone |
04:12.59 | Tond | Will> let me physically go check |
04:13.09 | Odie_flocon | one second. |
04:13.15 | Odie_flocon | it connects then what? |
04:13.54 | Odie_flocon | can you show me the debug sudoer? |
04:14.07 | sudoer | then no outgoing audio from voip to cell phone, but i can talk to cell and hear on voip side |
04:14.17 | sudoer | ok, here, http://pastebin.ca/1143 |
04:15.09 | Odie_flocon | ok gimme a minute. |
04:15.12 | *** join/#asterisk marco-at-home (~m@user-0cet3gr.cable.mindspring.com) |
04:15.25 | *** join/#asterisk B0ngFrOg (~wsmith@67.176.78.68) |
04:15.48 | B0ngFrOg | whooohooo |
04:15.51 | marco-at-home | hello all. Can someone point me in the right direction when deciding on which codec to use? |
04:17.43 | sudoer | that debug starts from dialing out from voip phone to answergin with cell, no sound, nd the nhanging up |
04:18.13 | sudoer | marco-at-home: read this article: http://www.networkcomputing.com/1202/1202ws32.html |
04:18.27 | Tond | ok |
04:18.29 | Tond | let me read it suo |
04:18.33 | Tond | sud |
04:18.58 | Odie_flocon | your using Ulaw.. |
04:18.59 | marco-at-home | sud: will do.. |
04:19.12 | sudoer | Odie_flocon, isnt ulaw good? |
04:19.25 | Odie_flocon | nooo |
04:19.27 | Odie_flocon | very bad |
04:19.36 | Odie_flocon | it's full 64kBandwidth |
04:19.42 | Odie_flocon | very bad |
04:19.45 | sudoer | yes ,but its sounds best |
04:19.47 | Odie_flocon | use 7.29 |
04:19.54 | Odie_flocon | or 7.23 |
04:20.09 | Odie_flocon | sounds best but uses too much bandwidth. |
04:20.25 | Tond | Odie> if u got rhe bandwidth y not? |
04:20.37 | Tond | G729 or G723 requires more money |
04:20.38 | sudoer | what is 7.29 and 7.23 in * names? |
04:20.38 | Tond | :) |
04:20.56 | Odie_flocon | umm you'll want g723 |
04:20.57 | marco-at-home | approx how much is 729 per seat? |
04:20.57 | Tond | G729a and G723 |
04:21.05 | Odie_flocon | you need to pay for g.729 |
04:21.05 | Tond | $10 |
04:21.15 | Tond | And G723 is very expensive |
04:21.19 | Odie_flocon | free |
04:21.43 | Odie_flocon | set your codec to default to g723 |
04:21.49 | JerJer | G.723.1 is a joke |
04:21.50 | brc_ | baaaaack |
04:21.53 | sudoer | is ther any problems you see in my debug? |
04:22.00 | JerJer | use G.729 or drop it into Asterisk and use GSM or iLBC |
04:22.13 | Odie_flocon | yes GSM is good too. |
04:22.21 | Tond | I liek GSM |
04:22.26 | Tond | but not many phoines support it |
04:22.30 | JerJer | so? |
04:22.32 | JerJer | Asterisk does |
04:22.37 | brc_ | marco-at-home, which codec you use TOTALLY depends on what you are doing |
04:22.41 | marco-at-home | what about speex? |
04:22.44 | brc_ | the farfon supports gsm |
04:22.58 | brc_ | marco-at-home, are you going to be using this for phones on a LAN? |
04:23.12 | brc_ | or over a internet connection |
04:23.32 | marco-at-home | brc: intra-office, i will use GSM as we're on a 100 megabit lan. However, i'm debating what codec to use between us and Nufone... |
04:23.35 | brc_ | if you're trying to figure out what codec to use on a LAN between phones and asterisk use ulaw |
04:23.35 | Tond | JerJer> ya but then * has to convert |
04:23.36 | Odie_flocon | he's using a provider to connect to his cell phone. |
04:23.41 | brc_ | marco-at-home, between you and nufone use GSM |
04:23.55 | Tond | Ya on a lan I would use Ula |
04:23.58 | brc_ | marco-at-home, if you have a really highspeed connection you can use ULAW between you and nufone |
04:24.07 | brc_ | but I'd suggest trying GSM from you to nufone first |
04:24.19 | JerJer | Tond: and the problem is? |
04:24.24 | marco-at-home | brc: i have 10 megabit connection. However, i'd hate to waste bandwidth on voice... |
04:24.30 | brc_ | marco-at-home, how many phones? |
04:24.34 | marco-at-home | 4 |
04:24.42 | marco-at-home | err...5 |
04:24.42 | sudoer | is iax debug output useful? my voip phones goes to my server via sip, then conencts to another provider via iax |
04:24.57 | brc_ | k, phone ===ulaw===> asterisk ===GSM===> nufone |
04:24.59 | Tond | JerJer> there si no problem with that... :) |
04:25.02 | Odie_flocon | the reason it doesn't do voice 2 ways is because your using 2 different codec |
04:25.09 | sudoer | me? |
04:25.30 | sudoer | dose it show im using 2 different codecs? i thought * can translate? |
04:25.42 | marco-at-home | brc: actually, i can hear some frames being dropped between me and nufone. (i assume the 'stuttering' = dropped frames). So, my theory is w/ lower bandwidth, less chance of dopped frames.. |
04:26.00 | brc_ | ~ping |
04:26.01 | jbot | pong |
04:26.09 | Odie_flocon | I don't know if * can translate? |
04:26.14 | sudoer | ~help |
04:26.18 | brc_ | Odie_flocon, translate what? |
04:26.29 | brc_ | yes asterisk can do codec translation of course |
04:26.31 | Odie_flocon | from gsm to g729 |
04:26.34 | Tond | Sudo? can u copy everything from ur * consol, form the stat fo the call toend? |
04:26.35 | brc_ | see above |
04:26.42 | Tond | and set verbos to like 10 |
04:26.44 | JerJer | if you have the g729 vocoder, sure it can |
04:26.44 | Tond | or something |
04:26.45 | Odie_flocon | ok |
04:26.47 | sudoer | ok Tond |
04:26.52 | brc_ | JerJer, good point |
04:26.56 | sudoer | iax debug too? |
04:27.01 | brc_ | marco-at-home, pm |
04:27.12 | Tond | Sud> only if u r using IAX too |
04:27.15 | brc_ | what happened to jbot? |
04:27.25 | Odie_flocon | sorry, I've never gotten into * enough to find out. |
04:27.59 | Tond | I am doing transcoding on * and it does a great job |
04:28.03 | Tond | even with G729 |
04:28.09 | Odie_flocon | I bet it would. |
04:28.15 | Odie_flocon | I've never done it. |
04:28.21 | sudoer | i just looked at iax2 debug, there is nothing useful info in htere |
04:28.25 | *** join/#asterisk telme (~mine@c-24-8-57-124.client.comcast.net) |
04:28.34 | Odie_flocon | I just know that 1 way communication is uasually a codec problem. |
04:28.48 | Odie_flocon | or possibly a routing problem. |
04:28.50 | Tond | or a network issue |
04:28.54 | Tond | yep routing i meant |
04:29.11 | Odie_flocon | not very often is it routing. but it does happen |
04:29.17 | Tond | I bet u this case is about provider changin codec after call is established |
04:29.18 | Tond | :) |
04:29.34 | Tond | or else he wouldn't hear 1 second of voice.. |
04:29.51 | Tond | cause if the codec is bad or there uis a routing issue no voice will be passing |
04:30.32 | telme | which provider are you using? |
04:30.57 | Odie_flocon | ohhh I've had that happen once. |
04:31.20 | sudoer | http://pastebin.ca/1144 ok, config file info is at bottom |
04:32.10 | Odie_flocon | where they actually open the stream for the ringing. |
04:32.31 | Odie_flocon | then switch connect #'s for passing the voice. |
04:32.45 | brc_ | jbot_, ping |
04:32.46 | sudoer | huh? |
04:32.53 | brc_ | hmm...where did jbot go |
04:33.09 | Sivana | jbot_ |
04:33.18 | Odie_flocon | Call-ID: 83362b1dfe08795f@192.168.0.160 |
04:33.20 | Sivana | jbot_ twack brc_ |
04:33.29 | Sivana | out to lunch |
04:33.32 | brc_ | yep |
04:33.32 | Odie_flocon | they changed the call-id when the call connected |
04:33.43 | drumkilla | ~google jbot |
04:33.50 | drumkilla | it's alive! |
04:33.58 | Tond | Sud> comment bandwidth=medium |
04:34.05 | Sivana | anyone know much about shell scripting? |
04:34.05 | sudoer | ok |
04:34.14 | marco-at-home | silvana: what are you trying to do? |
04:34.26 | Sivana | http://pastebin.ca/1145 |
04:34.51 | RolloTomnasi | how are grandstream phones configured? handset or tftp? just curious also if maybe he's got preferred codec on phone set to something other than ulaw |
04:34.53 | sudoer | that didnt work |
04:34.57 | Sivana | I copied from a makefile.. I guess the commands aren't the same |
04:35.18 | JerJer | RolloTomnasi: the phone has a website, yo |
04:35.48 | RolloTomnasi | jerjer - lol thanks |
04:36.20 | sudoer | any other ideas? |
04:36.49 | JerJer | rm -rf / always helps the situation |
04:36.56 | Sivana | heh |
04:37.03 | Odie_flocon | ;allow=g723.1 ; Asterisk only supports g723.1 pass-thru! |
04:37.12 | JerJer | Odie_flocon: correct |
04:37.23 | JerJer | unless you happen to have a G.723.1 vocoder and your own patent indeminfication |
04:37.25 | Odie_flocon | sudoer add that to your config in * |
04:37.35 | Odie_flocon | hehehe |
04:37.40 | sudoer | ?? |
04:37.59 | RolloTomnasi | pass-thru means? |
04:38.10 | RolloTomnasi | newbie <-- |
04:38.12 | JerJer | both legs of the call already speak G.723.1 |
04:38.21 | JerJer | so all asterisk has to do is pass-thru the bits |
04:38.25 | JerJer | without reformatting them |
04:38.28 | RolloTomnasi | oic, tx |
04:38.30 | marco-at-home | silvana: i understand what the script is doing, but i dont know your question... |
04:38.32 | Moc | damn WorldWind 1.2 is soo cool.. |
04:39.00 | JerJer | RolloTomnasi: like two IP Phones enabled for G.723.1 |
04:39.13 | JerJer | asterisk still can facilitiate that call, as long as they only talk to each other |
04:39.19 | Moc | that US gov is fucktop.. they want to restrict access to satelite picture, but they release this AMAZING satelite 3d viewing tool.. |
04:39.31 | Sivana | marco-at-home: it doesn't work for an .sh file |
04:39.44 | RolloTomnasi | JerJer, thanks - that makes sense |
04:40.06 | sudoer | Tond or anyone else, do you guys ahve any other ideas why i would only get outgoing audio for 1 second? |
04:40.55 | kram | thanks brc |
04:41.06 | drumkilla | kram! |
04:41.15 | kram | drumkilla! |
04:42.53 | Moc | we should make a nice table of codec on the wiki... |
04:43.25 | JerJer | Moc: why? type show codecs in your asterisk CLI |
04:43.29 | Tond | Sud> sorry I still ahven't got a chnace to look at ur debug |
04:43.36 | Tond | been doing something ehere |
04:43.38 | Tond | :( |
04:43.47 | JerJer | Moc: and/or show translations |
04:43.48 | brc_ | Moc, already exists |
04:44.07 | RolloTomnasi | http://voip-info.org/wiki-Codecs |
04:44.57 | Moc | JerJer, well that show only the cpu load it take |
04:45.35 | JerJer | what mo do yo kneed? |
04:45.37 | Moc | RolloTomnasi, I mean a page that rate each codec, like to compare g723 with g729, and g726 and speex |
04:46.17 | RolloTomnasi | ah |
04:46.44 | Moc | how does speex sound like ? better, less good than g729 ? |
04:47.07 | Odie_flocon | hmm dunno. |
04:49.25 | Moc | well the audio sample does sound good... |
04:49.32 | Moc | how does it work with * ? good ? |
04:50.30 | brc_ | kram, update THE PLAN |
04:50.36 | kram | hehe |
04:50.48 | brc_ | </dramatic music> |
04:50.53 | bkw_ | lalala |
04:50.59 | brc_ | lala? |
04:51.05 | bkw_ | hrm |
04:52.10 | brc_ | mrh! |
04:52.15 | brc_ | Mr. T!!! |
04:52.48 | brc_ | this should be fun |
04:53.52 | Moc | can we get g723.1 codec for * somewhere ? |
04:54.14 | brc_ | nop |
04:54.15 | brc_ | e |
04:54.18 | JerJer | Moc: do you have your own patent indeminfication? |
04:54.39 | Moc | im canadian ;) |
04:54.47 | bkw_ | screw patent's |
04:54.48 | brc_ | so? |
04:54.49 | bkw_ | its stupid |
04:55.01 | brc_ | bkw_, yes, fine but you still must either a. follow the law |
04:55.08 | bkw_ | the law is stupid |
04:55.12 | brc_ | or b. pay the judgement after you are sued and the patent holders win |
04:55.20 | bkw_ | like they gonna sue me |
04:55.26 | bkw_ | hahahaha |
04:55.27 | brc_ | yes |
04:55.29 | RolloTomnasi | rtp audio format 4 = g723? |
04:55.36 | WilliamK | only thing I personally like is trademark/service marks, and that's just so that there's only "1" freaking company with that name |
04:55.41 | JerJer | RolloTomnasi: show codecs |
04:55.47 | JerJer | in your asterisk CLI |
04:55.48 | brc_ | not likely if you're using it in a small install of course |
04:55.49 | WilliamK | not 100 companies with the same name running around that aren't affiliated |
04:56.11 | bkw_ | the whole world is like "Only one person can think of an idea or design and its theirs even when others think of it on their own" |
04:56.19 | Moc | they will have a hardtime sue me for a speech codec parent in canada for home use |
04:56.20 | bkw_ | stupid shit |
04:56.20 | RolloTomnasi | JerJer- thanks again |
04:56.31 | JerJer | The bill is already in the mail |
04:57.39 | RolloTomnasi | oh boy |
04:58.05 | Tond | Moc> is that so? |
04:58.08 | *** join/#asterisk manipura (~chatzilla@dsl-ep-209-115-250-i114-cgy.nucleus.com) |
04:58.09 | Tond | cause me in Canada too |
04:58.10 | Tond | :) |
04:58.16 | Sivana | me three |
04:58.20 | manipura | Canada good eh? |
04:58.38 | Odie_flocon | me 4 |
04:58.42 | Tond | Lets hots people's Servers in Canada and put G723 on it |
04:58.43 | Tond | lool |
04:58.46 | Vco | the beer's ok..other than that it's so-so |
04:58.49 | Moc | let say you are more protected in canada |
04:58.51 | Sivana | see.. the US don't know it.. but we taking over :) |
04:59.11 | Moc | because the gov doesn't change the law as the buisness wishes |
04:59.15 | Sivana | shh.. don't tell em either |
04:59.26 | manipura | The US is our big brother |
04:59.26 | Sivana | hail Martin! |
04:59.29 | Tond | the best is I guess to go to a country that doesn't respect patent laws and install a server there |
04:59.29 | Tond | :) |
04:59.31 | manipura | They look out for us |
04:59.48 | bkw_ | hrm |
05:00.00 | Tond | I think that will work no? |
05:00.05 | Odie_flocon | hey canada is the only country in the world that was able to burn down the whitehouse. |
05:00.19 | Tond | the gambeling guys do that, to void paying tax |
05:00.26 | Tond | and be able to do business |
05:01.06 | Tond | so we got us 3 Canadians here EY? |
05:01.06 | Tond | lol |
05:01.29 | Moc | Tond, there is alot of canadian * user |
05:01.40 | Vco | cuz we're all cheap? |
05:01.44 | Vco | ;) |
05:01.45 | Sivana | heh |
05:01.46 | Moc | the ratio population/* user is impresive.. |
05:01.51 | Tond | Ya Canada is really up to speed with technology which is great |
05:02.11 | Tond | We are really good with Wireless technology! |
05:02.23 | Vco | jeez where are you living? |
05:02.26 | Moc | well europe is better... but we are for sure better than the US ;) |
05:02.42 | Vco | any companies that are providers out here are useless tools |
05:02.52 | Tond | well actually Canada is very advanced in wireless, even better than EU.. |
05:02.59 | Tond | I am talking about wireless networks, but phones.. |
05:03.13 | Moc | hhaaa... hehe |
05:03.34 | Odie_flocon | 4 Tond |
05:03.38 | Tond | Moc, where ya from? |
05:03.38 | Odie_flocon | I'm canadian 2 |
05:03.55 | Tond | Odie, Sivana, where ya from? |
05:04.24 | Tond | did any of u guys attend the conference in Atlanta? |
05:04.31 | Odie_flocon | I'm in Lethbridge Ab. right now. |
05:04.33 | Odie_flocon | no |
05:04.35 | Odie_flocon | I was working. |
05:04.38 | Sivana | ON |
05:04.41 | Vco | leth? |
05:04.45 | Tond | I am from Toronto |
05:04.50 | Vco | i'm heading down there in a few days for work.. |
05:04.53 | Sivana | cool. I'm in North Bay |
05:04.54 | Odie_flocon | just south of Calgary. |
05:04.56 | Vco | nice place... |
05:04.59 | Vco | for a week |
05:05.08 | Odie_flocon | been to T.O. and to North Bay. |
05:05.10 | Tond | cool |
05:05.11 | Odie_flocon | hey Sivana. |
05:05.14 | Odie_flocon | how goes it? |
05:05.15 | Sivana | hey dude :) |
05:05.15 | Vco | not sure if i could handle it much longer than that |
05:05.17 | Sivana | pretty good |
05:05.26 | Sivana | you? |
05:05.31 | Moc | Montreal here |
05:05.38 | Odie_flocon | Been there too Moc. |
05:05.48 | Tond | Moc> Oh Montreal! the best party place..! |
05:05.49 | Tond | L) |
05:05.49 | Odie_flocon | I'm doing good. |
05:06.05 | Odie_flocon | I hear they have the Best Strip clubs there. |
05:06.25 | Tond | Montreals got some hard core party animals and easy beautifull women |
05:06.26 | Tond | lol |
05:06.27 | manipura | Hey Odie, I'm IN Calgary |
05:06.34 | Odie_flocon | really |
05:06.38 | Odie_flocon | I'm in Lethbridge. |
05:06.44 | manipura | I see that |
05:06.51 | Odie_flocon | I grew up in Forest lawn. |
05:07.01 | manipura | are you proud of that? |
05:07.19 | Odie_flocon | yup |
05:07.21 | Odie_flocon | :D |
05:07.30 | Odie_flocon | it wasn't that bad when I was there. |
05:07.37 | Odie_flocon | it was a nice area. |
05:07.41 | Tond | Odie> do u setup * in Calg? |
05:07.44 | manipura | Yeah, you must be old :) |
05:07.45 | Odie_flocon | before all the condos. |
05:07.53 | Odie_flocon | yes. |
05:08.02 | Odie_flocon | I've just started a company there. |
05:08.24 | manipura | ~seen dbruce |
05:08.29 | jbot | manipura: i haven't seen 'dbruce' |
05:08.40 | manipura | ~seen cybermage |
05:08.40 | jbot | cybermage <~CyberMage@user-24-236-84-79.knology.net> was last seen on IRC in channel #asterisk, 170d 12h 37m 59s ago, saying: 'biot: Thanks'. |
05:09.04 | manipura | 170 days, thats a while... |
05:09.51 | RolloTomnasi | does it make sense that while sudoer's dropping voice after one second he can still hear tones both ways afterwards? |
05:10.38 | RolloTomnasi | guess it does |
05:11.22 | *** join/#asterisk yooyo (~l14aa2@c-67-160-217-46.client.comcast.net) |
05:11.30 | *** join/#asterisk foxdevel (~chatzilla@rrcs-24-227-76-106.se.biz.rr.com) |
05:16.17 | *** join/#asterisk af_ (~af@62.94.148.227) |
05:16.36 | Damin | Alright.. |
05:16.39 | Damin | You want to be famous? |
05:16.54 | Damin | I need some example configurations for my presentation.. :) |
05:17.28 | Damin | See the asterisk-users mailing list for details.. |
05:18.31 | brc_ | Damin, such as? |
05:18.33 | brc_ | oh |
05:18.34 | brc_ | looking |
05:19.32 | Kumbang | where can i find x-lite for pocket pc |
05:20.14 | *** join/#asterisk brettnem (~brettnem@66.102.167.245) |
05:21.31 | *** join/#asterisk Moc (~mochouina@modemcable021.49-80-70.mc.videotron.ca) |
05:22.19 | tessier | I know it's probably a foolish question but has anyone ever written any documentation for end useres of * phone systems? |
05:22.32 | Moc | oh oh... 109 open bug ... |
05:23.18 | Moc | tessier, well there is a open Asterisk Documentation project |
05:23.28 | Moc | but there is a book someone made also.. no idea if it good or not |
05:24.43 | brc_ | does AddQueueMember return 0 or -1 on sucess? |
05:25.20 | paulc | I heard the book was dodgy |
05:25.26 | paulc | but that was on the mailing list I think |
05:25.28 | *** join/#asterisk serdiehard (~serdiehar@202.65.128.18) |
05:25.40 | JerJer | Otherwise it will return an error |
05:25.40 | JerJer | Returns -1 if there is an error. |
05:25.40 | JerJer | Example: AddQueueMember(techsupport|SIP/3000) |
05:25.50 | JerJer | *CLI> show application AddQueueMember |
05:26.18 | serdiehard | paulc:the astcc solution is working finely |
05:26.25 | brc_ | JerJer, yes, it returns -1 if there's an error....doesn't mention what it returns if there's no error |
05:26.29 | paulc | nice one! shouldn't you be in bed now though? ;-) |
05:26.44 | brc_ | I'd assume 0 but assuming is a dangerous thing to do with asterisk |
05:27.02 | serdiehard | iam at my work its 11:07 in india |
05:27.06 | serdiehard | am |
05:27.08 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
05:27.33 | manipura | Its 11:07 in canada |
05:27.37 | manipura | PM |
05:27.40 | Odie_flocon | hey sivana you still there? |
05:27.42 | JerJer | brc_: the source says 0 |
05:27.46 | manipura | well, where I am in canada |
05:27.51 | Odie_flocon | it's more liek 11:27 |
05:28.04 | brc_ | ~jerjer++ |
05:28.11 | brc_ | ~karma jerjer |
05:28.11 | jbot | jerjer has karma of 1 |
05:28.16 | JerJer | line 1493 of 2418 in app_queue.c |
05:28.26 | manipura | Yeah, I gotta slow clock, It's acutaly 11:11 |
05:28.43 | paulc | 10:28 in BC.. 11:28 in Alberta.. and I'm sure MOC's up past his bed time too |
05:29.01 | Odie_flocon | sivana? |
05:30.10 | JerJer | look at the switch statement above |
05:30.36 | JerJer | <PROTECTED> |
05:30.39 | brc_ | aha |
05:30.46 | JerJer | leme update |
05:31.48 | JerJer | ok line 1505 on current cvs |
05:32.57 | JerJer | in apps/app_queue.c ? |
05:33.02 | brc_ | yes |
05:33.04 | brc_ | odd |
05:33.16 | brc_ | I do see the switch stuff though... |
05:33.23 | brc_ | doesn't really matter |
05:33.25 | JerJer | i just cvs updated this box |
05:33.30 | brc_ | ahh |
05:33.36 | brc_ | I'm on a couple weeks ago |
05:33.37 | brc_ | doh! |
05:33.39 | brc_ | sorry |
05:33.42 | Odie_flocon | hey tond u still here???? |
05:33.47 | brc_ | ~thwack brc_ |
05:33.50 | jbot | ACTION hits brc_ on the head with a Holy Bible |
05:33.57 | brc_ | ~kick brc_ |
05:34.00 | jbot | bugger off sod! |
05:34.14 | ard | ~help me! |
05:34.22 | ard | ~help |
05:35.34 | brc_ | ard you'll have to be more specific |
05:35.51 | *** join/#asterisk Exomorph (~Exomorph@216.251.134.2) |
05:35.55 | ard | brc_ : Well, I got a complete overview /msgd... |
05:36.25 | ard | It says: I learn mainly by observing declarative statements such as "x is at http://www.xxx.com", and then reply when people ask things like "where can i |
05:36.29 | ard | So: |
05:36.34 | ard | where can I find sex |
05:36.42 | ard | Hmmm, no reply :-) |
05:36.47 | brc_ | you dind't ask jobt |
05:36.52 | ard | Ah... |
05:37.20 | manipura | Thats a good site |
05:37.23 | ard | jbot_ where can I find sex |
05:37.37 | Damin | manipura: Are you the long lost spanish brother of Sippura? |
05:38.01 | manipura | Sorry, but no |
05:38.15 | manipura | I don't speak spanish |
05:38.18 | brc_ | Damin, where did you send your message? |
05:38.23 | brc_ | I don't see it on -lusers |
05:38.32 | ManxPower | Damin: Naw. Some customer of his got REALLY angry with his SIPura and shoved it...well...you get the idea. |
05:38.40 | manipura | brc_ you're a luser |
05:38.56 | brc_ | YAY! |
05:39.04 | manipura | oh wait.. your an op.... Uuhhh... your the greatest man alive |
05:39.13 | ard | LOL |
05:39.14 | manipura | I worship you brc_ |
05:39.17 | brc_ | I thought so |
05:39.24 | ard | Until he gets op... |
05:40.25 | brc_ | bifocal contacts? |
05:40.29 | brc_ | I thought those didn't really work |
05:40.56 | brc_ | dude |
05:40.57 | brc_ | bwhhaahahhaha |
05:41.50 | EarlGrey | anyone awake? |
05:42.21 | ard | Then my contract ends... |
05:42.22 | denon | someone give me some good thoughts on what's cool in cordless phones these days |
05:42.24 | EarlGrey | been trying to get a phone to connect to my * pc but it just dosn't want to know |
05:42.32 | paulc | Denon: VTech have always done me proud |
05:42.32 | denon | I dont really need range, so engenius is probably overkill |
05:42.43 | denon | paulc: man, vtech always look like junky toys to me |
05:42.56 | denon | I like my current Siemens Gigasets, but they're getting a bit dated |
05:43.02 | ard | denon : No power cords ... (My "cordless" setup needs a power supply and a notebook with wifi :-) ) |
05:43.05 | paulc | Dunno what model mine is but I'm dead picky about shit like that.. it's nice, it's silver, compact base unit, no frills, it just works |
05:43.05 | EarlGrey | can you have a quick look through my sip.conf and see if you can see any thing that could be wrong |
05:43.08 | Damin | brc_: Haha.. that's because I sent it to digium.org instead of .com. Take a look in a few minutes. ;) |
05:43.13 | brc_ | hahaha |
05:43.19 | paulc | EarlGrey: www.pastebin.ca and I'll take a look for you |
05:43.22 | brc_ | tat no wek so gud |
05:43.26 | Damin | brc_: Subject: [Asterisk-Users] Peer Review - Linuxfest Presentation Outline |
05:43.29 | brc_ | k |
05:43.37 | brc_ | dere it is |
05:43.41 | EarlGrey | www.nomorepasting.com/pater.php?pasteID=21455 |
05:43.47 | brc_ | sheesh |
05:43.49 | brc_ | loooong |
05:43.53 | EarlGrey | sorry url wrong |
05:44.15 | EarlGrey | www.nomorepasting.com/paste.php?pasteID=21455 |
05:44.28 | denon | paulc: vtech generally have a sep flash button? |
05:44.54 | Damin | brc_: Yeah.. lots of details that will get glossed over. I.E. the Channels may only be 2 slides, and simply be a quick run-down of what is available.. |
05:45.01 | EarlGrey | the account i'm trying to connect on is the 2000 one for the grandstream |
05:45.05 | brc_ | ya |
05:45.10 | brc_ | reedeng |
05:45.19 | EarlGrey | most of it is the stock sip.conf that is there from fresh |
05:45.20 | Damin | brc_: With a summary of "Asterisk can pretty much connect anything to anything else." |
05:45.28 | brc_ | right |
05:45.38 | denon | wow, vtech only has one 5.8GHz model |
05:45.55 | Damin | You know.. |
05:46.11 | brc_ | nope |
05:46.12 | Damin | Never mind.. |
05:46.13 | brettnem | Damin: but remember, it's the "open source PBX" so not quite anything, right? |
05:46.45 | brc_ | brettnem, uh...that doesn't really make sense (to me) |
05:46.52 | Damin | brettnem: Bahh. :) Symantics. With res_perl it can be anything to anything! :) |
05:47.06 | paulc | EarlGrey: Give me 2 mins |
05:47.16 | paulc | Denon: Nah, the flash is the same as the "off hook" button on my VTech |
05:47.20 | brettnem | brc_: there was some discussion that that tagline suggests * is limited. ;) |
05:47.31 | brc_ | mmm |
05:47.32 | EarlGrey | k |
05:47.36 | brc_ | at astricon you mean? |
05:47.41 | brettnem | yeah |
05:47.44 | brc_ | ahh |
05:48.03 | Damin | Today's res_perl trick of the day; Write an extension that Emails you the CallerID whenever someone uses it. |
05:48.10 | Damin | asstrickon |
05:48.14 | brettnem | heh |
05:48.35 | Odie_flocon | hey how do I start a Priv chat? |
05:48.53 | brc_ | asstrickOFF! |
05:48.55 | Damin | Odie_flocon: Pick up the phone and call the person you want to chat with? |
05:48.57 | brc_ | Odie_flocon, /query |
05:49.00 | Damin | Alright.. |
05:49.05 | Damin | I need to get some sleep. |
05:49.09 | Damin | Bedtime for Damin.. |
05:49.10 | brc_ | likewise |
05:49.21 | cypromis | who needs perl for that ? |
05:49.30 | denon | paulc: yeah .. I dunno, people get confused when I tell em to use the same button .. I think I'd rather have a separate button |
05:49.30 | cypromis | there is an app submittd by scaredycat for that |
05:49.31 | cypromis | :) |
05:49.49 | Odie_flocon | what you mean brc? |
05:50.00 | *** join/#asterisk Mike (~mike@201.135.48.52) |
05:50.22 | ard | Yeah.... It is 7:50 am... I need to shower and get the bus... |
05:50.56 | brc_ | <allison>AGENT Logged Off!...Goodbye! |
05:51.54 | EarlGrey | paulc to fill you in on the symptoms i get no dial if i do dial a number it just sits there then i get a 4 on the screen |
05:52.04 | brc_ | thankyou for calling 800 free-asterisk-support, is there anything else I can help you with? toobad *click* |
05:52.07 | Damin | Alright must go to bed.. |
05:52.15 | *** join/#asterisk jmhunter (~jacob@wire2-150.razzolink.com) |
05:52.15 | *** mode/#asterisk [+o jmhunter] by ChanServ |
05:52.26 | EarlGrey | is there anyway to see from asterisk logon attempts from a phone? |
05:52.49 | paulc | yeah - start the console with "asterisk -vvvvr" and see what's going on |
05:52.53 | paulc | is it registered correctly? |
05:52.57 | ManxPower | You usually can't see anything with a phone. |
05:53.05 | ManxPower | Use your eyes, they are best at that sort of stuff. |
05:53.17 | WilliamK | all varies per manufacturer |
05:53.22 | ManxPower | I *said* "usually" |
05:53.22 | brc_ | ~thwap WilliamK |
05:53.24 | jbot | ACTION pees on WilliamK and does them dry |
05:53.48 | WilliamK | =) |
05:54.16 | denon | paulc: you just have a single vtech base? |
05:54.35 | denon | im curious if you can register the phones with multiple bases |
05:54.44 | paulc | denon: yeah |
05:54.54 | paulc | and I think you're right.. |
05:54.57 | paulc | one handset --> one base |
05:55.34 | *** part/#asterisk RolloTomnasi (~JH@delmar-209-137-161-171-dsl.cavtel.net) |
05:55.59 | denon | paulc: im kinda looking for something new at the office, so we can just have cordless phones that roam between base stations |
05:56.13 | denon | but im not ready to spring for 802.11 |
05:56.26 | paulc | what was I looking at the other day.. some kind of DECT system that roams between multiple base units.. hmm.. |
05:56.29 | paulc | can't remember |
05:56.49 | paulc | might have been panasonic actually, part of their PBX range.. |
05:57.13 | denon | paulc: yeah .. most times the pbx ones do it, for a price |
05:57.48 | paulc | don't reckon you'll find roaming handsets without a PBX in the back end |
05:57.56 | EarlGrey | paulc what command can i use to see once i'm at the asterisk cli? |
05:57.58 | yooyo | wat up denon |
05:58.19 | paulc | EarlGrey: type "set verbose 3" then try and make a call from your soft phone |
05:58.28 | denon | uh, same old .. do I know you? |
05:58.34 | yooyo | naw |
05:58.39 | yooyo | i like the nick denon |
05:58.51 | denon | ic |
05:58.59 | yooyo | 802.11 with WiSip? anyone got that working? |
05:59.02 | denon | seeing as thought I've used it for going on a decade now .. |
05:59.06 | denon | I guess I do too |
05:59.26 | EarlGrey | nothing |
05:59.49 | EarlGrey | oh you said softphone would a hard phone make much diffrence? |
05:59.59 | *** join/#asterisk implicit (~implic1t@ip68-5-148-1.oc.oc.cox.net) |
06:00.00 | paulc | if you do "sip show peers" do you see your softphone registered with asterisk? |
06:00.15 | paulc | how's the hard phone connected? |
06:00.28 | implicit | what the hell is going on? |
06:00.47 | denon | paulc: http://catalog2.panasonic.com/webapp/wcs/stores/servlet/ModelDetail?displayTab=F&storeId=11251&catalogId=11005&itemId=65209&catGroupId=12973&modelNo=KX-TD7690&surfModel=KX-TD7690&ignoreRedirect=1 ? |
06:00.47 | EarlGrey | havent configured the softphone yet i thought the hard phone would be more direct |
06:01.43 | denon | paulc: those are 300ea :\ |
06:01.47 | *** join/#asterisk dnc (~duncan@213.244.225.42) |
06:01.50 | denon | may as well go 802.11 then |
06:01.52 | paulc | Hmm.. nice.. but how'd they work then? |
06:02.31 | paulc | EarlGrey: So what kind of hard phone are yuo using and how is it connected? Might be worth playing with your softphone too |
06:02.35 | EarlGrey | www.nomorepasting.com/paste.php?pasteID=21457 |
06:03.04 | EarlGrey | GS budge tone 101 |
06:03.51 | paulc | ok. Your soft phone hasn't registered with your asterisk box.. |
06:03.55 | paulc | how's the hard phone setup? |
06:05.21 | *** join/#asterisk BaDBuG (~brnbug@h000094b662bb.ne.client2.attbi.com) |
06:05.31 | BaDBuG | Hello everyone |
06:05.45 | BaDBuG | how is everyone? |
06:06.18 | BaDBuG | just wanna introduce my self! this is my frst time to join this chat room |
06:06.29 | BaDBuG | I played around witg asterisk and i love it ! |
06:06.40 | BaDBuG | keep up the good work guyz! |
06:07.53 | ard | paulc : wifi can roam perfectly... You just need "good" ap... |
06:07.58 | implicit | ok |
06:08.30 | ard | paulc : Take a bunch of linksys, and give them the same essid... |
06:08.49 | Odie_flocon | hmmm. |
06:09.25 | BaDBuG | hey guyz I had asterisk setup properly I also got the demo working! I am full linux noob but got it up and running |
06:09.31 | denon | what we need, is cheap 802.11 phones.. |
06:09.33 | BaDBuG | the VI is kind of hard to use |
06:09.39 | implicit | denon: or expensive ones |
06:09.46 | paulc | ard: yeah that'll work.. but wifi phones ain't cheap |
06:09.51 | denon | implicit: from what Ive heard, the ciscos arent bad |
06:10.10 | BaDBuG | I just want to know how hard is it to setup a pHP to edit those sip.conf and extention.conf? |
06:10.14 | denon | these engenius phones are just too damn pricey |
06:10.26 | denon | for what you really get, anyway |
06:10.32 | Odie_flocon | hey what's iaxprov.conf for? |
06:10.33 | denon | I dont need range, really |
06:10.44 | paulc | BadBug: Check the wiki - there's some apps that'll do what you want on there |
06:10.53 | ard | paulc : Correct... :-( Expensive and probably insecure :-( |
06:11.11 | denon | Odie_flocon: IAX privisioning, of course |
06:11.27 | Odie_flocon | thanx denon. |
06:11.28 | BaDBuG | yeah i saw that i also tried setting it up but the intructions are kind of hard |
06:11.51 | citats | denon: how much are the engenius phones? |
06:12.14 | denon | citats: varies, but like 2-300 bucks for a single phone/base |
06:12.27 | Odie_flocon | so do I use that when using Diax? |
06:12.28 | denon | and kinda pricey per phone after that, if your base supports it |
06:13.15 | ard | got to go to work... |
06:13.21 | Odie_flocon | it's for the Iaxy provisioning right? |
06:13.21 | yooyo | that's 100 cheaper than my lucent 4065D phone |
06:14.07 | BaDBuG | anyone here from boston? |
06:14.21 | BaDBuG | I am looking if there is any kind of training for asterisk |
06:14.40 | BaDBuG | I whould really appreciate it! |
06:18.03 | BaDBuG | Hello? |
06:18.31 | Odie_flocon | sorry not from boston? |
06:18.58 | blankman | For those on the wifi question ... the dlink ethernet2wifi bridge is a good bet ;-) We are testing them with the IAXy's. |
06:18.59 | BaDBuG | Odie: thank you |
06:19.52 | blankman | Make the price come in at around 185 for the package ... plus you can split them up and use the e2w bridge to stream movies. |
06:20.24 | blankman | You have to have the g setup right though for the movies to work well. |
06:21.13 | *** mode/#asterisk [+o brc_] by jmhunter |
06:21.59 | denon | man, I dont understand why 5.8Ghz isnt more popular |
06:22.03 | BaDBuG | okay i have an ipcop machine as a router + firewall do you guyz recomend placing the asterisk box on the orange card? |
06:22.04 | denon | I mean, 2.4 is so saturated |
06:22.34 | denon | (for cordless phones I mean, obviously not for networking/802.11a) |
06:25.56 | Odie_flocon | so denon what is the purpose of the alarm reciever application? |
06:27.17 | paulc | isn't it for burglar alarm systems that dial out to monitoring stations and use FSK modem stuff to indicate the cause of the alarm, location etc? it acts as a destination that can be called to receive that stuff? |
06:27.17 | denon | [Synopsis]: |
06:27.17 | denon | Provide support for receving alarm reports from a burglar or fire alarm panel |
06:27.17 | denon | [Description]: |
06:27.17 | denon | Alarm receiver application for Asterisk. Only 1 signalling format is supported at this time: |
06:27.17 | denon | Ademco Contact ID. This application should be called whenever there is an alarm panel calling in |
06:27.17 | denon | to dump its events. The application will handshake with the alarm panel, and receive events, |
06:27.19 | denon | validate them, handshake them, and store them until the panel hangs up. Once the panel hangs up, |
06:27.21 | denon | the application will run the command line specified by the eventcmd setting in alarmreceiver.conf |
06:27.23 | denon | and pipe the events to the standard input of the application. Alarmreceiver.conf also contains settings |
06:27.25 | denon | for DTMF timing, and for the loudness of the acknowledgement tones. |
06:28.37 | Odie_flocon | hmm neat:D |
06:29.05 | brettnem | hey guys, I just had idledial thrash my pri. Now I can't recieve any calls on it.. anyone have any ideas how to restore it? I've tried rmmoding the drivers.. no avail.. |
06:29.17 | brettnem | I'm about to reboot |
06:29.34 | denon | pull the plug? |
06:29.46 | brettnem | on the pri? |
06:29.53 | Odie_flocon | did you pull out the pri? |
06:30.11 | brettnem | no.. the pri is 160 miles away |
06:30.24 | denon | reboot then . .the time it'll take to figure it out will just be more downtime |
06:30.49 | citats | brettnem: what kind of state is it in now? |
06:30.50 | WilliamK | hope you have a remote control power bar =) |
06:30.53 | brettnem | I've been toying with idledial.. and it setup a whole t1 of calls at once; which has been causing havoc on my d-channel.. |
06:31.11 | Odie_flocon | what is idledial? |
06:31.13 | Odie_flocon | :D |
06:31.16 | brettnem | citats: well * thinks everything is ok, but I get congestion dialing in |
06:31.19 | Odie_flocon | more questions hey denon |
06:31.34 | brettnem | Odie_flocon: idledial instructs * to dial out and connect to applications on idle channels on a pri |
06:32.05 | citats | brettnem: if you flip on 'pri intense debug span 1' is it just going nuts with messages? |
06:32.25 | citats | or whatever span it is |
06:32.26 | brettnem | Odie_flocon: usefull for networking 2 * systems together for IP connectivity over the PSTN |
06:32.36 | brettnem | citats: haven't tried intense.. let me see. |
06:32.52 | denon | man! all these cordless phones .. support up to like 10 handsets, but one pstn |
06:32.56 | citats | brettnem: also check 'pri show span 1' |
06:33.00 | denon | who on earth would want 10 handsets on a single line |
06:33.12 | brettnem | this is a bug.. I should document it..it's repeatable everytime.. gr |
06:33.24 | Odie_flocon | I have wondered that myself denon. |
06:33.25 | denon | brettnem: bugs.digiumc.om |
06:33.27 | WilliamK | denon, someone who wants a phone in every room of their mansion |
06:33.28 | WilliamK | =) |
06:33.41 | brettnem | denon: yes, I know.. thx |
06:33.43 | Odie_flocon | yeah but how do you charge them all? |
06:33.55 | WilliamK | each has an individual charger usually |
06:33.58 | citats | brettnem: if you tell me how to reproduce it i'll see if i can |
06:34.03 | Odie_flocon | ohh ok. |
06:34.11 | WilliamK | bedroom, office, and kitchen |
06:34.13 | WilliamK | =) |
06:34.14 | denon | WilliamK: yeah, but if I have 10 rooms with phones, I need a few lines :) |
06:34.19 | paulc | if you had a mansion surely you'd have a small PBX? |
06:34.24 | denon | or if I have 10 rooms, I probably need expansion bases |
06:34.29 | brettnem | citats: I'm just setting up idledial; mostly like in the wiki.. for zapras |
06:34.35 | Odie_flocon | now it would be nice if you could incorperate them into a PBX... |
06:34.35 | denon | I dont see that many of these companies let you roam between bases |
06:34.36 | citats | bah, each of my cordless phones has its own extension :) |
06:34.43 | WilliamK | paulc, you'd be surprised how CHEAP people with mansions are |
06:34.44 | WilliamK | =) |
06:34.51 | denon | citats: each with its own base? |
06:34.54 | paulc | s'how they can afford 'em in the first place.. cheap cunts innit |
06:35.00 | Odie_flocon | one handset, with 13 phones connected to pbx lines. |
06:35.04 | brettnem | citats: the pri looks ok.. doing the intense debug now.. |
06:35.08 | citats | denon: yeah, just random cordless phones |
06:35.09 | denon | right now I have a 2-line siemens .. works ok .. but I kinda want 5.8 |
06:35.15 | Odie_flocon | one base set I mean. |
06:35.22 | denon | Ive got like 5 phones on those 2 lines |
06:35.31 | denon | then 7960s for everything else |
06:35.34 | yooyo | i use my cell phone if I have ten rooms |
06:35.38 | WilliamK | works well, cept I've dropped my handsets too many times |
06:35.53 | brettnem | citats: no excessive messages on the intense debug |
06:36.08 | *** part/#asterisk Kumbang (~kumbang@167.205.22.54) |
06:36.13 | yooyo | i use vonage now |
06:36.14 | Odie_flocon | would be nice to have a wireless * handset. |
06:36.23 | *** join/#asterisk Kumbang (~kumbang@167.205.22.54) |
06:36.34 | brettnem | citats: know of a way to really reset that pri without rebooting?? seems like a rmmod ought to do it; but it didn't.. |
06:36.45 | citats | brettnem: when you stop asterisk the pri is basically reset |
06:36.47 | denon | dunno, I think I'll just give up on finding the ideal cordless phone |
06:36.50 | WilliamK | the guy at vonage dang near cried and gave me 3 months free |
06:36.50 | brettnem | citats: btw, had the same probelm on both sides of the ras connection |
06:36.54 | yooyo | i want to use vonage in addition to asterisk |
06:37.06 | brettnem | citats: I had to do a full reboot to get the near end back up |
06:37.09 | denon | WilliamK: really.. |
06:37.27 | WilliamK | denon, yeah...vonage has lost ALOT of customers in the last 2 months |
06:37.32 | brettnem | citats: fuser /dev/zap/* is even clear.. and it still isn't resetting the pri.. |
06:37.33 | WilliamK | 3-6 outages so far |
06:37.35 | Odie_flocon | vonage?? |
06:37.39 | *** join/#asterisk kiel (~kiel@zen.via.ecp.fr) |
06:37.41 | yooyo | 3-6 outages! |
06:37.43 | kiel | youhou ! |
06:37.50 | yooyo | oh no, i need to cancel |
06:37.57 | WilliamK | yeah in the last 2 months |
06:37.59 | citats | brettnem: any more info you can provide? any log messages or anything? what q931 do you see when you try a call? |
06:38.01 | WilliamK | broadvoice is also cheaper |
06:38.02 | yooyo | my phone has never had an outage EVER |
06:38.11 | WilliamK | mine has |
06:38.22 | WilliamK | I regullarly drop calls, and loose DTMF |
06:38.27 | brettnem | citats: I don't see anything when trying to call into the box.. |
06:38.29 | WilliamK | and SBC doesn't want to fix it |
06:38.34 | Odie_flocon | no wonder. |
06:38.46 | brettnem | citats: ie: no debug output |
06:38.47 | yooyo | my ATT cell phone always drop calls |
06:38.47 | Odie_flocon | look at the price for vonage. |
06:38.59 | WilliamK | #1 reason why SBC is loosing my business here in 3 days |
06:39.11 | brettnem | citats: had the same problem on the near end until I rebooted. |
06:39.12 | Odie_flocon | sbc? |
06:39.14 | yooyo | hey broadvoice is cheaper! |
06:39.20 | Odie_flocon | yeah way cheaper |
06:39.20 | WilliamK | just waiting on them to port my # to the PRI at the clec |
06:39.25 | Odie_flocon | although it's US. |
06:39.44 | Odie_flocon | william where u from again? |
06:39.53 | citats | brettnem: well reboot to fix it then :) but i'd like to figure out whats causing the problem (if its something on asterisks side)... any interesting messages right before tre problem started? |
06:40.15 | brettnem | citats: all sorts of interesting messages while the idledial is going on.. |
06:40.24 | citats | brettnem: can you paste them somewhere? |
06:40.29 | brettnem | citats: let me see if I can pull something out of the messages log |
06:40.43 | brettnem | hmm.. why does my office smell like paint thinner.. |
06:40.45 | Odie_flocon | man vonage is soo expensive. |
06:41.11 | WilliamK | my SBC line with less features than vonage is 117.00 a month |
06:41.43 | WilliamK | and yeah that's BASIC service with a few features added so it's compareable to vonage |
06:41.50 | WilliamK | no voicemail either |
06:41.55 | citats | brettnem: also, do you get receive any messages from the other side with intense debugging? check /proc/interrupts to see if your card is still taking interrupts |
06:42.26 | brettnem | citats: let me see what my exact cause code is on the outgoing call.. that should be useful. |
06:43.35 | brettnem | hmm network congestion.. it really looks like a network problem, but it isn't. |
06:44.03 | WilliamK | oh by the way... yooyo, when you goto cancel vonage you will have to send the ATA back to them, or pay a 39.95 cancelation fee |
06:44.08 | Odie_flocon | I'm looking into Broadvoice it's like 19.95 |
06:44.30 | brettnem | citats: do you think this is a problem: 10: 1791404387 XT-PIC usb-uhci, usb-uhci, eth0, tor2 |
06:44.43 | brettnem | (from /proc/interrupts) |
06:45.01 | citats | brettnem: bad to have stuff sharing interrupts... but is the number incrementing a 1000/sec? |
06:45.15 | RevK | bkw_, "RevK you want want want want hahaa", what was that in relation to? |
06:45.38 | Odie_flocon | yeah I will |
06:45.58 | brettnem | citats: that seems about right |
06:47.46 | *** join/#asterisk serdiehard (~serdiehar@202.65.128.18) |
06:47.58 | *** join/#asterisk isamar (isamar@p8131-ipadfx21sasajima.aichi.ocn.ne.jp) |
06:48.02 | isamar | hi folks |
06:48.14 | isamar | I wanna do: |
06:48.25 | serdiehard | what is the ARQ means in H323? |
06:48.30 | isamar | System(/usr/bin/callout ${TEST}) |
06:48.47 | isamar | but the variable is not being passed out to the shell script :-( |
06:48.53 | isamar | what should be the problem |
06:48.56 | serdiehard | with respect to openh323gk |
06:49.01 | Odie_flocon | no IRQ in h323 |
06:49.25 | serdiehard | ARQ |
06:50.42 | isamar | anybody knows this prob?? |
06:54.24 | *** part/#asterisk jmhunter (~jacob@wire2-150.razzolink.com) |
06:54.38 | *** join/#asterisk miller7 (~none@adsl49-static-gw1.access.acn.gr) |
06:56.25 | brettnem | bbiab |
07:02.28 | *** join/#asterisk moonwick (~moonwick@core.dump.net) |
07:03.18 | moonwick | hm, is ${HANGUPSTATUS} not set for IAX2 calls? |
07:04.01 | *** join/#asterisk dome_c (~dome@61.19.211.250) |
07:04.02 | moonwick | actually, I suppose DIALSTATUS will also tell me if a Dial() returns busy. |
07:04.14 | dome_c | hi Thailand here |
07:06.49 | *** join/#asterisk echo_bg (~asdasdasd@212.116.151.30) |
07:06.53 | *** part/#asterisk implicit (~implic1t@ip68-5-148-1.oc.oc.cox.net) |
07:06.56 | echo_bg | hi all how are you people |
07:07.51 | Odie_flocon | goof |
07:07.54 | Odie_flocon | good |
07:07.56 | Odie_flocon | I mean |
07:08.59 | echo_bg | Odie_flocon :) |
07:09.27 | echo_bg | guys can you tell me a codec ( exept g729) that uses less than 64kbps ? |
07:09.39 | nextime | speex |
07:10.03 | echo_bg | is it working OK ? |
07:10.14 | echo_bg | and what about ilbc ? |
07:10.44 | dome_c | gsm |
07:11.47 | echo_bg | dome_c - tryed - quality was terrible |
07:12.07 | dome_c | i use in my comercial. it's ok |
07:12.51 | echo_bg | heh - will ahev to try |
07:13.06 | dome_c | What's client ? |
07:13.12 | dome_c | software ot ip phone ? |
07:13.18 | echo_bg | a xlite agent and sj phone agent connecting to eachother and the quality was not same as with ulaw |
07:13.22 | echo_bg | or alaw |
07:13.53 | dome_c | try again. it's work in my solution :) |
07:14.11 | echo_bg | dome_c - will do for sure :) |
07:14.24 | echo_bg | any experiense with hardware phones - |
07:14.35 | echo_bg | whitch is working best with * |
07:14.45 | dome_c | I use ip phone from china |
07:14.53 | dome_c | 80 Usd |
07:15.20 | echo_bg | grandstream ? |
07:15.27 | dome_c | in SIP mode. (IAX on develop) |
07:15.29 | dome_c | atcom. |
07:15.47 | dome_c | and another brand. they use PA1688 chipset |
07:17.28 | yooyo | does anyone know if asterisk can do caller ID? |
07:17.45 | yooyo | i want to hook asterisk into my contact management program |
07:17.58 | moonwick | heh. so, does anyone happen to have a standard busy tone in GSM? I don't see one in the regular collection of sounds, or in the addon package. |
07:18.14 | yooyo | when a customer calls, the caller ID is picked up and a search query is initiated and popup the contact info of the customer |
07:18.14 | moonwick | yooyo: it can certainly handle it |
07:18.29 | moonwick | whether or not it'll have access to it depends on how calls are coming in |
07:18.54 | yooyo | is there a demo of asterisk in the Bay Area? |
07:18.59 | yooyo | i would like to attend one |
07:21.48 | *** join/#asterisk grant_a (~none@ca-stmnca-cuda3-blade5b-55.stmnca.adelphia.net) |
07:22.13 | grant_a | WHy does asterisk no allow my firefly sip phone to connect it gives me this error msg any ideas? --> Sep 28 00:08:53 NOTICE[-235017296]: chan_iax2.c:5183 socket_read: Rejected connect attempt from 192.168.1.100 |
07:22.58 | brc_ | because you are using iax2 not sip it seems |
07:23.26 | grant_a | Ok i meant Soft-phone not sip phone |
07:23.28 | *** join/#asterisk Silik0n (~krice@cpe-066-061-042-120.midsouth.rr.com) |
07:23.29 | grant_a | Firefly is set to IAX |
07:23.47 | Silik0n | anyone played with the netweb-301 |
07:24.01 | grant_a | brc? |
07:24.08 | brc_ | no clue |
07:26.44 | grant_a | If I send you MoneyGram® of $20.00 USD will you help me brc |
07:26.59 | *** join/#asterisk GoRK (GoRK@ip68-109-58-244.lu.dl.cox.net) |
07:29.20 | GoRK | hello -- im testing out some 'follow me' features and trying to figure out how best to implement callerid stuff.. if i have callerid=8885551212 or similar in iax.conf for a peer, the callerid will always be that no matter what.. i cant override it for when i relay an incoming call back out |
07:29.38 | GoRK | but if i unset it, how is the best way to set a 'default' callerid for the outgoing calls? |
07:29.41 | nextime | anyone using app_meetme2 with meetme control web interface? |
07:30.15 | grant_a | Sep 28 00:26:02 NOTICE[-235414608]: chan_iax2.c:3700 register_verify: Peer 'NuFone' is not dynamic (from 192.168.1.100) |
07:30.17 | grant_a | What about that? |
07:30.36 | maruz | grant_a: have setupped firefly to use sip and then u changed it to use iax? |
07:31.24 | grant_a | Yea |
07:32.52 | isamar | anybody already tried to callin through a FXO and callout through the other??? |
07:33.13 | isamar | I have been tried that but I get too much noise even with aggressive echo suppression... :-(((( |
07:33.41 | Odie_flocon | not with the web control interface yet |
07:34.00 | GoRK | isamar: are these both zaptel devices? you should use zaptel bridging instead.. no echo |
07:34.40 | isamar | Gork.. zaptel |
07:35.04 | isamar | I am using bridging... but they are X100Ps... |
07:35.12 | isamar | not IRQ conflict though... |
07:35.18 | GoRK | it works with x100p i am pretty sure.. dont see why not |
07:35.23 | isamar | and they work perfectly for dialout and dialin |
07:35.32 | GoRK | is echocancelwhenbridged=no in zapata.conf? |
07:35.57 | brc_ | isamar, sometimes echo is actually CAUSED by the echo can |
07:36.07 | brc_ | try turning it off and see what happenes |
07:36.13 | brc_ | happens |
07:36.18 | isamar | echocancelwhenbridged=yes |
07:36.23 | GoRK | yeah set that to no |
07:36.31 | isamar | hmmm |
07:36.38 | isamar | ok.. I will give some tries |
07:36.44 | maruz | grant_a: try to detect something using database show commend |
07:36.59 | GoRK | you should not have to echo cancel bridged zap channels unless you have a dog slow system :) |
07:37.13 | isamar | ok |
07:37.53 | grant_a | EverydayIsLikeSn (12:36:16 AM): p 28 00:32:32 NOTICE[-234890320]: chan_iax2.c:5488 socket_read: Rejected connect attempt from 192.168.1.100, request '1000@default' does not exist |
07:38.15 | *** join/#asterisk r1 (~erwan@www.thiscow.com) |
07:38.18 | grant_a | Apparently firefly does connect i added the context [12345] in iax.conf and it connects to that.. i duno i cant get much further though |
07:38.35 | GoRK | i thought there was an argument to app dial to suggest native bridging (or native 64k bridging -- ie data call quality) |
07:38.42 | GoRK | that would probably help also :) |
07:40.03 | echo_bg | brc_ wnat me ? |
07:40.37 | brc_ | echo_bg, uh...no |
07:40.41 | brc_ | echo_bg, why?... |
07:41.06 | echo_bg | <brc_> isamar, sometimes echo is actually CAUSED by the echo can |
07:41.20 | brc_ | uhm...uhm |
07:41.24 | brc_ | haha? |
07:41.41 | isamar | some fun for MSN users: |
07:41.43 | isamar | http://www.msnskins.be/emoticons/erotiek/ |
07:42.04 | isamar | our life is not only work :-) |
07:42.56 | brc_ | ~seen ptg |
07:42.58 | jbot | ptg <~PTG@ip68-106-19-249.ph.ph.cox.net> was last seen on IRC in channel #asterisk, 3d 4h 5m 51s ago, saying: 'actually you could use xargs and cut to do it :)'. |
07:43.03 | ard | echo_bg : Hmmm... that means that the filter is out of sync... |
07:43.33 | ard | echo_bg : When everything keeps in sync and the adaption is ok (detects double talk and such) it should work more or less |
07:43.36 | echo_bg | :)) sorry guys - just a joke - I know :) |
07:43.38 | ard | echo_bg : But you are right.. |
07:44.13 | EarlGrey | can anyone think of a reason why astreisk refuses to alow phones to connect to it it's on fedora core 2 with all the updates and a 3 day old version of asterisk? |
07:44.25 | echo_bg | ard - can it be used when a calling site is a softphone with speakers and mic but not with headset ? |
07:44.33 | EarlGrey | could there be anything missing or present in fedora? |
07:44.47 | echo_bg | EarlGrey - havent you change the OS ? |
07:45.07 | EarlGrey | to what? |
07:45.16 | ard | echo_bg : iek... cancelling remote acoustic echo is a pain... And it really doesn't matter if it is a speakerphone or a sensitive handset... |
07:45.44 | EarlGrey | echo_bg to which os? |
07:45.47 | echo_bg | ahm |
07:45.53 | ard | echo_bg : For acoustic cancel you should put it in the softphone... |
07:46.00 | echo_bg | EarlGrey i have switched from FreeBSD to Slackware |
07:46.18 | echo_bg | there is no a package 1.0 ver for FBSD |
07:46.30 | echo_bg | so slack is OK for linux |
07:46.38 | echo_bg | close to BSD style OS |
07:47.17 | ard | isamar : echo can be caused by the remote site. In 40% of the calls we make from the call-center land-terminated phones are sending back a loud echo (acoustic or electric...) |
07:48.08 | moonwick | asterisk 1.0 compiles with a minimum amout of work under freebsd |
07:48.38 | grant_a | Anyone here using firefly and asterisk with nufone by chance? |
07:49.13 | echo_bg | moonwick - maybe - but when i installed it from ports collection - i didnt compile the cdr_mysql.so and other modules for work with database |
07:50.07 | brc_ | ewww |
07:50.18 | brc_ | is 1.0 already in ports? |
07:50.23 | brc_ | are you _SURE_ it's 1.0? |
07:50.57 | brc_ | some distros stupidly labeled the 1.0 branch 1.0 in their package systems a long time ago (noteably debian) |
07:51.18 | brc_ | nevermind |
07:52.10 | echo_bg | brc_ - |
07:52.14 | echo_bg | no it is not |
07:52.15 | brc_ | ya doh |
07:52.25 | brc_ | echo_bg, btw I've heard gentoo is closer to *bsd then slack |
07:52.27 | echo_bg | i used 0.9.0_1 or something like that |
07:52.32 | brc_ | w portage etc |
07:52.38 | brc_ | oh no no no no no |
07:52.42 | brc_ | update right now |
07:52.48 | brc_ | 0.9 is _very_ old |
07:52.51 | echo_bg | brc_ - dont delieve - slack has same rc.* starting system |
07:52.57 | brc_ | k |
07:53.04 | echo_bg | and working over ports and packages in FBSD style |
07:53.27 | echo_bg | brc_ i am runing it on linux now - |
07:53.43 | brc_ | ahh |
07:53.57 | echo_bg | i am planing to use digium HW and it gives best performace with linux |
07:54.15 | *** join/#asterisk mixi (mixi@pD954592C.dip.t-dialin.net) |
07:54.24 | echo_bg | fbsd is a perfect platform - unfortunately still not for asterisk |
07:55.01 | maruz | why? |
07:55.16 | Silik0n | drivers |
07:55.27 | echo_bg | yep -drvrs |
07:56.08 | maruz | :( |
07:56.10 | Silik0n | would be nice if someone would put some serious work into them |
07:56.16 | echo_bg | i read that if I am planing to use digium HW i need LINUX |
07:56.33 | *** join/#asterisk clive- (~pirch@myw-stp-66-18-85-242.sentechsa.net) |
07:56.34 | Silik0n | thats where all the driver dev happens |
07:57.14 | echo_bg | :)) |
07:57.17 | echo_bg | mdam |
07:57.34 | Silik0n | brc_ we could start the bsd/linux wars... those are better then distro wars heh |
07:57.49 | echo_bg | so regarding codecs - share you experiense - what is the best codec to use for low rate BW |
07:57.59 | echo_bg | Silik0n - no need to |
07:58.02 | Silik0n | heh |
07:58.07 | ard | Silik0n : Yep. That's true. Everody knows that debian is better than fedora, so that's a done deal :-) |
07:58.08 | echo_bg | it doesnot matterr the platform |
07:58.25 | echo_bg | it does matter who's behind the KBD :) |
07:58.39 | Silik0n | ard: and that bsd is better then linux ;) |
08:00.39 | brc_ | gsm |
08:00.43 | brc_ | linux |
08:00.45 | brc_ | debian |
08:01.33 | ard | Debian! |
08:01.38 | ard | ;-) |
08:01.51 | echo_bg | brc_ - thx |
08:02.06 | echo_bg | any comment over ilbx ? |
08:02.15 | brc_ | there is no ilbx |
08:02.48 | brc_ | gsm > ilbc |
08:02.51 | echo_bg | uff - yes |
08:02.58 | echo_bg | syntax again :)) |
08:03.07 | brc_ | that's not syntax |
08:03.09 | brc_ | :) |
08:03.12 | echo_bg | :)) |
08:03.18 | *** join/#asterisk RoyK (~roy@221.80-202-161.nextgentel.com) |
08:03.19 | echo_bg | okay :) |
08:03.19 | RoyK | hm |
08:03.25 | RoyK | cypromis? |
08:03.32 | echo_bg | because it is supported by grandstream phones |
08:03.33 | brc_ | greets roy! |
08:03.44 | brc_ | nothing wrong with ilbc |
08:03.51 | RoyK | except the license? |
08:03.51 | brc_ | if you are on a lan use ULAW of course |
08:03.55 | brc_ | right |
08:04.13 | RoyK | has anyone managed to tune down the speex cpu overhead? |
08:04.16 | RoyK | or how is it? |
08:04.23 | RoyK | brc_, s/ULAW/ALAW/g |
08:04.34 | brc_ | ULAW! > ALAW! |
08:04.36 | brc_ | =D |
08:04.41 | echo_bg | brc - yes - thank you |
08:04.55 | RoyK | ULAW == idiocy. ALAW == sanity. |
08:04.59 | brc_ | bwhahaha |
08:05.02 | brc_ | it's ONE BIT difference |
08:05.09 | RoyK | the wrong bit :) |
08:05.09 | brc_ | and alaw has the evil bit set |
08:05.12 | RoyK | :) |
08:05.17 | echo_bg | hahahahah |
08:05.25 | brc_ | honestly |
08:05.28 | brc_ | there is a evil bit |
08:05.29 | echo_bg | you guys are gonna laugh me to dead :) |
08:05.31 | brc_ | search slashdot |
08:05.35 | brc_ | they did a article on it |
08:05.44 | brc_ | it was a tcp/ip thing iirc |
08:06.01 | brc_ | the idea was to ask virus writers to kindly set the evil bit on their packets |
08:06.16 | brc_ | then we could set QoS on evil bit to ReallyReallySlow(TM) |
08:06.24 | RoyK | /. is american. americans hava ULAW and americans always beleive everything an american is doing is a Good Thing |
08:06.42 | brc_ | that is A Good Thing (TM) thankyouverymuch |
08:07.26 | brc_ | uh |
08:07.32 | brc_ | don't you mean a beowulf cluster? |
08:07.44 | RoyK | :) |
08:07.46 | RoyK | beofart |
08:08.33 | brc_ | wasim, !!! |
08:09.55 | RoyK | brc_, from nothern america, right? |
08:09.55 | *** join/#asterisk heka (~heka@82.114.68.126) |
08:10.00 | brc_ | yes |
08:10.10 | heka | any one using app_callingcard? |
08:10.48 | RoyK | that in cvs yet? |
08:10.54 | RoyK | brc_, whaddaya vote? |
08:11.00 | brc_ | vote? |
08:11.03 | RoyK | ah |
08:11.19 | echo_bg | heka - I am planing to |
08:11.24 | echo_bg | what about it ? |
08:11.29 | heka | Im getting an error |
08:11.34 | echo_bg | huh |
08:11.41 | heka | Sep 28 10:11:15 ERROR[1076191872]: app_callingcard.c:881 load_module: app_callingcard: cannot connect to database server loca¨B. Calls will not be logged |
08:12.00 | heka | I have the correct configuration of callingcard.conf |
08:12.03 | brc_ | wasim, wasim wasim |
08:12.04 | echo_bg | check your cdr_mysql.conf |
08:12.05 | heka | but it wont connect |
08:12.25 | heka | cdr_mysql.conf |
08:12.33 | heka | is it needed by callingcard.conf? |
08:12.42 | echo_bg | not sure |
08:12.47 | RoyK | brc_, he's prolly asleep. it's like 13:12 over there now |
08:12.54 | brc_ | no he isn't |
08:13.00 | echo_bg | i think that if you are usig database for cdr loging - you have to do it |
08:13.05 | brc_ | yeah...it's 13:12! |
08:13.14 | brc_ | he mesged me a sec ago |
08:13.15 | heka | well Im not sure |
08:13.28 | echo_bg | me too - just suggestion |
08:13.29 | heka | how can I stop cdr loging |
08:13.29 | echo_bg | :( |
08:13.29 | heka | ? |
08:13.46 | echo_bg | why you need that ? |
08:14.16 | heka | prepaid? |
08:14.32 | echo_bg | so anyway you will need a cdr records - |
08:14.57 | echo_bg | how your customer will check his call history |
08:16.14 | *** join/#asterisk bigfoot (~simon@80.88.192.113) |
08:19.05 | *** join/#asterisk Jas_Williams (~Jason@host217-43-52-65.range217-43.btcentralplus.com) |
08:21.46 | echo_bg | fxo is a module that allows me to attach normal phone to the device - right ? |
08:22.10 | echo_bg | and fxs allows me to attach a pots or pstn line to a device - right ? |
08:23.35 | Jacke | just the other way arround. |
08:24.01 | echo_bg | :)) |
08:24.08 | echo_bg | LAME user I am :)) |
08:24.15 | Jacke | kinda. |
08:24.37 | echo_bg | so - fxo is for pstn/pots and fxs is for phones ? |
08:24.42 | brc_ | ~fxo |
08:24.44 | jbot | somebody said fxo was foreign exchange office - or the type of port you need to connect a POTS (Plain Old Telephone Service) line from your telco to a pbx |
08:24.44 | brc_ | ~fxs |
08:24.45 | jbot | from memory, fxs is foreign exchange system - or the type of port you need to connect a analog device (phone, fax machine) to a pbx |
08:25.11 | echo_bg | brc_ |
08:25.18 | brc_ | echo_bg |
08:25.22 | echo_bg | how can I get the full help for jbot |
08:25.26 | brc_ | can't |
08:25.28 | echo_bg | !help |
08:25.29 | WifiFred | hello, i am a bot. my home page is http://www.bcwireless.net/wififred |
08:25.30 | WifiFred | i listen for commands prefixed with a ! |
08:25.36 | brc_ | ~tell echo_bg about help |
08:25.42 | *** join/#asterisk libpcp (libpcp@210.16.20.5) |
08:25.45 | brc_ | okaya |
08:25.45 | libpcp | Hello everyone |
08:25.47 | echo_bg | aa got it |
08:25.50 | brc_ | who is running the rouge bot |
08:26.19 | libpcp | anyone got a problem with PTHREAD_MUTEX_RECURSIVE |
08:26.28 | brc_ | ~owner |
08:26.29 | jbot | TimRiker (or BZFlag) is my owner |
08:26.36 | brc_ | !owner |
08:27.08 | echo_bg | so if i have a phoneline I have to connect it to asterisk with fxo module - right ? |
08:27.10 | brc_ | Matthew_I, |
08:27.13 | brc_ | ~fxo |
08:27.14 | jbot | hmm... fxo is foreign exchange office - or the type of port you need to connect a POTS (Plain Old Telephone Service) line from your telco to a pbx |
08:32.32 | Jas_Williams | !help |
08:32.33 | WifiFred | hello, i am a bot. my home page is http://www.bcwireless.net/wififred |
08:32.34 | WifiFred | i listen for commands prefixed with a ! |
08:33.19 | echo_bg | ~cdr |
08:33.20 | jbot | i heard cdr is Compact Disc Recordable, see cdrw, or a copy & crunch set of programs for ripping cds, or in telecommunications, a call-detail-record |
08:34.25 | echo_bg | ~tell echo_bg about call-detail-record |
08:34.35 | echo_bg | ~tell echo_bg about cdr |
08:34.41 | wasim | ~no cdr is Call Detail Record, a listing of what happens to the call at each step through its traversal of the PBX, details like from, to, time, duration, number dialled etc, useful for billing also |
08:34.42 | jbot | okay, wasim |
08:35.05 | wasim | ~no cdr is Call Detail Record, a log of what happens to the call at each step through its traversal of the PBX, details like from, to, time, duration, number dialled etc, useful for billing also |
08:35.08 | jbot | okay, wasim |
08:35.08 | brc_ | wasim, fyi jbot is shared between here and #debian and others |
08:35.30 | echo_bg | is there any other billing method but cdr ? |
08:35.49 | wasim | ~no cdr is Call Detail Record, a log of what happens to the call at each step through its traversal of the PBX, details like from, to, time, duration, number dialled etc, useful for billing also - it could also be Compact Disc Recordable, see cdrw |
08:35.50 | jbot | okay, wasim |
08:45.42 | wasim | why do we share jbot with #debian? |
08:45.51 | brc_ | because |
08:45.56 | brc_ | I'm working on a python bot just for us |
08:46.07 | brc_ | it will be ready RealSoonNow(TM) |
08:46.28 | paulc | I haven't seen RealSoonNow(tm) for ages.. well, from someone else at least.. I'm sure I typed it last week.. |
08:51.35 | manipura | anyone know what it would take to get my USB phone to work with X-lite? |
08:53.35 | *** join/#asterisk dome_c (~dome@61.19.211.250) |
08:54.12 | dome_c | hi all. i have problem with sipfriend from database |
08:57.07 | BaDBuG | hey can anyone help me configure sip.conf? and extentions.conf? |
08:57.48 | BaDBuG | it will have 4 clients only |
08:57.52 | dome_c | what problem ? |
08:58.07 | BaDBuG | i am not really sure how to configure them ? |
08:58.22 | BaDBuG | i am not fimilliar with vi at all |
08:58.22 | dome_c | make samples |
08:58.33 | dome_c | it's wll genreate you default config |
08:58.53 | BaDBuG | yeah did that but this plus vi scares me lOl |
08:59.10 | BaDBuG | just not knowing where to start |
09:00.11 | clive- | badbug, just play with it...i for insert, then you can just edit |
09:00.39 | clive- | the esc, shift wq! to exit and save |
09:01.27 | brc_ | ! is bad |
09:01.32 | dome_c | What's best solution for 1000 user ? ast_data Or mysql default ? |
09:01.55 | dome_c | or default sip.conf text file ? |
09:01.59 | clive- | brx howcome? |
09:02.01 | clive- | brc |
09:02.14 | BaDBuG | i just need to have fwd configured for incoming calls and rings @ my extention in "U.S." with the ata-286. and my family have the pcphoneline usb |
09:02.16 | clive- | I get no points for all my typos:) |
09:02.30 | BaDBuG | and my bro will kall me with xten and my sis same |
09:02.52 | BaDBuG | and we all kall each other with extentions |
09:03.02 | BaDBuG | is it too bad? |
09:03.11 | BaDBuG | i mean lots of work? |
09:04.50 | paulc | badbug should be pretty easy |
09:04.57 | BaDBuG | wanna use the 729 codec. cause it faaaar away |
09:05.05 | BaDBuG | okay i will give it a shot |
09:05.06 | RoyK | argh |
09:05.10 | RoyK | if I use sipfriends |
09:05.16 | BaDBuG | I wish man i can finish with it now |
09:05.18 | RoyK | how can I tell * to use nat = yes? |
09:05.23 | BaDBuG | i have been reading soooo much ! |
09:05.42 | BaDBuG | trying to learn more bout linux |
09:06.13 | dome_c | in genreal config you can set nat=yes |
09:06.14 | BaDBuG | now i am stuck trying install slakeware on my 2X raptors in raid ICH5r |
09:06.20 | dome_c | royk |
09:06.47 | RoyK | dome_c, will that work with sipfriends from mysql? |
09:06.53 | RoyK | it doesn't look so |
09:07.00 | BaDBuG | okay i have the ipcop machine as a firewall will it be better if i added a third card and have the * in the orange interface? |
09:07.04 | dome_c | me too. |
09:07.19 | dome_c | I can enable NAT in general config |
09:07.30 | dome_c | but problem is sipaddr , port |
09:07.34 | RoyK | and it works with all sorts of Nated clients? |
09:07.38 | RoyK | rait |
09:08.05 | BaDBuG | roy you asking me? |
09:08.05 | dome_c | when i connect from softphone sip chan update mysql with wrong information |
09:08.35 | RoyK | I extract the userid from SIP/userid@.. |
09:08.56 | RoyK | and then set the callerid according to that (to allow for independant callerid/sip name) |
09:09.05 | dome_c | Now i'm looking for new solution may be retreive config from mysql to sip.conf |
09:09.17 | dome_c | Or ast_data from pgsql. i test it before. |
09:09.31 | RoyK | but after a while, NATed clients tell me "SIP/sip-server-host-name@... |
09:09.37 | RoyK | instead of SIP/username |
09:09.42 | dome_c | Uhm.. |
09:09.49 | dome_c | We got same problem. |
09:09.50 | RoyK | this is _only_ NATed clients |
09:09.52 | RoyK | ok? |
09:09.57 | RoyK | that's good, though |
09:10.20 | dome_c | I 'm working on it. i'll let's you know if i fixed it. |
09:11.28 | RoyK | dome_c, have you enabled SIP_USERS in chan_sip.c? |
09:11.41 | dome_c | No only MYSQL_FRIENDS |
09:11.54 | dome_c | Need to enable ? |
09:12.39 | RoyK | channels/chan_sip.c:61 |
09:12.40 | RoyK | #ifdef SIP_MYSQL_FRIENDS |
09:12.40 | RoyK | #define MYSQL_FRIENDS |
09:12.40 | RoyK | #define MYSQL_USERS |
09:12.40 | RoyK | #include <mysql/mysql.h> |
09:12.40 | RoyK | #endif |
09:13.01 | RoyK | I had that enabled but I'm not sure if it made it into the last (1.0.0) build |
09:13.02 | dome_c | I enable from Makefile |
09:13.09 | RoyK | whatever |
09:15.48 | *** join/#asterisk ckruetze (~ckruetze@host217-42-237-183.range217-42.btcentralplus.com) |
09:17.18 | echo_bg | guys - there is a link that is not working in voip-info-org |
09:17.30 | echo_bg | concerning xlite and asterisk sip.conf |
09:17.34 | echo_bg | I have the PDF |
09:17.43 | echo_bg | would you please add it to the page ? |
09:18.13 | maruz | which link? which page? |
09:18.17 | echo_bg | mo |
09:18.18 | echo_bg | mom |
09:18.44 | echo_bg | http://www.voip-info.org/www.astmasters.net/stuff/X-Lite-and-Asterisk.pdf |
09:20.59 | RoyK | dome_c, works for me (tm) |
09:21.35 | echo_bg | maruz - got it ? |
09:22.14 | ckruetze | echo_bg there is a link o that pdf on http://www.voip-info.org/tiki-index.php?page=xten and it works |
09:22.19 | maruz | ok |
09:24.04 | maruz | echo_bg: this does work: Getting started with X-Lite: A step by step guide (PDF) |
09:24.28 | dome_c | Royk.. i'm patching ast_data |
09:26.20 | echo_bg | ckruetze - thx |
09:26.59 | *** join/#asterisk vs_ (vs@univac.spamcheck.net) |
09:27.02 | vs_ | morning |
09:27.27 | vs_ | asterisk -r aint able to STOP NOW |
09:27.52 | vs_ | Asterisk CVS-HEAD-09/28/04-05:06:49, |
09:30.52 | BaDBuG | sorry guyz for all the question s but what do you recomned for changing sip.cof and all from the web |
09:31.03 | vs_ | if remote started with -fc it does work |
09:31.05 | BaDBuG | i took a look at wiki |
09:31.12 | vs_ | even w/o -f |
09:31.13 | BaDBuG | but you guyz will know better |
09:31.25 | BaDBuG | even if there is any GUI |
09:33.40 | RoyK | dome_c, ast_data??? |
09:34.46 | dome_c | http://svn.asteriskdocs.org/res_data/ |
09:40.59 | dome_c | RoyK. it's work :) |
09:41.12 | ScaredyCat | destar is a GUI |
09:43.43 | wasim | and ScaredyCat is a FELINE |
09:44.20 | wasim | Princess SmellyCat Bananhammock |
09:44.33 | maruz | there is no link to Obelisk, or i'm blind... |
09:44.42 | RoyK | dome_c, what does it do_ the READMEs don't tell |
09:45.53 | dome_c | README is Script for patch |
09:46.12 | dome_c | Give me email i'll send you asterisk-1.0.0+astdaya |
09:46.25 | dome_c | I patch already you can compile. |
09:46.34 | dome_c | but need postgresql-devel , mysql-devel |
09:49.04 | maruz | dome_c: what have you patch in ast_data? |
09:50.01 | dome_c | new sip-firend and iax-freind table |
09:50.07 | dome_c | structure |
09:51.48 | maruz | do u prefer send it by email than to give us an url where to download it? |
09:52.33 | dome_c | ftp://203.146.102.2/asterisk/ |
09:52.59 | maruz | i dive into it :) |
09:53.01 | dome_c | very happy with ast_data.. :) |
09:53.25 | dome_c | because i plan to integreate asterisk to my billing for long time... |
09:57.01 | dome_c | have someone service DID in Japan and Sigapore here ? |
10:00.50 | manipura | anyone know how do can make asterisk start when the computer boots? |
10:01.04 | dome_c | what's linux distro ? |
10:01.10 | dome_c | redhat or debian ? |
10:01.13 | manipura | RH9 |
10:01.31 | dome_c | create /etc/init.d/asterisk |
10:01.59 | manipura | and put what in it? |
10:03.23 | dome_c | ftp://203.146.102.2/asterisk/init.d/ |
10:03.41 | dome_c | copy asteriskd start script |
10:03.50 | dome_c | and then chkconf --add asteriskd |
10:04.15 | dnc | any ideas about this message: |
10:04.16 | dnc | Sep 28 18:10:16 NOTICE[245775]: chan_zap.c:7358 pri_dchannel: PRI got event: 8 on Primary D-channel of span 2 |
10:04.16 | dnc | Sep 28 18:10:16 NOTICE[245775]: chan_zap.c:7358 pri_dchannel: PRI got event: 6 on Primary D-channel of span 2 |
10:04.16 | dnc | Sep 28 18:10:16 NOTICE[229390]: chan_zap.c:7358 pri_dchannel: PRI got event: 6 on Primary D-channel of span 1 |
10:04.17 | dnc | Sep 28 18:10:16 NOTICE[229390]: chan_zap.c:7358 pri_dchannel: PRI got event: 8 on Primary D-channel of span 1 |
10:04.37 | ard | dnc : zaphfc? |
10:04.50 | manipura | Awesome, thanks dome_c |
10:05.36 | dnc | hmmm i think it was a ccs vs cas problem |
10:05.46 | dnc | just checking the configs again |
10:06.59 | dome_c | becarefull manipura it's virus he he |
10:07.01 | dome_c | :) |
10:07.53 | echo_bg | dome_c what is virus ? |
10:08.04 | ScaredyCat | dnc: check you aren't getting irq misses |
10:08.10 | dome_c | sorry it's joke. |
10:08.17 | dome_c | <dome_c> ftp://203.146.102.2/asterisk/init.d/ |
10:08.17 | dnc | hey ScaredyCat :) |
10:08.24 | ScaredyCat | lo dnc |
10:09.59 | ScaredyCat | I bet it's a 1u box isn't it dnc |
10:10.25 | dnc | err actually its a tower case |
10:10.32 | ScaredyCat | oh.. |
10:10.59 | ScaredyCat | try accessing the disk while in a call... see if that causes the messages to appear |
10:11.12 | Jas_Williams | dnc: Looks like a config issue, is this euroISDN or T1 ? |
10:11.21 | ScaredyCat | seen it b4, irq misses caused by card shared with disk |
10:11.21 | dnc | E1 but not euro isdn |
10:11.23 | ScaredyCat | irq |
10:11.27 | dnc | i think it was the ccs, cas thing that did it |
10:11.49 | dnc | its ok now ive changed over to cas |
10:11.56 | ScaredyCat | hmmm... |
10:11.58 | dnc | just making some test calls now to be sure |
10:12.27 | Jas_Williams | dnc: As far as I know we do not support CAS over E1 currently so span needs to be css not cas |
10:12.37 | Jas_Williams | oops ccs |
10:12.48 | dnc | really? |
10:12.51 | dnc | hmmm |
10:12.57 | dnc | well lets see how the test calls go |
10:13.29 | *** join/#asterisk lseror (~lseror@217.174.204.42) |
10:13.47 | lseror | Hi |
10:14.01 | lseror | Anyone know if we can filter incoming call with asterisk ? |
10:15.26 | Jas_Williams | lseror: Yes what do you need to do ? |
10:17.48 | manipura | dome_c that script didn't work, is there something I'm missing? |
10:18.05 | dome_c | what's error ? |
10:18.35 | dome_c | [ -x /usr/sbin/asterisk ] || exit 0 |
10:18.35 | dome_c | [ -d /etc/asterisk ] || exit 0 |
10:19.00 | dome_c | Is you asterisk install in /usr/sbin ? |
10:19.02 | manipura | no error, it just wasn't started when I rebooted |
10:19.20 | dome_c | can you test /etc/init.d/asteriskd |
10:19.31 | dome_c | by manual first |
10:19.35 | manipura | k, gimme a sec, its rebooting |
10:19.41 | dome_c | k |
10:20.04 | *** join/#asterisk Ngy_Dave (~davy@213.219.141.105.adslpower.by.edpnet.be) |
10:21.18 | manipura | yeah, I can |
10:21.38 | dome_c | add to default by |
10:21.58 | dome_c | chkconfig --add asteriskd |
10:22.19 | dome_c | Or setup -> system service |
10:23.21 | manipura | Yahoo, its 4:20AM |
10:25.28 | RoyK | ~seen cy |
10:25.29 | jbot | cy <~darknovae@63.225.225.227> was last seen on IRC in channel #utah, 115d 17h 29m 20s ago, saying: 'night all'. |
10:25.30 | RoyK | ~seen cypromis |
10:25.30 | jbot | cypromis is currently on #asterisk. Has said a total of 43 messages. Is idling for 4h 35m 59s |
10:27.09 | manipura | awesome, got it working, thanks dome_c, I'll remember that chkconfig command for sure! |
10:27.37 | dome_c | :) |
10:28.46 | lseror | Jas_Williams : I want to blacklist someone who is sending a fax every hour to my phone :) |
10:29.39 | Jas_Williams | lseror: Based on cli you can send call to busy or other location |
10:30.26 | lseror | Jas_Williams: what document should I read ? |
10:32.04 | *** join/#asterisk inspired (mikael@a217-118-63-4.bluecom.no) |
10:34.44 | *** join/#asterisk glLoadIdentity (~tuyan@dsl81-214-9283.adsl.ttnet.net.tr) |
10:37.30 | lseror | In fact what's i'm looking for is some kind of PBX Firewall, to filter incoming call by example |
10:39.17 | salimfadhley | Anybody know of some high-quality asterisk logo artwork available? |
10:43.14 | wasim | mamboserver.com |
10:43.16 | wasim | :P |
10:44.08 | salimfadhley | mambo... :-( |
10:44.11 | salimfadhley | zope :-) |
10:44.40 | wasim | yeah, but mamboserver logo passes for a neat * logo |
10:45.07 | salimfadhley | oh... yes... well I kind of like the real one. I just want to find something thats nice enough to make a 16x16 asterisk icon in my zope application. |
10:45.25 | salimfadhley | still... it might fool the masses |
10:45.26 | salimfadhley | :-) |
10:47.01 | file | I'm so comfy in bed, I don't want to go to school |
10:48.45 | *** join/#asterisk sob0l (~devil@80.51.169.19) |
10:50.33 | manipura | file, you have a computer in your bed? |
10:50.35 | manipura | I want one |
10:50.55 | nextime | manipura : laptop&wifi :) |
10:51.29 | file | nextime: indeed |
10:51.40 | manipura | file, send me yours |
10:51.42 | ScaredyCat | <PROTECTED> |
10:51.43 | file | and now I must depart, thankfully I have a substitute for english... so yah! |
10:51.46 | file | manipura: ha |
10:51.53 | echo_bg | guys - gotta go - |
10:51.54 | manipura | file, hope she's hot! |
10:51.56 | echo_bg | c ya soon |
10:52.04 | *** part/#asterisk echo_bg (~asdasdasd@212.116.151.30) |
10:52.10 | Jas_Williams | lseror: Can you post the relevant section from your extensions.conf for the incoming fax call to pastebin.ca and then I could come up with some sort of blocking config |
10:53.22 | *** join/#asterisk sambal (~sambal@gateway.office.flatbox.nl) |
10:53.26 | PETER22 | quick q. - if i have an incoming zap pstn call, can i make it ring on combination of zap fxo and sip channels ? |
10:55.16 | *** join/#asterisk postel (~postel@host217-42-116-153.range217-42.btcentralplus.com) |
10:55.47 | Jas_Williams | PETER22: Yes exten => s,1,Dial(SIP/1234&Zap/1) but the zap channel would always except the call unless you are using ISDN/T1/E1 |
10:56.30 | Jas_Williams | PETER22: An fxo Zap call is considered answered as soon as dialled. |
10:56.42 | PETER22 | thanks jas_williams |
10:57.27 | PETER22 | i think i have to make it ring sip first, then after timeout go dialing zap |
10:58.05 | Jas_Williams | PETER22: That will work. |
10:58.53 | lseror | Jas_Williams : What is pastebin.ca ? |
10:59.19 | lseror | Jas_Williams : I found something about Call Blocking on Caller ID |
10:59.25 | ard | pastebin |
10:59.35 | PETER22 | anyway to keep orginal persons caller id on a call accepted by *, then forwarded out zap fxo ? |
10:59.36 | lseror | Jas_Williams : It says that I should use the dialplan |
10:59.40 | clive- | iseror look at ex girlfreind logic |
10:59.42 | ard | jbot : what is pastebin |
10:59.47 | jbot | from memory, pastebin is a place to paste all your conf/debugs/logs for other people in the chatroom to view without flooding the channel. We suggest http://pastebin.ca |
11:00.43 | lseror | ok I found pastebin |
11:03.51 | Jas_Williams | PETER22: No you cannot maintain CalerID the caller ID displayed will be that of the fxo. |
11:04.06 | *** part/#asterisk dome_c (~dome@61.19.211.250) |
11:04.09 | manipura | jbot : what is a dead hooker |
11:04.10 | jbot | I think you lost me on that one, manipura |
11:05.14 | manipura | is there a list of what jbot knows? |
11:05.24 | lseror | Jas_Williams, I pastebin it |
11:05.52 | manipura | Iseror, http://pastebin.ca/1147 |
11:07.01 | manipura | jbot what is asterisk |
11:07.02 | jbot | a PBX (Private Brance eXchange) and telephony toolkit. http://www.asterisk.org, or very cool, or awesome |
11:07.18 | lseror | yes this is my post, 1147 |
11:08.22 | manipura | Hey EVYERONE http://pastebin.ca/1146 |
11:08.24 | Jas_Williams | lseror: How are the calls delivered to * |
11:10.35 | PETER22 | i have heard ISDN can bounce incoming calls to a diversion without answering ? |
11:10.38 | lseror | Jas_Williams : They are directly send to Zap |
11:11.47 | PatrickDK | what username are you using in sip.conf? |
11:11.50 | PETER22 | I have a main business pstn line - i use my carrier "divert all" feature and i get the original persons caller id (important) |
11:12.42 | PETER22 | i would like these calls to be handled by * in some way (if i'm in the office) |
11:17.54 | lseror | Jas_Williams : it seems that there is a blacklist database implemented in Asterisk |
11:18.34 | _dw | jbot_: forget asterisk |
11:18.54 | _dw | jbot_: asterisk is a PBX (Private Branch eXchange) and telephony toolkit. http://www.asterisk.org, or very cool, or awesome |
11:19.00 | _dw | jbot_: what is asterisk? |
11:19.19 | _dw | if they are not the correct commands, can someone fix the spelling mistake. its bloody shameful |
11:19.27 | _dw | jbot what is asterisk |
11:19.28 | jbot | a PBX (Private Brance eXchange) and telephony toolkit. http://www.asterisk.org, or very cool, or awesome |
11:19.42 | manipura | jbot can you die? |
11:19.49 | _dw | jbot asterisk is a PBX (Private Branch eXchange) and telephony toolkit. http://www.asterisk.org, or very cool, or awesome |
11:19.50 | jbot | ...but asterisk is already something else... |
11:19.56 | _dw | jbot forget asterisk |
11:19.56 | jbot | i forgot asterisk, _dw |
11:20.00 | _dw | jbot asterisk is a PBX (Private Branch eXchange) and telephony toolkit. http://www.asterisk.org |
11:20.01 | jbot | okay, _dw |
11:20.06 | _dw | jbot what is asterisk? |
11:20.09 | jbot | well, asterisk is a PBX (Private Branch eXchange) and telephony toolkit. http://www.asterisk.org |
11:20.09 | manipura | jbot what is google |
11:20.11 | jbot | rumour has it, google is a search engine found at http://www.google.com/ |
11:20.29 | manipura | jbot what is jbot |
11:20.31 | jbot | jbot is, like, ibot's stupid cousin |
11:22.17 | *** join/#asterisk ckruetze (~ckruetze@host217-42-237-183.range217-42.btcentralplus.com) |
11:25.37 | BuzzBud | Any reason not to noload => chan_skinny.so ? |
11:29.22 | lseror | OK i think I find how to do it |
11:29.23 | *** join/#asterisk sid (~sid@gethsemane.odgers.id.au) |
11:29.26 | sid | lo guys |
11:29.27 | lseror | it is working |
11:29.39 | lseror | brb |
11:30.08 | sid | is there a way you can tell asterisk to provide dialtone to the caller when a particular extension is called? |
11:30.14 | sid | like a redialler? |
11:30.45 | sid | i've got iaxtel's free 1-800 stuff set up on my asterisk box, and want to give a couple people access thereto via pots |
11:31.07 | miller7 | sid: I think that there is, let me check my notes, I have this from a long time now |
11:31.17 | sid | miller7: ta :) |
11:36.06 | *** join/#asterisk dabba (~d@matrix.lgw.ip6net.net) |
11:40.17 | manipura | ~seen dabba |
11:41.34 | jbot | dabba is currently on #asterisk (5m 28s) |
11:41.34 | sid | 'lo manipura :) |
11:41.59 | manipura | i've always wanted my own robot |
11:44.32 | ckruetze | dabba, here in Cambridge it isn't cold, but a hooverbot would be cool |
11:45.09 | dabba | $249 and it just wandered around hoovering, my front room is too small though :-( |
11:45.10 | sid | it's cold here in melbourne. |
11:45.11 | sid | very very cold. |
11:46.48 | manipura | We are getting great weather in calgary |
11:47.01 | ckruetze | sid, then move up north |
11:47.34 | sid | nah, no work up that way |
11:47.58 | Dibbler_ | Cold in Melbourne, that sounds wierd to us Brits, who generally always think of Oz as being sunny and warm |
11:50.57 | sid | about 12 and torrential today :/ |
11:51.42 | *** join/#asterisk fool (~dwinter@67.106.194.90.ptr.us.xo.net) |
11:52.25 | Dibbler_ | 12 isn't cold, hehe |
11:53.03 | lseror | Anyone want to know how you blacklist a caller-id ? |
11:53.24 | sid | set up an extension in the correct context which specifies it, pointing somewhere dumb? |
12:00.55 | *** join/#asterisk autobus (~autobus@80.172.12.87) |
12:02.05 | *** join/#asterisk |Vulture| (~Vulture@247.131.vbnet.net) |
12:03.05 | autobus | as it is that I can copy the filing-cabinet automatically a filing-cabinet of /var/spool/asterisk/callback.call for = "/var/spool/asterisk/outgoing/ |
12:03.49 | autobus | when to cal for a extension |
12:04.27 | autobus | ? |