irclog2html for #asterisk on 20060718

00:00.04carl0s-told you I was poor at Geography :)
00:01.51andrejkwSo, is anyone here nice enough to help a poor guy that can't get incoming calls to work? :)
00:03.40mattfletchercarl0s: can you try ringing me again? 01524888922
00:03.46carl0s-1 sec
00:04.00carl0s-ringing :D
00:04.04mattfletcheryay
00:04.16mattfletcheri think i need to tell it where to go internally now
00:04.20mattfletcherbrb
00:04.21carl0s-I'll hang up. save the bill :)
00:04.33carl0s-yep.. I think you will.
00:04.54mattfletchersorry, i assumed it would be free, i will try myself again now
00:05.04carl0s-it's not a problem :D
00:05.54mattfletcherwrong context in sipgate's example i think
00:05.59mattfletchernearly there...
00:06.25carl0s-I didn't take the extensions bit from sipgates example because I'm using trixbox.. I just took the inbound/outboud sip trunk settings from there.#
00:06.57morexG'night all
00:08.47asterisk-dudcan anyone point me in the right direction for enabling call forwarding in asterisk
00:08.51mattfletcherWAHOO!!!!
00:08.59asterisk-dudlike a weblink or someting
00:09.21andrejkwmattfletcher: lucky :'(
00:09.34mattfletchersorry andrejkw
00:09.42mattfletcherthank you so much to everyone who helped me
00:09.57mattfletcheryet another reason to spread the linux love
00:10.15andrejkwaww mam :'(
00:10.16mattfletchernet job - convert the windows xp media center 2005 box to mythtv
00:10.20andrejkwmy still won't ring
00:10.40carl0s-:D
00:10.47carl0s-working then? groovy :)
00:10.55andrejkwanyone willing to help me? :'(
00:11.28mattfletcheri will try andrejkw, but as you have seen i am a newbie myself
00:11.38mattfletcherwho are you with?
00:11.38MoutaPTandrejkw what is your problem?
00:11.45andrejkwVoiceStick
00:11.56andrejkwI can make calls, but not recevie calls with Asterisk.
00:11.58pdthomeandrejkw: just ask your question, usually if somebody can help they will
00:12.13andrejkwIf I use the information directly in my PAP2, it works fine (call & receive).
00:12.14mattfletcherdo they say they support asterisk? any config files examples etc?
00:12.29MoutaPTcalls from FXO ?
00:12.31mattfletcher"the information"... what do they provide?
00:12.59andrejkwThere is a few people online that got it to work (they proveded their configuration), but I still can't get it to work.
00:13.11andrejkwThey provide the SIP proxy and the Outbound Proxy.
00:13.18*** join/#asterisk rene- (n=rene-@dsl-200-67-175-250.prod-empresarial.com.mx)
00:13.20*** join/#asterisk AJaymn (i=AJmn@70.59.126.197)
00:13.33rene-when a channel is hangup can it be said that it was destroyed?
00:14.04andrejkwBut like I said, if I use the SIP proxy and the Outbound proxy directly in my Phone Adapter it works just fine.
00:14.30mattfletcheryou said you can make calls, correct?
00:14.34andrejkwYes.
00:14.41mattfletcherwhat happens when you dial in?
00:14.58andrejkwI hear the ringing tone for a while and then the service voice mail comes up.
00:15.09andrejkwBut while I hear ringing, the actual phone isn't.
00:15.21pdthomeit could be a natting issue
00:15.28mattfletcheri found out a lot by following the debugging on the console:
00:15.35mattfletcherasterisk -rvvvvvvvv
00:15.42andrejkwI tried nat=yes, but it makes no difference.
00:15.57mattfletcherdo you see anything coming in on the console?
00:16.31andrejkwYes, when I call out stuff shows up. But when I call in, it's dead.
00:16.58andrejkwI tried doing "sip debug", and I actually see something showing up when I call in. Bunch of weird stuff.
00:17.03*** join/#asterisk ivanfm_ (n=ivanfm@201.52.162.52)
00:17.03pdthomeandrejkw: http://www.voip-info.org/wiki-Asterisk+SIP+NAT+solutions
00:17.15*** join/#asterisk postel_ (n=jp@unaffiliated/postel)
00:17.16andrejkwThat says "Destroying call" all the time.
00:17.49andrejkwI have #4.
00:17.54mattfletcherthat surely suggests that the calls are getting through then. can you try that command i put: asterisk -rvvvvvvvv
00:18.13andrejkwI did.
00:18.21andrejkwStill nothing, it's dead when I call in.
00:18.28mattfletcherjust now, my calls were coming in, but my routing wasn't set up to make any extensions ring
00:18.40pdthome~pb
00:18.43jbotsomebody said pb was a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/
00:18.53pdthomeandrejkw: send a link to the console dump
00:19.01mattfletcher[inbound]
00:19.01mattfletcherexten => 4888922,1,Answer
00:19.01mattfletcherexten => 4888922,2,Dial(USTM/2002@2002)
00:19.01*** join/#asterisk tessier_ (n=treed@gw.drjays.com)
00:19.04tessier_Hello all!
00:19.15andrejkwpdthome: what do you mean?
00:19.20mattfletcherthat is how my inbound routing works
00:19.31pdthomethe wierd stuff, go to one of those links above and paste it in
00:19.36andrejkwok
00:19.37pdthomeit will give you a link you can paste in here
00:19.56tessier_Soon I will be implementing another asterisk based phone system. I need a PRI interface and I am not going to get burned on digiums stuff again. What would most people recommend? Some level of Cisco with a PRI module?
00:20.04mattfletcherUSTM as I am using a silly nortel proprietary phone i've "borrowed" from work
00:20.52andrejkwhttp://pastebin.ca/90741
00:21.03andrejkwFor some reason that stuff also shows up randomly.
00:21.13andrejkwEven when I don't call in, so I am not sure anymore.
00:22.01pdthomeandrejkw: this is a copy of what you got when you tried calling in?
00:22.19andrejkwYes, but even after I hang up, it just keeps showing up and flooding the console.
00:22.30andrejkwLike one every 1 - 5 minutes.
00:23.46andrejkwI am not sure if this has anything to do with the call.
00:23.56andrejkwBut I am not sure, something tells me it doesn't.
00:24.01*** join/#asterisk ariel_ (n=Ariel@dsl-20-177.cofs.net)
00:24.04mattfletcheri'm out - means nothing to me. no point misguiding you
00:24.38andrejkwIt just looks like the calls are not even reaching Asterisk.
00:24.39andrejkwAt all.
00:25.26andrejkwI've been sitting in front of this all day long, and I just can;t figure it out.
00:25.27harryvvany zoneminder users here
00:25.46pdthomeandrejkw: so if you turn off sip debug and make a call in you get no console messages?
00:25.55andrejkwNope.
00:25.58andrejkwAbsolutely nothing.
00:26.28hadsandrejkw: Did you read the book yet?
00:26.45mattfletcherwhat is your external ip, have you tried port scanning yourself to see if anything can get in?
00:26.58andrejkwNo, I didn't.
00:27.08hads~thebook
00:27.10jbot[thebook] a book called  Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
00:27.10andrejkw<PROTECTED>
00:27.12andrejkwoops
00:27.15andrejkw71.57.143.216
00:27.18mattfletcheri can for you if you have no machine outside to do it from
00:27.28hadsDo yourself a favour and read the book.
00:27.35andrejkwIt would be nice if you could.
00:27.59andrejkwAnd how is the book going to help me in this situation?
00:28.12andrejkwWhen I am having problems getting the SIP provider to work properly.
00:28.22*** join/#asterisk P-NuT (n=P-NuT@CPE-60-227-93-75.nsw.bigpond.net.au)
00:28.24hadsErm... to understand Asterisk
00:29.17MoutaPTandrejkw: do you have an incoming context for your SIP user?
00:29.26carl0s-Well I just tried to setup my sipgate.co.uk account again, and I think I'm mostly there. The calls are being answered by Asterisk, but Asterisk is going straight to "the number you have dialled is not in service". However my only inbound route is "any CID / any DID" so I don't know why that is.
00:29.50andrejkwMoutaPT: yes
00:30.05MoutaPTsip show registry show your provider?
00:30.26pdthomecarl0s-: are you sure it's going into the context you expect?
00:30.34andrejkwNope
00:30.42andrejkwit doesn't
00:30.46P-NuTHi all. Does anyone use cisco 7905G phones here? and has a config for one including different ringtones?
00:31.15MoutaPTandrejkw do you have the register=> user:pass@sip.provider.XXX in your sip.conf?
00:31.42andrejkwMoutaPT: nope
00:31.49andrejkwMoutaPT: Does it go under general?
00:32.05MoutaPTmy * is unaccessable now, so i can't check it
00:32.20MoutaPTlook over the wiki for voipbuster examples
00:32.24AJaymnAnyone use Shellshark VoIP?
00:32.28MoutaPTvoipbuster is sip provider
00:32.33MoutaPTworks fine with asterisk
00:32.37MoutaPTat least withe me
00:32.51MoutaPTfirst you should get registred
00:33.15carl0s-pdthome: I have no idea what a blooming context is. I really am going to print off that book and start reading it in bed. I'm cheating so far. Probably not even worth it.
00:33.24MoutaPTthen you may think about your dialplan! I must say, as far as i know, not all cases registry is compulsory
00:33.33MoutaPTbut for newbie i would recomend
00:33.34andrejkwOk.
00:33.52andrejkwI put it in sip.conf and now it shows up.
00:34.01andrejkwBut I still get no calls.
00:34.10MoutaPThttp://www.voip-info.org/wiki/view/Asterisk+VoIPBuster
00:34.24MoutaPTcan you outbound calls?
00:34.24andrejkwI don't have Voipbuster.
00:34.26andrejkwo.O
00:34.33andrejkwFor incoming.
00:34.37MoutaPTcan you outbound calls?
00:34.42andrejkwFor incoming I have voipstick.
00:34.47carl0s-ah. my context should be from-trunk. I remember now. I followed a guide last time.
00:34.49andrejkwAnd yes, I can do outbound out of both.
00:34.58andrejkwVoipbuster and voipstick.
00:35.04MoutaPTdo you have any router?
00:35.09andrejkwYes
00:35.15andrejkwThat is weird, I am getting sip_reg_timeout:
00:35.17MoutaPTdoes your ASterisk runs in  DMZ ?
00:35.23carl0s-yup. that fixed it.
00:35.46AJaymnWhats a VoIP provider that offers CID spoofing?
00:35.55MoutaPTdo you have correctly port forward 5060 for your AStBox?
00:36.22andrejkwHmm, no I don't actually. Let me forward it.
00:37.12MoutaPTthat is a MUST DO
00:38.01andrejkwOk I did.
00:39.22MoutaPTtest it now
00:39.27andrejkwnope
00:39.29andrejkwdoesn't ring still
00:39.30MoutaPTlook your asterisk CLI
00:39.40MoutaPTsip debug enable
00:40.06andrejkwJan  1 04:49:31 NOTICE[3747]: chan_sip.c:5267 sip_reg_timeout:    -- Registration for '1305831xxxx@i2telecom.com' timed out, trying again (Attempt #7)
00:40.09*** part/#asterisk P-NuT (n=P-NuT@CPE-60-227-93-75.nsw.bigpond.net.au)
00:40.11andrejkwI keep getting that
00:40.26*** join/#asterisk enjay- (i=enjay-@wsip-24-249-169-168.ph.ph.cox.net)
00:40.37MoutaPTit seems you don't have correct config in sip.conf
00:40.53MoutaPTdo you have anny sipphone from this provider?
00:41.02MoutaPTjust to try to register there
00:42.16MoutaPTyou  may need to put settings like qualify=yes
00:42.24MoutaPTinsecure=very
00:42.26MoutaPT...
00:42.30*** join/#asterisk anonymouz666 (n=anonymou@20151155235.user.veloxzone.com.br)
00:42.49MoutaPTit's better to check it with your sip provider or @ wiki
00:44.13MoutaPTandrejkw did you port forward UDP on 5060 or TCP ?
00:44.24andrejkwI did both.
00:44.27MoutaPTok
00:44.31*** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com)
00:45.12MoutaPTyou r not getting registry... until there nothing will work i think....
00:45.20riddleboxwhat could I have done wrong if asterisk doesnt always give me ringback when I call someone?
00:45.24MoutaPTi must go, sorry i didn't solve your problem
00:45.32andrejkwthank you anyway
00:48.41*** join/#asterisk sugardave (n=not@cpe-66-68-164-115.austin.res.rr.com)
00:51.17*** join/#asterisk unit (n=doom@Toronto-HSE-ppp3780161.sympatico.ca)
00:53.08*** join/#asterisk Mr-packet (n=a@222-154-239-122.adsl.xtra.co.nz)
00:54.31Mr-packetby default does asterisk bind itself to all ports on a machine when it starts, or is that configured somewhere.
01:01.06Sponge_bobanyone use a 7970 with asterisk?
01:01.31Mr-packetSponge. not personally, but i've seen it done
01:02.35Sponge_bobMr-packet: i"m curious to hear how it scales with asterisk
01:02.46Mr-packetyour talking about a Cisco 7970?
01:02.51Sponge_bobMr-packet: what is that H.264? somthing like that?
01:03.02knarflyI have an extension to just play moh...works great locally. but when I connect from my office it plays one songe then hangs up. conference room with moh will play all day can anyone tell me about this?
01:04.30*** join/#asterisk Samoied (n=Samoied@ip47092.static.poa.virtua.com.br)
01:04.57*** join/#asterisk clyrrad (n=ddd@CPE001195f553c7-CM0011aea484a4.cpe.net.cable.rogers.com)
01:05.16*** join/#asterisk anthonyl (n=urmom@office.midphase.com)
01:06.27Sponge_bobknarfly: what's your qualify set to?
01:06.48andrejkwHow can I force the expire per SIP provider?
01:07.49knarflyyes
01:08.24clyrradI am having a rather strange issue - for some reason when Asterisk is writing voice mail its writing to a tmp folder - what I mean is instead of writing to /var/spool/asterisk/voicemail/[context]/[extension] it is writing to /var/spool/asterisk/voicemail/[context]/[extension]/tmp - this is fine except the MWI does not work on the phone - howerver if you dial into Comedian mail it tells you that you have messages - an
01:08.27knarflylet me double check but I'm sure it's "yes"
01:09.26andrejkwHow can I make the expirey different for each SIP provider?
01:09.35knarflySponge_bob: qualify=yes
01:09.44andrejkwAll I see is defaultexpirey that goes under general, is there even a way?
01:10.14clyrradandrejkw - you can also put that in each context which will override whats in general
01:10.23andrejkwOh ok :D
01:10.26andrejkwThanks then
01:10.31clyrradNP
01:10.44clyrradNow - Can anyone help me? :)
01:11.58*** join/#asterisk eDIsonxl (n=xian-lia@mail.artdio.com.tw)
01:12.02*** join/#asterisk jero (n=jero@modemcable235.87-82-70.mc.videotron.ca)
01:12.49andrejkwHey guys, I get "Jan  1 05:22:12 NOTICE[28529]: chan_sip.c:3588 process_sdp: No compatible codec!".
01:13.00Juggie#1, fix your system date.
01:13.01andrejkwHow do I figure out what codec, and how do I fix it?
01:13.22Juggie#2, make sure you allow a proper codec path between your clients
01:13.50hads#0 read the book
01:14.07clyrradJuggie or hads - do you know the answer to my voicemail question?
01:14.48Mr-packethow can i make asterisk bind to all interfaces when it starts up
01:14.58anthonyl0.0.0.0
01:15.04clyrradMr-packet set to 0.0.0.0
01:15.17Mr-packetwhere do i configure that?
01:16.00hadsclyrrad: I don't know off the top of my head, what type of phone are you using? SIP/ZAP etc.
01:16.43*** join/#asterisk andrejkw (n=andrejkw@c-71-57-143-216.hsd1.fl.comcast.net)
01:16.56Mr-packetclyrrad.. is that in the [general] section of manager.conf?
01:17.02clyrradhads - the phones are all SIP phones and they work (Tested on another server) - just the MWI light is not comming on and i cant figure out why
01:17.12riddleboxdoes anyone have a sipura spa-2100 working with both lines?
01:17.38hadsclyrrad: Do you have a mailbox=blah in sip.conf for each friend>
01:17.58Sponge_bobclyrrad: under your sip configuration does regexten match your mailbox?
01:18.52clyrradhads - yes i have it set to [extension]@[context]
01:19.25clyrradSponge_bob - i do not have a regextension= parameter set - do i need that?
01:19.40hadsis context other than default?
01:19.51clyrradhads - yes
01:19.53*** join/#asterisk nortex (n=barracud@adsl-69-149-173-94.dsl.amrltx.swbell.net)
01:20.00clyrradit is NOT 'default'
01:20.26hadsHave you tried, just as a test, if it will work with a mailbox in default?
01:20.29clyrradthe context is the the client id which is the phone number
01:20.34Sponge_bobclyrrad: try it.  otherwise are you using macros?
01:20.43clyrradyes I am using Macros
01:20.59*** join/#asterisk xxoxx (n=xxoxx@tor/regular/xxoxx)
01:21.07Sponge_bobis it happening for all phones?
01:21.11hads"the context is the the client id which is the phone number" - you lost me.
01:21.11Sponge_bobcheck to see if are macro syntax is correct.
01:21.23clyrradhads - i have not tried it under default as the macro tells it to write to the specific context
01:21.41clyrradhads - I have the context names in voicemail.conf as the phone number
01:21.51clyrradwith the respective extensions below
01:22.12hadsOK, interesting.
01:22.15clyrradSponge_bob - the voice mail is getting written to the right spot becase if you dial into Comedian Mail you can check and listen to the messages
01:22.21clyrradjust the MWI light is not comming on
01:23.01Sponge_bobclyrrad: correct. i had this problem before. it was my macro that was not right
01:23.06Mr-packetwhen i 'debug sip' to see whats happening ( why my calls are not answered ), i see my from says " From: "Jenny" <sip:6444602512@192.168.99.2>;tag=as78a1322f ". the device at the other end has no idea where 192.168.99.1 is, and is expecting the call from the address of the vtund..  how can i get asterisk to set the from, to the address of the vtund interface?
01:23.37Sponge_bobactually not the macro but the exten to call the macro
01:23.41clyrradSponge_bob - interesting - do you recall what you were doing wrong in your Macro?
01:23.57clyrradand your voice mail too was able to be checkd with Comedian Mail?
01:24.00clyrradjust now MWI light?
01:24.05clyrradno*
01:24.20Sponge_bobno light and no indication
01:24.26KryojenikAnyone know where I can get an Israeli DID?
01:24.41Sponge_bobit was my syntax that was wrong.  basically i copied another line and forgot to change some things
01:24.43*** part/#asterisk Samoied (n=Samoied@ip47092.static.poa.virtua.com.br)
01:24.59Mr-packetKryo.. the last one got blown up yesterday
01:25.00Sponge_bobcheck the values you pass the marco
01:25.09Kryojeniklol...
01:25.35clyrradSponge_bob - ok checking
01:26.03carl0s-I just had a bad thought.
01:26.04*** join/#asterisk andrejkw (n=andrejkw@c-71-57-143-216.hsd1.fl.comcast.net)
01:26.27Sponge_bobcarl0s-: no i will not rub you there. :-)
01:26.59carl0s-When I get my GSM <-> SIP gateway, and set it up with a spare SIM card to have my incoming sipgate.co.uk calls forwarded over to my mobile.. well.. I'm going to lose Caller-ID functionality aren't I :( Every call will show up as being from the other mobile :( that sucks.
01:27.03carl0s-Sponge_bob: lol :)
01:27.10clyrradexten => s,11,Voicemail(u${ARG1}@${ACCOUNTCODE})
01:27.14clyrradthat is my syntax
01:28.07Sponge_bobhum...
01:28.19Sponge_bobclyrrad: can you paste the rest to pastebin?
01:28.40clyrradthe entire macro?
01:29.08Sponge_bobclyrrad: does the problem exist for all sip phones?
01:29.22clyrradyes on all phones
01:29.38Sponge_bobhave you tried the regexten?
01:29.44clyrradnope
01:29.49clyrradhow would i set that?
01:29.50Sponge_bobtry it first
01:29.56clyrradto just the extension?
01:30.00clyrrador extension@context?
01:30.21Sponge_bob*i think* regexten=what ever exten the sip is
01:30.35Sponge_bobthat goes under the sip.conf
01:30.43clyrradok i am adding that now
01:31.10Sponge_bobso if your extension is 100 you put regexten=100
01:31.22*** join/#asterisk overworked554 (n=overwork@209.242.52.25)
01:32.34clyrradok i have added that - and still no change
01:32.47Sponge_bobrealod?
01:32.50clyrradyes
01:33.02Sponge_bobpm me your macro
01:33.07clyrradok
01:35.30*** join/#asterisk NoRemorse (n=bah@eth2462.vic.adsl.internode.on.net)
01:35.33NoRemorsehi all
01:35.56NoRemorseWhen I call someone on a sip client with call waiting enabled, I dont get a ring time just silence, any idea how this can be fixed please?
01:36.06NoRemorse*ring tone
01:38.58SplasPoodhah thats pretty slick.. *67 is useless
01:39.54andrejkwHey guys
01:41.26NoRemorseWhen I call someone on a sip client with call waiting enabled, I dont get a ring tone just silence, any idea how this can be fixed please?
01:45.32*** join/#asterisk trbldwine (n=trbldwin@c-71-194-161-170.hsd1.il.comcast.net)
01:47.33*** join/#asterisk jeebusmobile (n=jeebusmo@29palms-cuda1-68-170-42-234.losaca.adelphia.net)
01:53.25NoRemorseWhen I call someone on a sip client with call waiting enabled, I dont get a ring tone just silence, any idea how this can be fixed please?
01:54.06NoRemorseor is it a function of the client hardware
01:58.46*** join/#asterisk babyju (n=babyju@h-67-102-255-186.nycmny83.covad.net)
02:00.45clyrradI am having a rather strange issue - for some reason when a person leaves a voice mail the MWI is not activated - Asterisk records the voice mail and if you dial into Comedian Mail you can check the message - just the MWI light is not comming on - does anyone know what could be the cause of this?
02:03.25*** join/#asterisk littleball (n=littleba@26.203-123-30.leased.qala.com.sg)
02:04.24*** join/#asterisk file2 (n=IrcNet@out.clearnet.com)
02:04.24*** mode/#asterisk [+o file2] by ChanServ
02:04.33file2mooo
02:04.50Qwelllame
02:09.42andrejkwHey guys
02:09.58andrejkwI have a small problem, when I hang up, it doesn't get hung up on the other side.
02:10.16enjay-speakerphone?
02:10.48clyrradQwell - do you have any ideas for me?
02:10.53enjay-you hear the beep/beep/beep/beep when one side hangs up is that what you are referring to?
02:11.05clyrradandrewjwk - most likely a Firewall issue
02:11.15andrejkwYes
02:11.24enjay-Yes to my question or what?
02:11.30andrejkwexactly, if the  toher side stays on long enought he beep beep beep noise comes up
02:11.48enjay-thats because it HAS been hung up and you are still on speakerphone with no peer..
02:11.58andrejkwno speakerphone
02:12.03enjay-well handset on then..
02:12.10andrejkwno, cellphone
02:12.11andrejkw:\
02:12.18enjay-ah..
02:12.23enjay-yea cell should automatically hang up..
02:12.31andrejkwwell it doesn't, it sites there
02:12.33andrejkw*sits
02:12.43andrejkwforever...
02:12.52andrejkwwaiting for the other side to respond.
02:13.08enjay-hmm
02:13.29andrejkwand then I can't call the other side again'
02:13.35AJaymnAnyone use ShellShark for a Provider?
02:13.35andrejkwuntil I restart Asterisk
02:14.25*** join/#asterisk bjohnson (n=bjohnson@i216-58-59-1.cybersurf.com)
02:14.35enjay-you have a hangup macro defined?
02:14.38enjay-or statement rather..
02:15.10andrejkwYes
02:15.43enjay-might want to pastebin your output of the entire call process..
02:15.44NoRemorsehey guys I am trying to upgrade asterisk from 1.0.9 to 1.2.10. I have taken the "delete the old evrsion completely " approach, installed zaptel 1.2.7 and libpri 1.2.3 beforehand and addons 1.2.3 after, and everything runs fine until.... I create a default cdr_mysql.conf and I get a seg fault. I have the same install on another box and it works fine, what could be polluting my new install?
02:16.09NoRemorsethe old 1.0.9 talked to a local mysql 4 database
02:17.24andrejkwit never hangs up
02:18.25*** join/#asterisk kuku5 (n=kuku5@c-71-201-213-102.hsd1.il.comcast.net)
02:18.38kuku5I cant seem to stay registered with broadvoice - any suggestions?
02:19.08Qwellswitch providers
02:23.58*** join/#asterisk zepmantra (i=what@125.212.110.117)
02:24.26kuku5Got SIP response 409 "Conflict" back from 147.135.12.128
02:24.32[andromeda]Does anyone know a VoIP service that offers Toll-Free numbers, and will work with asterisk?
02:24.46kuku5and  NOTICE[21659]: chan_sip.c:4045 sip_reg_timeout:
02:25.57[andromeda]NoRemorse: the addons1.2.3 package does not compile the mysql function for asterisk correctly, it's missing some headers
02:28.08*** join/#asterisk clyrrad (n=ddd@CPE001195f553c7-CM0011aea484a4.cpe.net.cable.rogers.com)
02:30.20*** join/#asterisk P-NuT (n=P-NuT@CPE-60-227-93-75.nsw.bigpond.net.au)
02:31.46*** join/#asterisk c1sco (n=loke@c-24-5-215-69.hsd1.ca.comcast.net)
02:32.08c1scohey guys im having trouble making inbound calls via the pstn to my asterisk box
02:32.28c1scothe asterisk box receives the call, but the phone on the pstn never gets a response from the asterisk box
02:32.56c1scoi believe i have something misconfigure in extensions.conf
02:35.25clyrradwhat does the CLI say when it receives the call?
02:35.37c1scoshould i paste it here?
02:35.48clyrradif its not more than 2 lines
02:35.52clyrradotherwise pastebin it
02:36.21c1sco<PROTECTED>
02:36.21c1sco<PROTECTED>
02:36.21c1sco<PROTECTED>
02:36.34enjay-is the phone not ringing?
02:36.36c1scoyes
02:36.43c1scothen this..
02:36.43c1scoSIP/100-3805 answered SIP/trxtel.com-3c61f000
02:36.43c1sco<PROTECTED>
02:36.59c1scowhat is the native bridge?
02:37.01clyrradthese all on the same lan?
02:37.04*** join/#asterisk aketchel (n=Eraser@216.189.3.251)
02:37.04c1scono
02:37.17enjay-trxtel.conf is your provider?
02:37.18c1scoSIP/trxtel.com-3c61f000 is a sip provider
02:37.20enjay-err.com
02:37.21clyrradhave you checked the firewalls?
02:37.41c1scoi am port forwarding 6000-65000 ->> asterisk
02:37.49enjay-5060 = SIP
02:37.50c1scojust to make sure i dont miss any rtp, lol
02:37.53c1scoand 5060
02:37.57enjay-UDP
02:37.59c1scoyep
02:38.11c1scoits the response that is not getting there
02:38.19clyrradtemporarly disable the firewalls on each side - bet its one of them - I had this issue many times where it sits on native bridge - and it was always firewall related
02:38.20c1scowhen i answer the x-lite phone
02:38.49c1scoclyrrad i only have one firewall, the other side is the provider
02:39.00enjay-nat involved?
02:39.06c1scoyes
02:39.14enjay-nat=yes on the sip device?
02:39.17c1scothe asterisk box and the x-lite are behind nat
02:39.24enjay-not that it should matter actually since its local to your server..
02:39.31c1scothe asterisk box and the x-lite are on the same lan
02:39.41c1scoim trying to call from pstn to my xlite
02:39.51clyrradthen disable it on your side
02:39.51clyrradare you behinde a router?
02:39.51clyrradwith NAT?
02:39.56c1scoyes look
02:39.58enjay-SIP trunk have nat=yes?
02:40.03andrejkwOk, I have a SIP section called i2telecom.com (don't ask, my provider requires it).  And now I am trying to dial through it, but I don't want Asterisk thinkinking that I am trying to dial through the i2telecom.com server.
02:40.04c1scosip trunk?
02:40.12andrejkwHow can I come around this?
02:40.13c1scoi have no sip trunk
02:40.17c1scoi dont think
02:40.19enjay-ahh..
02:40.25enjay-thought you had a sip trunk to your provider..
02:40.38clyrradso your asterisk is behinde the NAT then
02:40.40c1scowell, explain how you refer to a trunk in extensions.conf?
02:40.47c1scoyes, asterisk is behind nat
02:40.55c1scoand my xlite is behind nat on the same lan
02:40.57enjay-c1sco from CLI type sip show registry
02:41.08*** join/#asterisk johnnyb (n=jonathan@207.155.33.225)
02:41.12c1scoenjay- nothing
02:41.21clyrradthen your xlite is not registered
02:41.22andrejkwAnyone, please?
02:41.23clyrradits not connecting
02:41.34*** join/#asterisk bjohnson (n=bjohnson@i216-58-59-1.cybersurf.com)
02:41.39enjay-not true..
02:41.42c1scoenjay- is is under sip show users
02:41.52c1scoit is*
02:42.03enjay-yea thats where xlite would show up..
02:42.14clyrradhow bout sip show peers?
02:42.27clyrradxlite will show under users instead of peers?
02:42.29c1sco100/100                    192.168.1.100    D          255.255.255.255  4996     OK (101 ms)
02:42.29c1sco2 sip peers [1 online , 1 offline]
02:43.08clyrradalright so Asterisk does see the xlite
02:43.14c1scoi believe so
02:43.23enjay-I dont wanna lead you down the wrong path cause from my experience you'd have to register with trxtel.com to send/receive "authenticated" calls..
02:43.24c1scoyes, i can make outgoing calls from xlite
02:43.26clyrradcan you dial out using the xlite?
02:43.48clyrradpastebin your extensions.conf
02:43.52c1scothis is what trx telecom says
02:43.56c1scoYou do not need to register with our servers
02:44.08c1scoclyrrad sorry, how do i pastebin?
02:44.10clyrradthis is SIP or IAX?
02:44.11c1scoits a url?
02:44.13c1scosip
02:44.15clyrradpastebin.com
02:44.40c1scoclyrrad just warning u my extensions.conf is ghetto, because im just learning
02:44.53c1scoim trying to bridge the gap from call manager to asterisk
02:46.21c1scoalmost done on pastebin
02:46.32c1scoits going...
02:47.12enjay-should give you a url for us..
02:47.18c1scoits just hangin
02:47.27c1scogotta another way to do this?
02:47.37clyrradno
02:47.41c1scook
02:48.00andrejkwOk, I have a SIP trunk nam i2telecom.com (don't ask, my provider requires it).  And now I am trying to dial through it, but I don't want Asterisk thinkinking that I am trying to dial through the i2telecom.com server. How can I do this?
02:48.22enjay-eh andrejkw?
02:48.39c1scopastebin gave me this error
02:48.40c1scoWarning: unlink(/home/pastebin/public_html/../cache/recent): No such file or directory in /home/pastebin/lib/pastebin/db.mysql.class.php on line 243
02:48.44andrejkwThe trunkname is i2telecom.com. For some reason it won't register without it.
02:48.50clyrradtry pastebin.ca
02:49.07andrejkwNow I am trying to call through it using Dial, would I do Dial(SIP/0000@i2telecom.com) ?
02:49.41c1scohttp://pastebin.ca/90872
02:50.15clyrradim guessing those first 2 lines are NOT in the file correct?
02:50.16c1scoi know im doing something horribly wrong...
02:50.20c1scocorrect
02:50.30clyrradphew
02:50.31c1scowell, line 2 is
02:51.09c1scoim basically focusing on, default1, globals, and pstn-inbound
02:51.55clyrradyea your file is a big mess
02:52.05clyrradfirst off your [general] should be at the top of the file
02:52.44c1scook
02:53.16c1scothe part that im confused about is the USERID part
02:53.27c1scoand how the ani number is matched
02:53.38c1scoor if the ani is matche
02:53.51enjay-did?
02:54.00c1scoum no
02:54.12Dovid.
02:54.12c1scomy number is [globals]
02:54.12c1scoUSERID=9255651913
02:54.13c1scoPHONE1=100
02:54.13c1scoPHONE1VM=100
02:54.13c1scoXLITE=SIP/1000
02:54.13c1scoEXTEN=100
02:54.15c1scowhops
02:54.17c1scosorry
02:54.20Dovid~pb
02:54.21jbot[pb] a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/
02:54.23c1scomy bumber is 7124325415
02:54.34c1scoand my xlite extension is 100
02:54.35*** join/#asterisk clyrrad1 (n=ddd@CPE001195f553c7-CM0011aea484a4.cpe.net.cable.rogers.com)
02:54.36Dovidclsco: please use pastebin
02:54.42Dovid~pb
02:54.43jbotsomebody said pb was a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/
02:54.47c1scosorry
02:54.51c1scoaccident
02:55.13*** join/#asterisk carrar (i=tim@osburn.com)
02:55.21carraranyone using postgres with their asterisk?
02:55.26carrarI see a -07 in the date
02:55.29carrar2006-07-17 13:18:05-07
02:55.38c1scoclyrrad any idea?
02:55.39carrarodd
02:55.53c1scocarrar its july
02:55.55c1sco?
02:55.57carrarno
02:56.01carrarafter the seconds
02:56.02carrarsee
02:56.15c1scomilliseconds?
02:56.23carrarevery entry has -07
02:56.26c1scolol
02:57.12carrarit's not in the master.csv file
02:57.12carrarjust in th db
02:58.36*** join/#asterisk johnnyb (n=jonathan@207.155.33.225)
02:59.25carrarhahah
02:59.25carraroh man
02:59.36carrarthats funny shit
02:59.51carrartime zone
03:00.14carrarI'm just not used to seeing the date like that
03:00.21c1scowow
03:00.23c1scoso im lost
03:00.33carrar-07  PST8PDT
03:00.41c1scomy inbound calls are not connecting to my xlite
03:00.43carraris what that is
03:00.55carrarI use xlite
03:01.00c1scocarrar pst is -8
03:01.06carrarnot right now
03:01.09carrarit's -7
03:01.11c1scois it
03:01.11carrarwe change
03:01.14c1scoahh
03:01.29c1scoi use xlite also
03:01.57c1scobut, when i call from pstn to xlite/asterisk, the pstn side never gets a response back from the asterisk/xlite
03:02.00andrejkwOk, I have a SIP trunk name i2telecom.com (don't ask, my provider requires it).  And now I am trying to dial through it, but I don't want Asterisk thinkinking that I am trying to dial through the i2telecom.com server. How can I do this?
03:02.49carrar1sco
03:02.52carrardo you Answer it?
03:02.55c1scoyes
03:02.56c1scoi do
03:03.02carrarin the exten
03:03.04c1scoand the pstn side gives me an error
03:03.10carrarweird
03:03.15c1scocannot connect
03:03.35carrarnone xlite sip work?
03:03.40c1sco?
03:03.47c1scowell, xlite makes outbound calls
03:03.56c1scoits something wrong i think with my extensions.conf
03:04.02c1scohttp://pastebin.ca/90872
03:05.04carrarpaste paste
03:05.04carrarheh
03:06.15*** join/#asterisk clyrrad1 (n=ddd@CPE001195f553c7-CM0011aea484a4.cpe.net.cable.rogers.com)
03:06.28carrarwhich one it taking the call?
03:06.34c1sco100
03:06.45carrarwhere is the n,Answer
03:06.56c1sco?
03:07.12carrartry Answer then Dial
03:07.24c1scoin pstn-inbound?
03:07.30carraryeah
03:07.46c1scoshould it be 1,Answer and 2,Dial?
03:07.56carrarI'm using that
03:08.00carrarfor my xlite
03:08.05andrejkwIs there any other way how to identify a SIP connection?
03:08.12c1scolet me try
03:08.15andrejkwInstead of by the name inside [ ]?
03:09.25c1scohey carrar what should USERID be?
03:09.51carrarwhatever number you want to match on
03:09.58c1scothats the caller id right?
03:10.05c1scoit should match all numbers
03:10.18c1scoinbound caller id numbers right?
03:10.39carraruse pattern matching
03:10.53carrarNXXNXXXXXX
03:10.56carrarNXXXXXX
03:11.06carrar_NXXXXXX
03:11.20carraror .
03:12.03Mr-packethow can i change what IP address asterisk trys to contact an upstream with..
03:12.05carrarandrejkw, why not set a customer caller id name
03:12.06*** part/#asterisk P-NuT (n=P-NuT@CPE-60-227-93-75.nsw.bigpond.net.au)
03:12.09carrarcustome
03:12.16Mr-packetmy asterisk is running on a multi-interface machine
03:12.20andrejkwwould that work?
03:12.24c1scoMr-packet externip in sip.conf
03:12.33c1scoi think
03:12.44Mr-packetc1sco. thats what i thought
03:12.57Mr-packetdoes'nt seem to want to work though
03:13.09c1scois it under global?
03:13.12carrarc1sco, could use the switch option too
03:13.27carrarif you want a catch all
03:13.50c1scohey
03:14.07carrarI always match the exact number
03:14.20c1scowhen i call from pstn to my xlite, the rtp should go sip proivder --> asterisk box ---> xlite and back right?
03:14.48carrarpstn->*->xlite?
03:14.55c1scono asterisk?
03:14.59andrejkwcarrar: doens't work, I get  No route to destination.
03:15.16andrejkwI need some other way to reffer to the connection in the Dial function.
03:15.26carrarcaller id name
03:15.30*** join/#asterisk topping (n=topping@001-785-676.area1.spcsdns.net)
03:15.36carrarset it to whatever you want to identify what call it is
03:15.42andrejkwI can't use the name of it, because it contains a domain, and then it thinks I am trying to call out using that domain.
03:15.55andrejkwcarrar: I just tried, it doesn't work.
03:16.00carrarit does work
03:16.26c1scocarrar when i sniff the rtp, i see xlite --> asterisk and thats it
03:16.31carrarSet(CALLERID(name)=Call From SIP)
03:16.43andrejkwBut you don't understand.
03:16.48andrejkwThat's not what I mean.
03:16.49*** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn)
03:16.54c1scothis is what i get in console
03:16.54c1sco-- Executing Dial("SIP/trxtel.com-3c619000", "SIP/100|12|tr") in new stack
03:16.54c1sco<PROTECTED>
03:16.54c1sco<PROTECTED>
03:16.54c1sco<PROTECTED>
03:16.54c1sco<PROTECTED>
03:17.06carrarxlite should be talking only to asterisk
03:17.21carraris xlite natted?
03:17.25c1scodo i need stun if im behind nat?
03:17.32carrarshouldn't
03:17.33andrejkwI have a SIP provider with [i2something.com] and now I have to reffer to it in Dial().
03:17.40carraras long as your FW is passing it
03:17.43*** part/#asterisk aketchel (n=Eraser@216.189.3.251)
03:17.52andrejkwWhen I try to reffer to it, it thinks I actually want to use i2something.com to dial.
03:17.54carrarsip.conf
03:17.54carrarnat=1
03:17.55carrar?
03:18.03c1scothe firewall if forwarding 5060 and 6000-65000 to the asterisk
03:18.19c1scosip.conf nat=no
03:18.30carrarenable nat for your xlite client
03:18.32c1scoxlite and asterisk are on the same lan
03:18.34carrarin sip.conf
03:18.38carraroh
03:18.43c1scoboth behind nat
03:19.11carrarcan you call your xlite from another xlite?
03:19.15c1scoyes
03:19.18c1scou wanna try
03:19.23carrarnot really
03:19.24c1scou can use 101
03:19.26Mr-packeti'm slowly going insane
03:19.32c1scocarrar yes u can
03:19.41c1scoand i can make outgoing calls
03:19.48c1scojust this inbound is not working
03:19.50carraris your PSTN TDMA?
03:19.59c1scoTDM?
03:20.01carrarwhat is your pstn connection?
03:20.13c1scoits txr telecom thats all i know
03:20.21carrarmaybe * is trying to bind the two together
03:20.26c1scohttp://www.trxtel.com/
03:20.28carraris it sip?
03:20.29c1scoi think so
03:20.36c1scocarrar i think you are right
03:20.48c1scoits trying to go pstn -> xlite
03:20.53c1scoand i cant do that
03:20.58*** join/#asterisk pdtmobile (n=ptinsley@c-68-53-40-50.hsd1.tn.comcast.net)
03:21.00c1scobecause my portfowarding is going to asterisk
03:21.07xachencarrar : trxtel is SIP and IAX
03:21.15xachener, c1sco I mean
03:21.23carrarwhat is in the console?
03:21.29c1scoscroll up
03:21.31carrardoes it say it is trying to do that?
03:21.31c1scoi pasted it
03:22.07carraryeah Attempting native bridge
03:22.12c1scowhat is that shit
03:22.16carrarbut does it faile or succed?
03:22.22c1scook
03:22.24c1scohere is how it goes
03:22.31c1scoi call from the pstn to my xlite
03:22.36c1scoxlite rings
03:22.38c1scoi answer xlite
03:22.50c1scothe pstn call just keeps ringing
03:22.59c1scoxlite has dead air
03:23.15c1scoand pstn call says sorry call cannot be completed
03:23.19carraris your pstn g.711?
03:23.26c1sconot sure
03:23.55carrarso try putting a xlite client on the internet
03:23.58carrarsee if that works
03:24.06c1sco?
03:24.08c1scowhat u mean?
03:24.12carrarusing a internet IP
03:24.17carrarroutable one
03:24.22c1scohow
03:24.24c1scoi dont have one
03:24.42c1scou think its nat?
03:24.51carrarcould be
03:24.58carrarcould be trying to bridge
03:24.59carrarOR
03:25.04carrarcodec missmatch
03:25.19carrarI would think it would say something baout codec though
03:26.26carrarso enable nat for the hell of it
03:26.36carraron sip.conf & xlite
03:26.37c1scoenable nat?
03:26.40c1scoheh?
03:26.48c1scowhat do you mean?
03:26.53carrarnat=1
03:26.58c1sconat=yes?
03:27.03carrarsame thing
03:27.22c1scowhen i do that i think the phone doesnt register
03:27.32carrarshould
03:27.38*** join/#asterisk P-NuT (n=P-Nut@CPE-60-227-93-75.nsw.bigpond.net.au)
03:27.40c1scolet me try
03:27.54c1scoit registered
03:27.59c1scolet me try
03:28.06carrarI use nat=1 just cause I never know where they are coming from
03:28.39carraralso what are you allowing in sip.conf for codecs?
03:28.49c1scoi pickup the xlite and the pstn phone keeps ringing
03:28.58c1scocarrar all
03:29.02c1scoi think
03:29.11carrartry these
03:29.11carrardisallow=all
03:29.11carrarallow=ulaw
03:29.12carrarallow=alaw
03:29.15c1scok
03:29.28carraralthought this sounds like call setup issues
03:29.47c1scoi agree
03:29.54c1scoyeah codec did not work
03:30.02c1scoit really sounds like call setup
03:30.27andrejkwIs there a way to give a SIP connection 2 identifiers?
03:30.42carrarc1sco, sets see your sip.conf
03:30.53c1scok
03:31.31andrejkwCome on guys :(
03:31.34c1scohttp://pastebin.ca/90911
03:31.35andrejkwThere has to be a way.
03:31.37andrejkwThis is crazy.
03:31.47carrarok
03:31.49carraractaully
03:31.55carrarback in your extensions.conf
03:31.59c1scoyeah
03:32.00carrarwhere are you calling you xlite phone
03:32.05c1scoyeah
03:32.10carrarI see it defined in globals
03:32.13c1scoumm, pstn-inbound ???
03:32.22carrarPHONE1?
03:32.27carrar== 100
03:32.37pdtmobileandrejkw: what do you mean?
03:32.42c1scoyes
03:32.51c1scothats for pstn-inbound
03:32.55c1sco2,Dial
03:32.57carrarreplace PHONE1 with c1sco
03:33.00carrarerr
03:33.06carrarXLITE
03:33.10c1scowhere?
03:33.13c1scoin globals?
03:33.17c1scoor pstn-inbound?
03:33.18carrarline 64
03:33.27carrarthats where you want to dial it right?
03:33.29c1scowhat the url?
03:33.34carraryeah
03:33.38c1scogive me the pastebin url
03:33.40carrarhttp://pastebin.ca/90872
03:33.55andrejkwWell I have a SIP connection and my provider requires that I have "i2telecom.com" as a trunk name (which also ebcomes the identifier). And when I try to dial through it (number@i2telecom.com), Aterisk thinks that I am actually trying to dial through the domain i2telecom.com and not the SIP connection.
03:34.04c1scook PHONE1 should be XLITE?
03:34.09c1scoline 64?
03:34.29carrarxlite phone is username 1000 right?
03:34.33c1scono
03:34.37carrarwhat is it?
03:35.05carrarYou can't dial just 100
03:35.07c1scousername im using is extenion 100
03:35.09Dovidlol
03:35.10carrarYou can dial SIP/100
03:35.15c1scook
03:35.20c1scoso what should i change
03:35.26carrarchange:
03:35.26carrar#
03:35.27carrarexten => ${USERID},1,Dial(${PHONE1},30)
03:35.28carrarwith
03:35.28Dovidclsco: what r u tryin to do ?
03:35.31carrar#
03:35.33c1scoin global XLITE=SIP/100
03:35.38carrarexten => ${USERID},1,Dial(SIP/100,30)
03:35.40c1scoDovid dial inbound from pstn --> xlite
03:35.44carrarsee if that works
03:35.47Dovidok
03:35.50c1scocarrar let me try
03:36.13*** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar)
03:36.16andrejkwAnyone, please? :'(
03:36.31*** part/#asterisk mog (i=ejabberd@68.62.237.103)
03:36.33carrarcan also change PHONE1=100 to be PHONE1=SIP/100
03:37.02carraralso
03:37.07c1scoalso?
03:37.17carrarI'v never used a variable in the patern maching area
03:37.28*** join/#asterisk mog_home (n=mogorman@68.62.237.103)
03:37.28*** mode/#asterisk [+o mog_home] by ChanServ
03:37.34carrarexten => 9255651913,1,Dial(SIP/100,30)
03:37.36carraruse that
03:37.39carrarjust to test
03:37.40c1scok
03:41.10c1scocarrar no luck
03:41.34carrarwhat does console say
03:41.39c1scosame thing
03:43.36carrarpstn IAX2?
03:43.57c1scosip
03:44.02carrari don't see it
03:44.29andrejkwAnyone, please? :'(
03:44.54carrarYou have the contet commented out
03:45.04c1scono
03:45.07c1scothats another sip account
03:45.35carrarpstn-inbound is the context in extension.conf right?
03:45.44c1scoyeah
03:45.52carrarSo where is it in sip.conf?
03:46.26carrarHow is the inbound call getting put into the pstn-inbound context?
03:46.34c1scodunno man
03:46.39carrarheh
03:46.49c1scotrx telecom sends me the call because i gave them a url
03:46.56c1scoSIP/100@loke.sciarrilli.com
03:48.02carrarYou need to setup a entry in sip.conf that will take incoming calls from then and put them in the pstn-inbound context
03:48.09carrarthem
03:48.14c1scohow?
03:48.19carrarRTFM :)
03:48.25c1scowhich one
03:48.34c1scodocumentation for asterisk doesnt seem the greates
03:48.39carrarjust the sip.conf original file
03:51.45*** join/#asterisk bmg505 (n=leon@c1-114-9.rndf.isadsl.co.za)
03:51.47carrarI am sure your sip provider can send you a example if they are any good
03:52.03c1scowhat is a good sip provider?
03:52.06c1scofree one
03:52.16docelmogood luck
03:52.34docelmoinexpensive one is plainvoip
03:52.36carrarfree sip is like fre gas
03:53.12*** join/#asterisk ZX81 (n=ZX81@203-173-176-166.bliink.ihug.co.nz)
03:53.20ZX81problems with mailing list?
03:53.30ZX81Diagnostic-Code: SMTP; 451 4.4.1 reply: read error from lists.digium.com.s8b2.psmtp.com.
03:53.38*** join/#asterisk campbell (n=csteven@ip-202-37-228-10.internet.co.nz)
03:53.51campbellhi
03:54.09hads|homehey ZX81
03:54.13ZX81hi
03:54.13ZX81:D
03:54.31hads|homeHow's tricks?
03:54.35campbellanyone here running multiple TDM400p cards? if so would you recommend it?
03:55.10hads|homecampbell: Depends on the motherboard. I have one place running two with no trouble.
03:55.23Dovidcampbell: depends on the MB
03:55.32andrejkwWell I have a SIP connection and my provider requires that I have "i2telecom.com" as a trunk name (which also ebcomes the identifier). And when I try to dial through it (number@i2telecom.com), Asterisk thinks that I am actually trying to dial through the domain i2telecom.com and not the SIP connection. How can I fix this?
03:55.55campbellyeah that's what i'd read so far, i guess i'll give it a go and see how i get on, cheers
03:57.06ZX81hads|home good
03:57.08ZX81:D
03:57.12ZX81busy but good
03:57.12ZX81:D
03:57.21ZX81in New Zealand for another two weeks
03:57.25ZX81before returning to Italu
03:57.28ZX81*italy
03:57.38hads|homeCool. Having fun over there?
03:57.45ZX81yeah, in love!!!
03:57.46ZX81:D
03:57.50hads|home:)
03:57.54ZX81campbell: how many is multiple?
03:59.27andrejkwanyone? please?
04:00.01hads|homeandrejkw: How about Dial(SIP/i2telecom.com/12345)
04:00.17andrejkwlet me try
04:00.22*** join/#asterisk mdiehl (n=mdiehl@c-69-252-219-76.hsd1.nm.comcast.net)
04:00.40mdiehlHi all.
04:00.49*** join/#asterisk l0ke (n=loke@c-24-5-215-69.hsd1.ca.comcast.net)
04:01.02l0keim back
04:01.03l0keits c1sco
04:01.07l0kei got dropped
04:01.10l0kecarrar u here?
04:01.15l0keDovid u here?
04:01.18mdiehlAnyone gotten Gnomemeeting to work with Asterisk?
04:01.44andrejkwNope, didn't work
04:01.52mdiehlIn h.323, that is.
04:03.48*** part/#asterisk ZX81 (n=ZX81@203-173-176-166.bliink.ihug.co.nz)
04:04.07l0kei changed password
04:06.38andrejkwWell I have a SIP connection and my provider requires that I have "i2telecom.com" as a trunk name (which also ebcomes the identifier). And when I try to dial through it (number@i2telecom.com), Asterisk thinks that I am actually trying to dial through the domain i2telecom.com and not the SIP connection. How can I fix this?
04:08.24mdiehlHas anyone gotten ANY h.323 client to work with Asterisk?
04:09.39hads|homemdiehl: Doesn't ekiga use SIP these days?
04:10.21mdiehlYes, but I'm trying to get some h.323 phones to work and I thought I'd test with an old gnomemeeting client.
04:10.30mdiehlI'm mostly concerned with getting h.323 working.
04:11.01hads|homeAha, I know nothing of h323.
04:13.42mdiehlBummer.
04:13.44drrayandrewjk , place a [i2telecom.com] section in sip conf?
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04:57.23NoRemorsehi all
04:57.33NoRemorsewhat do I have to do to get * to create  auniqueid again please?
04:57.36Doviddoes asterisk from asterisk.org work on FeeBSD ?
04:59.49QwellDovid: yes
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05:03.04NoRemorsewhat do I have to do to get * to create  auniqueid again please?
05:03.28Dovid@Qwell for free BSD do i need to do ztdummy ?
05:03.42Qwellif you want meetme or timing
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05:05.16russellbNoRemorse: it does it automagically
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05:10.10NoRemorseYou have two options in /usr/src/asterisk-addons:
05:10.10NoRemorse1. Add a CFLAGS+=-DMYSQL_LOGUNIQUEID to the Makefile.
05:10.10NoRemorse2. Add a #define MYSQL_LOGUNIQUEID to the top of the sourcefile.
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05:17.37mdiehlAnyone gotten Gnomemeeting to work with Asterisk? In h.323, that is.
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05:52.57JThmm
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06:07.11FuriousGeorgei think the switch built into my snom messes with the quality of the call
06:07.31FuriousGeorgesounds like packet loss
06:07.38FuriousGeorgesmells like it too
06:11.16stoffell_hFuriousGeorge: are you running so much traffic over the pc ?
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06:26.57docelmowhat does packet loss smell like?   Im real curious
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06:38.35negativecreephi all
06:44.27docelmomost are sleeping like where I am going to
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06:50.23diLLecanyone there ?
06:51.00docelmonope..  going to BED
06:51.33diLLec:)
06:51.38*** join/#asterisk sadiqsb (n=sad@213.132.231.57)
06:52.17sadiqsbhi can anyone help me setup asterisk for my voip provider
06:52.25sadiqsbhi can anyone help me setup asterisk for my voip provider
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06:52.55sadiqsbam using only IPbased SIP trunking
06:53.15sadiqsbinternally i can communicate , i cont call put side
06:53.16negativecreepsadiqsb: what the problem?
06:53.32sadiqsbi installed asterisk on fc5
06:53.38negativecreepyou can't call other sip providers?
06:53.42sadiqsbinternal extention  are working fine
06:53.55sadiqsbi cont call through sip provider
06:54.07sadiqsbyes
06:54.14sadiqsbcan u help me to set it up
06:54.18negativecreepwhich providers?
06:54.28negativecreepfwd, sipgate?
06:54.33sadiqsbsip.kingcalls.com
06:55.18negativecreephave you registered with them?
06:55.23sadiqsbi thnik some problem with sip.conf /extention.conf
06:55.24sadiqsbyes
06:55.29negativecreepas a peer?
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06:55.37sadiqsbyes
06:55.39negativecreepthey should allow calls from ur sip users.
06:55.45negativecreeppaste ur extensions.conf
06:55.50sadiqsbk
06:59.04*** join/#asterisk sadiqsb (n=sad@213.132.231.57)
07:00.24sadiqsbhi am back
07:01.28sadiqsbmore extensions.conf
07:01.28sadiqsb;KY
07:01.28sadiqsb;;;;;;;;;;;;;
07:01.28sadiqsb[vsky]
07:01.28sadiqsbexten => 1001,1,Dial(SIP/bbc)
07:01.29sadiqsbexten => 1002,1,Dial(SIP/test)
07:01.30sadiqsbexten => 1003,1,Dial(SIP/krish)
07:01.32sadiqsbexten => 1004,1,Dial(SIP/ravi)
07:01.34sadiqsbexten => 1005,1,Dial(SIP/sad)
07:01.36sadiqsbexten => 1006,1,Dial(SIP/zen)
07:01.38sadiqsbexten => 84.11.110.56,1,Dial(SIP/1001)
07:01.40sadiqsb;exten => _0[1-9].,1,Dial(SIP/out/${EXTEN})
07:01.42sadiqsb;exten => _00[1-9].,1,Dial(SIP/out/${EXTEN})
07:01.45sadiqsbexten => _9.,1,Dial(SIP/${EXTEN:1}@out,60
07:01.46sadiqsbinclude => default
07:01.48sadiqsbthis is my extentins.conf file
07:02.01my007msr_evolution, never POST in the channel
07:02.15my007mslol sadiqsb
07:02.19sadiqsbok
07:02.25sadiqsbi have a probel
07:02.36sadiqsbsome one is helping me
07:02.42sadiqsbi forgort his nick
07:02.59my007msu can use http://pastebin.ca/
07:03.04sadiqsbi got problem with sip truning with provider
07:03.23hads|home~pb
07:03.30jbotfrom memory, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/
07:03.30sadiqsbcan u healp me in this
07:05.22hads|homesadiqsb: It's rude to PM people.
07:06.11sadiqsbsorry
07:06.20sadiqsbcan u help  please
07:06.38FuriousGeorgestoffell_h: i know this is a bit late, but its just constant traffic.  they use windows remote desktop
07:10.36sadiqsb<FuriousGeorge> can u help me in sip truning with provider through asterisk
07:12.53sadiqsbhi can anyone help me Asterisk SIP trunk with VOIP Provider
07:14.21FuriousGeorge~docs
07:14.24jbotdocs is probably probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html
07:14.31hads|home~thebook
07:14.32jbotwell, thebook is a book called  Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
07:15.31sadiqsbi went through docs
07:15.31sadiqsbinternally extention are working
07:15.46sadiqsbproblem for longdistence calls through provider
07:16.08sadiqsbJBOT---can u help me
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07:18.04sadiqsbnegativecreep --can u help with sip trunk
07:20.16FuriousGeorgenegativecreep is a nirvana song, right
07:21.10hads|homeYep
07:21.23sadiqsbFuriousGeorge> he (neg) was helping me  he went off
07:21.37sadiqsbFuri can u help me in sip trunng
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07:26.44negativecreepback
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07:31.26sadiqsb<negativecreep> can u hewlp me
07:31.42sadiqsbin sip truning with provider
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07:54.01CrashHDcan anyone explain why I'm getting 99.975586 during my zttest (this is with a sangoma a104d) and not higher?
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08:08.11CrashHD*crickets*
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08:13.51Nobbieheya =)
08:14.41Nobbiei'm struggling to find documentation on call transferring. currently when users do a blind transfer to an extensions which is busy, the call is dropped. is it possible to make it ring back to the transferee instead of dropping it ?
08:15.51my007msNobbie, make mirco for  transferee and add this in it
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08:18.37Nobbieadd which in it ?
08:18.39Nobbie;)
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08:38.35Bert-hello there
08:39.35Bert-I have a little question : yesterday, all was working fine, and this morning, I'm unable to register to asterisk : 401 unauthorized. I really don't understand why, as I changed nothing :(. Is there a way to find why can't I register to * anymore please ?
08:42.10CrashHDis 99.975586 acceptable with a sangoma card?
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08:49.58Nobbieheya =)
08:50.23Nobbiesorry, got cut off. what was that code i had to add for the macro to not have calls dropped when blindly transferred to a busy extension ?
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08:55.55|oranjia|hello guys :)
08:58.04Nobbiehi
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09:23.26smlcoffeeevening
09:23.34Sonderbladei have a python AGI script which ends with sys.exit(99), but on asterisks console it outputs: AGI Script testagi.py completed, returning 0, strange?
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09:27.47heffernanhi.
09:28.04heffernani want to set CFIM in asterisk with exten => _*21*X.,1,Set(${DB(CFIM/${CALLERIDUM})}=${EXTEN:4})
09:28.10heffernanbut that does not work.
09:28.14*** join/#asterisk kindor (n=roy@office.open-ict.nl)
09:28.16Strom_CCFIM?
09:28.21*** join/#asterisk tparcina (n=tparcina@lns02-1506.dsl.iskon.hr)
09:28.22kindorsup
09:28.27tparcinahi channel!
09:28.35kindorhow can i disable the application /system/preferences toolbar?
09:28.38Strom_Clol internets
09:28.39kindoror remote it even
09:28.44kindoroops
09:28.45heffernanStrom_C: that's not important anyway, what i'm wondering about is how i can set a db value in the dialplan?
09:28.47kindorsorry
09:28.52heffernani can't find anything documented.
09:28.58Strom_Cheffernan: show application set
09:29.03Strom_Cyour syntax should be:
09:29.19Strom_CSet(DB(Family/key)=value)
09:29.31heffernanokay, no $-s then?
09:29.43|oranjia|has anyone tried to send faxes over premi cells?
09:29.49tparcinaheffernan: and Set(variabla=DB(family/key))
09:29.50Strom_Cno, not when you're storing data into a variable or function using Set()
09:29.57Strom_Ctparcina: NO NO NO
09:30.10Strom_Ctparcina: Set(variable=${DB(family/key)})
09:30.57heffernanStrom_C: and dbdel?
09:31.03tparcinastrom: ok, i was writing from my head
09:31.19Strom_CDBDel(family/key)
09:31.41heffernanStrom_C: okay, so they've kept dbdel.
09:32.00tparcinaStrom: do you know why Set(DB(Family/key)=$VARIABLE) doesn't work?
09:32.04*** join/#asterisk effectiveape (n=nick@82.153.22.16)
09:32.05Strom_Cheffernan: yes, dbdel is still there; dbput and dbget have been deprecated
09:32.20Strom_Ctparcina: because you have to type ${VARIABLE}
09:32.22effectiveapeHowdy. As exepcted still having problems ;)
09:32.24Strom_Cnot $VARIABLE
09:33.43tparcinaStrom: I have this - exten => 2,3,Set(DB(forward/${CALLERID(number)})=${FORWARD})
09:33.52tparcinaand it doesn't work
09:34.18Strom_Cbecause you're passing the wrong argument to the CALLERID function
09:34.26Strom_Cit's CALLERID(num)
09:34.35effectiveapebt complaining that the number isn't recognised even though it seems correct
09:34.35Strom_Cshow function CALLERID
09:35.30tparcinao hope that's it. yesterday i have spent two hours trying to figure it out :))
09:35.36Strom_Ctparcina: try it
09:36.14tparcinai will as soon as i eat my donuts :))
09:37.15tparcinayou know every thing has it's priority ;))
09:37.42Strom_Cso that explains why your emoticons have double chins!
09:37.54Strom_C:)
09:38.10tparcinai have exten => 2,2,Eat(cake) and then exten => 2,3, Set(DB... :))
09:38.44Strom_Cheh
09:38.47Strom_Cmmm, cake
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09:39.34tparcinai don't know why but i like to put double chins :)) it looks bether this way (at lest for me)
09:39.50effectiveapei was wondering if bt required full msisdn but it still complains :/
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09:42.11tparcinaeffectiveape: bt is british telecom?
09:42.23effectiveapeyeah
09:42.30tparcinaand oyu are using PRI?
09:42.38effectiveapei'm just getting the 'this number is not recognised' woman
09:42.44effectiveapebri
09:43.06effectiveapeisdn2e
09:43.15tparcinawell, what do you see on CLI, what number does asterisk dial out?
09:43.43effectiveapeit shows a Dial with the number i've entered (which is my mobile)... 07970xxxxxx
09:43.59negativecreepif i dial outgoing calls through a provider like fwd, i will be able to meter those calls...or will i lose track of them?
09:44.11Strom_Cnegativecreep: what do you mean "meter"?
09:44.35tparcina<negativecreep: yes, they will be recodred in CDR or database (if oyu use one)
09:44.36negativecreepStrom_C: calculating their usage..accumulating CDR
09:45.10tparcinaeffectiveape: sorry, can't help you out. but maybe you can help me. how can i see who is the owner of one mobile number?
09:45.35negativecreepWhat do you recommend as an easy to use solution for prepaid billing.
09:45.41negativecreepI dont need much fancy stuff though.
09:45.44effectiveapeyou mean the provider (ie orange etc...)
09:46.17effectiveapeor the indivdual?
09:46.24tparcinaeffectiveape: the individual
09:46.27Strom_Cnegativecreep: for postpaid you could just process the CDRs
09:46.38Strom_Cnegativecreep: for prepaid, I think there's something called astcc
09:47.35negativecreepStrom_C: All i need is a solution that can check that if the user connecting has xx number of minutes left to call..if yes, then he can proceed. also, if the number of minutes drop to 0, call should be disconnected.
09:47.57Strom_Cnegativecreep: like i said, check out astcc :)
09:48.07effectiveapeEach phone company will have directory enquiriesif you know the provider.
09:48.34tparcinaeffectiveape: it starts with +44795.......
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09:49.21negativecreepStrom_C: thnx
09:50.03effectiveapeYeah you can't rely on that since numbers are ported and such.
09:50.39effectiveapeYou might try them all - there are only 5 ;)
09:50.41tparcinaeffectiveape: what does it mean that number is ported?
09:50.47effectiveape(or well 6 now)
09:51.16effectiveapeWhen you are on one contact (eg vodofone) you can move your number to another contract (eg orange)
09:51.26effectiveapeTo keep your number
09:51.37tparcinaeffectiveape: ok, thank you
09:51.37carl0s-hmm
09:51.48effectiveapesorry i couldn't be more help.
09:51.57Strom_Cit's so weird to hear people talk about vodafone again :)
09:52.02effectiveapeheh
09:52.24effectiveapeyep still got the name here ;)
09:52.25Chris-NBhi
09:52.25carl0s-I have been getting read to ask questions on least-cost routing UK mobile numbers. Ported numbers are certainly a problem. I wonder if there is a public database of uk mobile numbers and associated GSM operators.
09:52.26tparcinaeffectiveape: it realy isn't that inportant anyway. she probably won't call me... :((
09:52.39Chris-NBcan someone plz look at that: http://pastebin.ca/91121 and can tell me what I'm doing wrong?
09:52.42effectiveape<PROTECTED>
09:52.50Chris-NBI'm trieing to connect two asterisk via IAX2
09:52.51Strom_CChris-NB: lemme have a look
09:53.02effectiveapeCan you simulate a dial from the cli?
09:53.08effectiveapeTo external
09:53.22Chris-NBbut get err : /
09:53.23Chris-NB<PROTECTED>
09:53.23Chris-NB<PROTECTED>
09:53.42Strom_Cbecause you're using md5 auth on one box and no md5 auth on the second?
09:53.54Strom_Ceffectiveape: yes, just type "dial"
09:54.54effectiveapedoes that just workj for known extensions?
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09:55.19appelzaHi
09:55.43appelzaWhat permisions should I change in AMP to allow mp3 file uploads for on hold music?
09:55.49Chris-NBStrom_C, tried even with md5 auth on both boxes
09:55.54Strom_Cappelza: read the topic and go to #freepbx
09:56.11appelzaok
09:56.11Chris-NBwhen I try to call, i get this IAX debug output:
09:56.12appelzatnx :p
09:56.14Chris-NBTx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX     Subclass: REJECT
09:56.18Strom_CChris-NB: and with auth=md5 commented out?
09:56.23Chris-NB<PROTECTED>
09:56.23Chris-NB<PROTECTED>
09:56.28Chris-NBon both boxes
09:56.32Chris-NBone moment
09:56.46frenzyhi.. I've got the grandstream bt102. When I hit the hangup for just a tiny second I get a new line while the other goes over to hold... how to do get the line again ?
09:56.53*** join/#asterisk zepmantra (i=what@125.212.110.115)
09:57.03effectiveapecan you show me what i'd type to dial 07970xxxxxx on Zap/g1 ?
09:57.03tparcinaStrom, exten => 2,3,Set(DB(forward/${CALLERID(num)})=${FORWARD}) works, thank you!
09:57.36Chris-NBsame without auth
09:57.41Chris-NBWARNING[5683]: chan_iax2.c:6986 socket_read: Call rejected by 192.168.68.156: No authority found
09:57.56Chris-NBbut there is an authority!
09:58.16Strom_CChris-NB: do an iax2 debug and pastebin the output
09:58.33Strom_Ctparcina: the wonders of reading documentation :)
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09:59.27Chris-NBhttp://pastebin.ca/91128
09:59.47Strom_Coh wait, i'm a moron.  try username- instead of user=
09:59.49zepmantraheelo just wanna ask i have 2fxo, 2fxs , 1 fxs port is  not blinking at the back but i have put power on the card for the fxs , on "dmesg" it says "Unable to do INITIAL ProSLIC powerup on module 3 Module 3: FAILED FXS (FCC)" <---is my card destroyed
09:59.52Strom_Cs/-/=/
10:00.09Chris-NBStrom_C, on both iax.conf?
10:00.13Strom_CChris-NB: yes
10:00.21Chris-NBjust a mom.
10:00.27frenzy?
10:00.28*** join/#asterisk darviria (n=dvr@194-105-181-29.ifb.co.uk)
10:00.29Strom_Czepmantra: how long have you had the card
10:00.34*** join/#asterisk Assid (i=assid@203.115.83.215)
10:00.35*** join/#asterisk hi365 (n=hi365@bzq-167-158.dsl.bezeqint.net)
10:00.38Strom_Czepmantra: and it's a tdm400p, right?
10:00.38zepmantraabout 1 week
10:00.51zepmantratdm400p clone, openvox a20022p
10:01.00Strom_Cewwww
10:01.09Strom_Cgrind it up and use it as fertilizer
10:01.14Strom_Cthen get a real tdm400p
10:01.16Strom_C:)
10:01.18Chris-NBStrom_C, same thing! : /
10:01.32Strom_CChris-NB: you are doing an iax2 reload after every change, right?
10:01.38zepmantraStrom_C , my clone card dead?
10:01.49Chris-NBStrom_C, i dig a reload. do i have to do iax2 reload ?
10:02.03hi365Hello! has anyone ever seen this with a sangoma A200?
10:02.04hi365ZT_CHANCONFIG failed on channel 3: Invalid argument (22)
10:02.09Strom_CChris-NB: iax2 reload will reload just the iax2 configuration
10:02.26Strom_Chi365: you probably didnt correctly configure zaptel.conf or zapata.conf
10:02.29Chris-NBStrom_C, and reload the hole thing, so this should do.right ?
10:02.31frenzy<PROTECTED>
10:02.39Strom_Cfrenzy: flash again
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10:02.52Strom_CChris-NB: iax2 reload on both boxes, right?
10:02.53frenzyhuh ?
10:03.01Strom_Cfrenzy: a short hangup is called a hookflash
10:03.08Chris-NBStrom_C, do you know where I can lookup how to configure IAX between two boxes?
10:03.11Strom_Cor just glash for short
10:03.13Strom_Cer, flash
10:03.13Chris-NBStrom_C, jep, did it on both
10:03.17Strom_CChris-NB: hmm
10:03.25frenzytried that out
10:03.27Strom_Cdo an iax2 debug and pastebin the output
10:03.31frenzybut I just get another new line
10:03.40Chris-NBStrom_C, looked at the voip-info and followed that configuration
10:03.52Strom_CChris-NB: what version of asterisk are both boxes running?
10:04.06hi365Strom_C -> http://pastebin.ca/91132
10:04.09ZeeekChris-NB the iax to iax box thing was discussed in -users - check the archives
10:04.33Chris-NBStrom_C, iax2 debug lookes like i posted before
10:04.49Chris-NBStrom_C, Asterisk versions are 1.2.6 and 1.2.7.1
10:05.01Chris-NBZeeek, I'll look at that. thx
10:05.11Strom_Chi365: are you running asterisk@home?
10:05.32Strom_CChris-NB: ok, link me again
10:05.36hi365yup. trixbox 1.1.1
10:05.46Strom_Chi365: ok, so read the topic of this channel
10:05.53Chris-NBStrom_C, http://pastebin.ca/91128
10:05.56hi365hu?
10:06.01Strom_Chi365: you're supposed to go to #freepbx if you're using trixbox
10:06.19hi365right, but this is a card issue
10:06.41hi365this isnt a matter of configuring extensions or trunks
10:06.57Strom_Chi365: no, it isnt, but the config is generated by a utility that comes with trixbox
10:07.15Strom_Cso therefore it falls into the trixbox category
10:07.31hi365ok. thanks
10:07.36Strom_CChris-NB: is that the complete debug output?
10:08.51Chris-NBStrom_C, thats after the reject: http://pastebin.ca/91138
10:09.05Chris-NBStrom_C, but I think it's not from the call
10:09.44Strom_CChris-NB: just for kicks, try commenting out everything but type=friend and context=default on both systems
10:09.48Strom_CChris-NB: and see if it works
10:10.19Chris-NBStrom_C, also username and secret ? but I deactevated the guest accound
10:10.59Strom_CChris-NB: yes.  lacking a username, it will use the entry name
10:11.21Chris-NBStrom_C, k. I'll try
10:11.54*** join/#asterisk johnnyb (n=jonathan@207.155.33.225)
10:11.59*** join/#asterisk Dibbler (n=Dibbler@zidane.pi-net.net)
10:12.32Chris-NBStrom_C, same here: http://pastebin.ca/91144
10:13.01Chris-NBthis is the dial statement: exten => _8XXX,1,Dial(IAX2/tecLas0016/${EXTEN:3},30,r)
10:13.12Strom_Ctake off the ,30,r
10:13.19Strom_Cyou dont need it in this context
10:13.53E-bolaHey guys, which ip-adapter would u recomend to for use with asterisk? To connect normal pstn phones to an asterisk sip/iax server
10:14.08pnlarssonpap2
10:14.08E-bolaDo you have any prefered models/brands?
10:14.23pnlarssonlinksys/sipura
10:14.44Strom_CE-bola: digium tdm400p/tdm2400p, digium iaxy, or linksys pap2t are all good products
10:15.25E-bolawhat about a spa1001?
10:15.39Strom_CChris-NB: E-bola you'll get a lower per-port price with the pap2
10:15.41Strom_Cer
10:15.50Strom_CE-bola: you'll get a lower per-port price with the pap2
10:16.00Chris-NBStrom_C, same thing : /
10:16.04Strom_CChris-NB: im not really sure what the problem is
10:16.14Strom_CChris-NB: are these boxes publically accessible from the outside?
10:16.20E-bolastr0m_C: pap2 is quote expensive
10:16.29Strom_CE-bola: its only $60 last I looked
10:16.32E-bolacosts double the amount of a spa1001, and i only need 1 port
10:16.43E-bolastrom_C: mmm over 100$ here
10:16.51Strom_Cwhere's "here"?
10:17.03E-boladenmark
10:17.07Strom_Cah ok
10:17.12Chris-NBStrom_C, nop. they are within my lan. connected via the same switch
10:17.14Strom_Csure, get the spa1001
10:17.34smlcoffeewe have used the grandstream ht386 for our analog phones and have no problem
10:17.36Strom_CChris-NB: is there any way you can forward port 4569 to one of the boxes?  i want to see if I can dial it
10:17.51smlcoffeewe cant get the grandstream 2ooo to work
10:18.09Strom_Cgrandstream stuff is kind of junky quality
10:18.29E-bolaWe gonna try out the linksys spa922 as ip-phone
10:18.30Chris-NBStrom_C, sry, thats not possible. the firewall infront of me is blocking it
10:18.36E-bolado anybody havce comments/experiences with those?
10:18.49Strom_CE-bola: i have an spa942
10:18.55Strom_CE-bola: it's not a bad phone
10:19.16Strom_CE-bola: there are certainly better phones out there, but the spa942 I like for a cheaper phone
10:19.23Strom_CChris-NB: you cant open the port?
10:19.45Chris-NBStrom_C, nop, it's not my firewall : /
10:19.46E-bolastrom_C: i tihnk the 922 is an even me basic/cheap model
10:19.59Chris-NBStrom_C, here is the iax debug output of the calling box: http://pastebin.ca/91156
10:20.06Chris-NBStrom_C, with two different call statements
10:20.24Strom_CE-bola: same phone sans line keys
10:20.24*** join/#asterisk FlatFoot (n=simon@80.88.192.113)
10:20.37E-bolaok
10:22.06heffernanhm, if i do:
10:22.06heffernanexten => s,1,Set(hidden=${DB(HIDENUM/${CALLERIDNUM})})
10:22.06heffernanexten => s,n,GotoIf(${hidden}?hidden:nohidden)
10:22.06heffernanexten => s,n(hidden),SIPAddHeader(Remote-Party-ID: <sip:${CALLERIDNUM}@area7.appsvrslip11.prigw.com>\;party=calling\;privacy=full\;screen=yes)
10:22.10heffernanexten => s,n(nohidden),SIPAddHeader(Remote-Party-ID: <sip:${CALLERIDNUM}@area7.appsvrslip11.prigw.com>\;party=calling\;privacy=off\;screen=no)
10:22.13heffernanups, maybe i should have pastebinned.
10:22.17Strom_Cheffernan: PASTEBIN
10:22.21heffernansorry, sorry.
10:22.25Strom_Cgrrrrrr
10:22.32Zeeekfl00d
10:22.43*** part/#asterisk smlcoffee (n=kvirc@adsl-69-106-203-69.dsl.irvnca.pacbell.net)
10:22.47heffernananyway, if i do that, it will do both Sipaddheaders. why?
10:23.02heffernani only want it to do one of them, depending on the hidden variable.
10:23.06Strom_Cheffernan: because it's falling through the dialplan priorities
10:23.38heffernani thought it would only do the one corresponding to the value.
10:23.42E-bolahanks for the input strom_c
10:23.48Strom_Cheffernan: why not have a label after the second sipaddheader statement that the first one does a goto() to
10:23.50heffernan(hidden) for hidden=1
10:23.52Strom_Cheffernan: no
10:24.03heffernanand (nohidden for hidden=""
10:24.03Strom_Cit goes to that priority and then continues execution
10:24.20heffernanhow can i achieve what i want?
10:24.27Strom_Ci just told you
10:24.32heffernanah, there.
10:24.39Strom_Cyay reading
10:24.41heffernanthanks.
10:24.57heffernanhow do i add a label?
10:25.13Strom_Cthe same way you added the other two labels
10:25.20heffernanoh, okay.
10:25.21heffernanthanks.
10:27.38heffernanStrom_C: so like this? http://pastebin.ca/91164
10:28.09Strom_Cwhy do you have cont,1?
10:28.17Strom_Ca priority cant have a priority
10:28.48Strom_Cand you shouldnt have two priorities labelled hidden
10:29.01heffernanjust trying to read the docs on voip-info.
10:29.03Zeeekas long as one is always hidden...
10:29.07Strom_Cexten => s,n(hidden),addheader
10:29.14Strom_Cexten => s,n,goto
10:29.20Strom_Cexten => s,n(nothidden),addheader
10:29.27Strom_Cexten => s,n(cont),whatever
10:29.44Strom_Cotherwise the goto statement that goes to hidden will get confused
10:29.46heffernangoto(cont) you mean.
10:29.48heffernan?
10:29.55Strom_Cheffernan: its pseudocode
10:30.05heffernananyway, looks like it worked now.
10:30.08heffernanthanks a bunch Strom_C.
10:30.15Strom_Canytime
10:30.39ZeeekStrom_C now that you solved that, how abput working on world peace?
10:30.56Strom_Cthat one is easy
10:31.05Zeeekkill all humans?
10:31.09Strom_Ckill everyone off
10:31.11Strom_Cyes, exactly
10:31.21Zeeekyeah, ya see, great minds and all that :)
10:31.33Zeeekbut...
10:31.58Zeeekwhat all humans were dead and an asterisk went into a deadlock condition just as several other pbx were calling it?
10:32.27Zeeeka) if there were no answer() would it ring?
10:32.40Zeeekb) if there were no 'r', would it answer ?
10:32.43Strom_Cthis is like philosophy class gone horribly horribly wrong
10:33.17Zeeekhow many deadlocked sip channels in a martian asterisk install?
10:33.27Zeeeksee it's way too hot to work today
10:33.49Zeeekor even be serious
10:33.55Chris-NBStrom_C, I know what the problem is : /
10:34.04Strom_CChris-NB: what is it
10:34.10ZeeekI recommend Crucial for buying memory. They really did a great job
10:34.18*** join/#asterisk johnnyb (n=jonathan@207.155.33.225)
10:34.18Chris-NBStrom_C, 1. it was a type *grrrrrr 2. is it possible/not possible to #include another file into iax.conf ?
10:34.44Chris-NBStrom_C, I generate a file out of a database and include it into iax.conf (did it successfully with sip.conf)
10:34.56Strom_CChris-NB: I don't understand your first comment
10:35.06Chris-NBStrom_C, #include "path/to/file.conf" <-- that correct ?
10:35.24Chris-NBStrom_C, *grrr again a typo!! I misstyped the path to the file
10:35.43Strom_CChris-NB: wait wait wait, you're doing this with includes?
10:35.56Chris-NBStrom_C, jep. thats not good ?
10:36.13Zeeekit's ok if you reload after changes
10:36.29Strom_Ci don't think you can't do includes with iax.conf
10:36.30Chris-NBZeeek, i'm doing that
10:36.39Chris-NBStrom_C, I'm not sure
10:36.40ZeeekStrom_C why not?
10:36.47Strom_C*shrug* just a hunch
10:37.02Chris-NBStrom_C, It works within sip.conf. just assumed it works in iax. too?
10:37.04Zeeekeasy enough to try Chris-NB
10:37.15Zeeeklet us know, I'd be curious
10:37.17Strom_CChris-NB: try this
10:37.28Strom_CChris-NB: type iax2 show peers at the CLI
10:37.36Strom_Cdo the included sections show up?
10:37.44Chris-NBjep, nothing shows up
10:37.53Strom_Cwell, then that settles it :)
10:38.11ZeeekI object strenuously
10:38.12Strom_Cno includes!
10:38.38Strom_CZeeek: it's half past three, so I'm looking for the simplest solution :)
10:40.20Chris-NBwait. it works : D
10:40.32Chris-NBthink I've to go back to school and learn to type
10:40.33Chris-NB*grml
10:40.39Strom_Cwhat was the typo?
10:40.42Zeeek<PROTECTED>
10:40.48Zeeekit reads the file
10:41.03*** join/#asterisk _MDC_ (n=marcus@c-6efde255.06-72-6c6b7013.cust.bredbandsbolaget.se)
10:41.04Strom_CChris-NB: what was the typo?
10:41.28Chris-NB*damned !!! #inclue
10:41.33Chris-NBI'm going for a break
10:41.35Strom_Coh
10:41.44Strom_Cyeah, well if you fuck up the include then it won't be included
10:42.17Chris-NBStrom_C, anyway, thanks for the support!
10:42.33Chris-NBnext time i read the section, and again, and again, then I ask you ; )
10:43.13Zeeekit would appear that the chan_iax2 reads the included file but doesn't parse it into the peers
10:43.31ZeeekI can't imagine why not
10:44.43_MDC_Hi all, i'm trying to get asterisk to connect to a h323 gatekeeper, but I can't get it to regsiter my h323 login, is it the alias setting in oh323.conf that sets ther login?
10:45.11Strom_Ch323: the h stands for headache
10:46.00_MDC_Strom_C, hehe, but I have no other options unfortunatly
10:46.22Strom_C_MDC_: I have no experience with h323
10:46.56Dr-Linux|workanyone knows any good mp3 musiconhold link? or free prompt messages site?
10:47.23Strom_CDr-Linux|work: I'll make recordings for cheap!
10:48.15Dr-Linux|workStrom_C: we also use our own recordings, even we have an complete IVR team
10:48.23Dr-Linux|workbut i need some personally
10:49.09*** join/#asterisk Goni (n=blah@217.17.247.70)
10:49.18Gonihello friends :)
10:49.56GoniI am facing a problem with  my ata-188 .. it was authenticating fine earlier but suddenly it not authentication anymore. Giving me SIP/2.0 401 Unauthorized all the times
10:50.03Zeeekjoin #myspace
10:50.11Strom_Chahahahahahahahahah
10:50.15*** join/#asterisk Modcuts (n=bob@lan.proporta.com)
10:50.22Strom_CGoni: what did you change
10:50.39GoniI think I just reset the ata nothing else
10:50.49Strom_CGoni: are the credentials correct in the ata?
10:50.53Goni7940 connects fine, but 188 .. it keeps on saying unauthorized
10:51.05Goniyes, i reset .. re-created accounts many times
10:52.39Goniany idea?
10:53.03Strom_C*shrug*
10:53.13*** part/#asterisk tparcina (n=tparcina@lns02-1506.dsl.iskon.hr)
10:53.34Goniyea, me too .. trying to get it fixed for like 2 days :S
10:53.46Strom_Ctry a sip debug
10:53.54Strom_Csee if the device is passing the correct credentials
10:54.46GoniJul 18 16:32:54 VERBOSE[2042] logger.c: Transmitting (NAT) to 203.81.198.157:2020:
10:54.46GoniSIP/2.0 401 Unauthorized
10:54.46GoniVia: SIP/2.0/UDP 192.168.1.100:2020;received=203.81.198.157
10:55.02Gonicredentials seems to be ok
10:55.14GoniFrom: <sip:8449347@sip.digitallinx.com;user=phone>;tag=2735791333
10:55.20GoniTo: <sip:8449347@sip.digitallinx.com;user=phone>;tag=as0e31350c
10:55.24Goniis this ok?
10:55.29Goniuser=phone ?
10:56.07Gonisame acoun its working fine with x-lite
10:56.45ZeeekGoni is the Grandstream product?
10:56.58Zeeeks/the/this a/
10:59.01Gonino, this is Cisco ata-188
10:59.08Gonirunning trixbox
10:59.31Strom_Coh good god, not trixbox
10:59.38GoniAsterisk
11:01.55Bert-erf
11:01.58Bert-trixbox :)
11:02.13Strom_Ci mean, seriously, it's right there in the topic
11:03.02mitchelocStrom_C, i don't know, the topic isn't exactly very clear, also on some irc clients it's not easy to read....
11:03.19Gonijust trying to get help ":)
11:03.42Strom_Cmitcheloc: when you join the channel, the topic is printed out in full right there in the client
11:04.33mitchelocStrom_C, yes, but it's just not, easy to read you know? i'm thinking big red letters would do the trick ;)
11:04.44mitchelocStrom_C, or better yet, some ascii art
11:05.07Strom_Cmitcheloc: no no, we need a sound file that goes "ATTENCION!  ATTENCION!  29453 29453 14452 14452..."
11:05.25*** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin)
11:05.36*** join/#asterisk DarKnesS_WolF (n=wolf@212.103.170.135)
11:05.50Zeeekoh, there's a topic?
11:06.06Strom_Cyes
11:06.06Zeeekjoin #pr0n
11:06.07mitchelocStrom_C, you just jinxed the entire channel...thanks
11:06.18Strom_C?
11:06.39_MDC_When doing Dial(), does asterisk do the call self or does it redirect the client to the new destination?
11:07.10Strom_C_MDC_: huh?
11:07.28mitchelocStrom_C, whoops, i thought you meant "4 8 15 16 23 42" for some reason
11:07.37mitchelocoops... now i jinxed it, great
11:08.24_MDC_well, I'm trying to get asterisk to act as a proxy between SIP and H323, so when I call SIP/555 asterisk would call OH323/*externalnumber* with the registred gatekeeper
11:09.32Strom_C_MDC_: that doesnt explain your question though
11:09.47mitchelocStrom_C, that's a lost reference for you, in case you don't follow the show
11:10.02Strom_Cwhat show
11:10.08mitchelocLost
11:10.18Strom_Chuh?
11:10.19mitchelocerrr -- http://thelostnumbers.blogspot.com/'
11:11.08mitchelocsupposedly they are jinxed, and bad things happen to the people around the person that knows them
11:11.17frenzyany php gurus around ?
11:11.21mitchelocbut to the person that knows them really good things happen...just due to bad circumstances
11:11.25mitchelocfrenzy, #php?
11:11.26Zeeekbe sure to watch http://GoodNightBurbank.com
11:11.28Strom_Cwhat the hell are you going on about, mitcheloc
11:11.53mitchelocStrom_C, nada, it's late here, i'm ranting, neveeerrrrmind ;)
11:12.12Zeeekanyone here use VoicePulse Connect?
11:12.23Strom_Cwhat's this lost thing you're talking about?
11:12.31Zeeeklost virginity
11:12.36Zeeeknot of this channel
11:12.54Chris-NBStrom_C, with the right words, the incluDe works fine : D can call a phone on box B from a phone on box A and the other side too : D
11:13.05frenzyI have a PHP statement...  $HD_Form -> FieldEditElement ('name, type, name-one'); but it wont work becuase of the hiphen... what is the correct syntax ?
11:13.30Strom_Cfrenzy: I think you are in the completely wrong channel
11:13.34Strom_Cthis is #asterisk
11:13.44frenzylol..
11:13.48_MDC_Strom_C: hmm, I'm doing the Dial(OH323/number) in asterisk, that call would go via the h323 gatekeeper via asterisk or will the client connect to that gateway directly?
11:13.57frenzyI'm doing some asterisk intergration..
11:14.09frenzyoh well will try #php
11:14.22Strom_C_MDC_: i don't know, can h323 do reinvites?
11:14.37*** join/#asterisk appelza (n=pieter@dsl-146-246-160.telkomadsl.co.za)
11:14.47appelzawhat is the filename of the default on hold music called?
11:15.11Strom_Cappelza: there are three of them
11:15.23Strom_Cappelza: and they live in /var/lib/asterisk/mohmp3
11:16.07appelzaOk, and when I add a new one, does it have to be owned by asterisk?
11:16.15_MDC_Strom_C; no idea what that is, ok this is what the situation looks like; the h323 gatekeeper have a access to PST and my asterisk will be acting as a SIP server, that will forward some numbers to that gatekeeper, with me?
11:16.50Strom_C_MDC_: if the asterisk box is converting between different signaling protocols, then it must stay in the media path
11:17.32_MDC_Strom_C, the problem is that the client is behind NAT and h323 does do well behind that, so therefore I'm using asterisk on the same net as the gatekeeper and connect to asterisk via SIP
11:17.49_MDC_Strom_C, what's a media path?
11:18.24Strom_C"IP Telephony"
11:18.35_MDC_Strom_C, sorry about that silly question...
11:18.35ZeeekBeginning asterisk docs:
11:18.35Zeeekhttp://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html
11:18.35Zeeekhttp://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html
11:18.35Zeeekhttp://www.automated.it/guidetoasterisk.htm
11:18.35Zeeekhttp://www.asteriskdocs.org
11:18.42mitcheloc~book
11:18.45jbotit has been said that book is a book called  Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
11:18.45Strom_C~docs
11:18.47jbotextra, extra, read all about it, docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html
11:18.53Zeeek~flood
11:18.54jboti heard flood is spewing loads of output into a channel; *very* rude in most channels and often grounds for banning.  If you want to show a lot of output to someone, ask them to join you in #flood and paste the output there.
11:18.57mitchelocheh, you think thats enough for the poor guy?
11:19.18Strom_Cno
11:19.18_MDC_yes...
11:19.29MrChimpyfloody jbot :)
11:19.36Strom_C:)
11:19.42_MDC_will come back when i've grow up
11:20.15Strom_C_MDC_: i was half-kidding
11:20.24*** join/#asterisk MatsK (n=mats@141.221.181.62.in-addr.dgcsystems.net)
11:21.03*** join/#asterisk moodperson (n=moodpers@ss13.lb4.ltk.com.ua)
11:21.05*** join/#asterisk [Airwolf] (n=airwolf@dsl51B67BC5.pool.t-online.hu)
11:21.11moodpersonhello people
11:21.22Strom_Cpeople?!  where?!
11:21.35Zeeekonly asterbots
11:21.46moodperson:)
11:21.53Zeeekwith kind suggestions for light reading
11:22.18Strom_Chey, ive got a two-inch-thick book on DSL that I consider light reading :)
11:22.19moodpersonany body help about symbian voip clients for smartphone ?
11:22.48_MDC_Strom_C, hehe, np
11:29.48CMikeHI all
11:29.58Strom_Chello
11:30.10moodpersonhi
11:30.16CMikehiyas
11:31.11Strom_Chookers!
11:35.30*** join/#asterisk RoyK (n=roy@236.84-48-82.nextgentel.com)
11:37.33CMikeanyone know why I cant set the callerid string ?  I set the string, but the outgoing call doesn't show it in the sipheaders.
11:37.38*** join/#asterisk RoyK (n=roy@236.84-48-82.nextgentel.com)
11:37.42CMikethe string looks ok in the cdr though
11:37.54*** join/#asterisk zotz (n=zotz@24.244.133.115)
11:38.11*** join/#asterisk effectiveape (n=nick@82.153.22.16)
11:38.12Strom_CCMike: what version of asterisk
11:38.17Strom_CCMike: what dialplan string
11:38.25Strom_CCMike: what SIP provider
11:38.39CMikezaptel card.
11:39.03CMikeAsterisk 1.2.9.1
11:39.05effectiveapeany bt isdn users?
11:39.08Zeeekzaptelcard + sipheaders...
11:39.17Strom_CCMike: what?
11:39.25Strom_CCMike: what the hell does zaptel have to do with sip? :)
11:39.34CMikezaptel = no sip provider :)
11:39.45CMikemaybe I was a bit unclear :)
11:39.55*** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com)
11:39.58Strom_CCMike: ok, so you're dialing out over a zaptel card?
11:40.12ZeeekCMike give the whole context
11:40.23CMikein the sip.conf I use callerid= 850122800   but the sipheaders show the username as callerid
11:40.53Strom_Cput the number in angle brackets
11:40.55*** join/#asterisk [Airwolf] (n=airwolf@dsl51B67BC5.pool.t-online.hu)
11:41.05Strom_Ccallerid="Name"<3115552368?
11:41.06Strom_Cer
11:41.08Strom_Ccallerid="Name"<3115552368>
11:41.12CMiketried that...
11:41.32ZeeekCMike post the EXACT line in sip.conf, just that line pls
11:41.33CMikethe cdr shows the right callerid in the src-field
11:41.33Strom_Cwhat are you dialing, and where is caller ID not showing up as expected?
11:41.51CMikecallerid = "test <850122850>"
11:41.55Strom_CNO
11:42.00Strom_Clook at my example
11:42.01effectiveapeor does anyone know who might so i can watch out for them
11:42.02Strom_Ccallerid="Name"<3115552368>
11:42.07CMikehold
11:42.17Zeeekspaces are bad in something=somethingelse
11:44.11*** join/#asterisk Greek-Boy (n=grb@193.220.93.162)
11:44.24Zeeekcallerid=Joe Blow <2002>
11:46.20CMikenow the line says: callerid="test"<850122850>
11:46.30Strom_Cgood
11:46.45Strom_CCMike: what are you dialing out over, and where is caller id not showing up as expected?
11:46.58CMikeand the outgoing invite = From: 0850122800 <sip:0850122800@
11:47.26Strom_CCMike: please answer my question
11:47.44Strom_CCMike: what is the final termination point of the call?
11:47.47ZeeekStrom_C it's often been said that quotes aren't good
11:47.48CMikeI'm dialing out over a ZAP channel (E1)
11:48.12CMikeand I have to put the outgoing callerid in the format 850122850 (without the 0 at the start)
11:48.45CMikeI dont want users to be able to set their own caller id. eg. I have to force the callerid before I dial out on the zapchannel
11:48.52CMikeif you understand how I mean..
11:49.18Strom_Cso you're forcing callerid based on the station placing the call?
11:49.23CMikesince I have about 50000 DIDs... I have to be able to put the callerid in the database for each sip client.
11:49.29CMikeyep .
11:49.44Zeeekdo it with set
11:50.01CMikeI guess I have to do that.. hm ..
11:50.29Strom_CCMike: so ok, now that you have the callerid string set correctly in sip.conf, does the callerid show up correctly on the called party's telephone set?
11:50.48CMikeI'm also running a very old version of asterisk, and there I could put the callerid field to any value, and that was the value the was used for outgoing calls (without have to use the set variable)
11:51.32CMikehold. lemme verify the sipheaders against the old asterisk
11:51.40effectiveapeand (i wonder if anyone will answer ;)) does anyone know why this might happen? http://pastebin.ca/91233 (and should i be worried?)
11:51.45Strom_CCMike: why are you worrying about sip headers
11:52.00Strom_CCMike: you're wasting your time with sip headers
11:52.20Strom_CCMike: if you're setting the callerid on asterisk itself, the sip headers from the phone are obviously not going to reflect that
11:52.34Zeeekeffectiveape does it work?
11:52.35CMikeI need the sipheaders to be correct, since some calls are routed to a sip -gw ... and not through my zaptel
11:52.55CMikeand that sip-gw uses the sipheaders for the callerid
11:53.11Strom_CCMike: lets debug one thing at a time, shall we?
11:53.20effectiveapeWell other ththe problem i'm here for (which i don't know if it's related or not) seems to.
11:53.25CMikeeg. for an anonymous call i use "Anonymous<850122850>"
11:53.27effectiveapeI can call in ok anyway
11:53.34effectiveapejust the calling out which is the problem
11:53.44Zeeekeffectiveape state your case
11:53.59Zeeektyping a line every few minutes makes it hard to see what the trouble is
11:54.19effectiveapeSure...
11:54.21effectiveapeAnyway....
11:54.51CMikethis is how I need the number to look like:  From: "134610120" <sip:134610120@
11:55.12Strom_CCMike: do you want me to help you debug the callerid problem, or not?
11:55.13CMikeand in that case the callerid field in the db looks like 134610120
11:55.35CMikestrom: sure.. maybe I misunderstood you,
11:55.46effectiveapeI'm trying to call out (over isdn2e bri using quadBri) my mobile number (07970xxxxxx) which comes up in the cli correctly. However all i get is the BT automated woman telling me the number isn't recognised
11:55.53Strom_Clet me make sure I understand the route your call is taking
11:56.13Zeeekeffectiveape looking at CLI what does the dial say?
11:56.19Strom_CSIP phone -> Asterisk -> E1 PRI -> PSTN -> Telephone set
11:56.44CMikein one case yep .. and also: SIP phone -> Asterisk -> SIP gw -> PSTN
11:57.01CMikeand the latest example need the sipheaders to match the caller id..
11:57.03Strom_CCMike: ok, well lets debug the first
11:57.05CMikeI think at least.
11:57.07CMikeok..
11:57.15Strom_CENOUGH WITH THE DAMNED SIP HEADERS ALREADY
11:57.20Zeeekheh
11:57.22Strom_Cone thing at a time
11:57.23Strom_Cplease
11:57.29CMikesure sure.. sorry :)
11:57.50effectiveapeExecuting Dial("SIP/11-94a5", "Zap/g1/07970xxxxxx") in new stack
11:58.03Strom_Cso now, with the callerid= string correct in sip,conf, does callerid show up correctly on a PSTN phone?
11:58.18effectiveape<PROTECTED>
11:58.19effectiveape<PROTECTED>
11:58.19CMikeno it doesn't
11:58.20Zeeekeffectiveape and one assumes the number shown to you is the one you expect to reach?
11:58.25effectiveapeyeah
11:58.30Strom_CCMike: it's a PRI, correct?
11:58.33effectiveapemuppet ;)
11:58.38effectiveapecall me if you like ;)
11:58.53Zeeekand g1 works for other calls?
11:59.13effectiveapethat's the only one i've actually tried at the moment.
11:59.20CMikeE1, and and the format of the callerid I send to the PRI must be in the format 850122850  (the username of the sip-client is 0850122850)
11:59.24effectiveapeSince that the only line i have available to call into
11:59.34CMikeso if I understand correctly I must remove the 0 in the callerid
11:59.41Zeeekcall infromation or BT or someone
11:59.47effectiveapeThing is that it's obviously getting futher than asterisk to bt
12:00.32Zeeekeffectiveape backing up, it's obvious that you're tyring to set a bunck of values the zaptel doesn't like
12:00.49*** join/#asterisk ariel_ (n=Ariel@70-46-87-158.ftl.fdn.com)
12:01.01Strom_CCMike: you're only giving me little bits of information
12:01.03effectiveapeyeah i've only just seen that though - wasn't complaining before.
12:01.10Strom_CCMike: I fail to see the total picture here
12:01.15CMikeI thought that useing the callerid string in the sip.conf resulted in the callerid on the ZAPchannel (without setting the callerid before the dial command)
12:01.16*** join/#asterisk esculapio__ (n=ESCulapi@200.88.44.66)
12:01.19effectiveapethose values were setup from jugnhanns setup tools
12:01.30CMiketo shorten it down:
12:01.30Strom_CCMike: pastebin a sip.conf entry and the zaptel dialout extension
12:01.41Zeeekeffectiveape well zaptel isn't pleased wit them :)
12:01.52CMikek .. hold
12:02.36*** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar)
12:03.05*** join/#asterisk hwt (n=hwt@curb.thorkildssen.com)
12:03.26effectiveapeyeah i guess but as i say - seems to be working
12:05.17CMikehm.. what now.. did pastebin just quit working?
12:06.51*** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net)
12:08.03*** join/#asterisk Strom_C (n=strom@gateway.digium.com)
12:08.08Strom_Cbloody hell
12:08.17effectiveapeah that error was just me
12:08.28CMike:)
12:08.30effectiveapeinsmod'ed 1q instead of 1r from bristuff
12:12.51CMikedoes pastebin work for u guys ?
12:13.03Strom_Cpastebin.ca
12:13.52CMikehttp://pastebin.ca/91256
12:14.22CMikeshouldn't the callerid in the sip.conf be the callerid that is used for the peer on the outgoing zap?
12:14.44effectiveapei might just ring bt
12:14.59CMikeor did I miss something.. ?
12:15.31Strom_CCMike: perhaps the telco doesnt like the callerid you're sending
12:16.22CMikesomething like that.. I think I'm sending 0850122850
12:16.33CMikelemme se of pri debug shows what I', sending
12:17.27CMikehm
12:17.35CMikewait a minute.. this looks right:  Presentation: Presentation permitted, user number not screened (0) '850122850'
12:18.00CMikeBRB.. Have to call the telco guys.
12:18.04CMikethnx ...
12:18.34Strom_Csee, i told you not to go balls-crazy over the sip headers
12:18.58CMike:)
12:19.12effectiveapejust got the bri debug output - does anything in here look kooky? http://pastebin.ca/91260
12:22.23*** join/#asterisk clive- (n=pirch@dsl-145-36-132.telkomadsl.co.za)
12:22.27MrChimpyballs crazy sounds like some ron jeremy porno
12:23.08|oranjia|has anyone used asterfax?
12:23.31MrChimpypresumably, yes.
12:23.58MrChimpyI clearly sorted that query out for him then.
12:25.32MrChimpyhmm. my now-fastagi app can start 240 sessions in 0.2s, whereas AGI version fell over in heap at 60 cos it was starting a perl for each line
12:25.43*** join/#asterisk PakiPenguin_ (n=uppal@linuxpakistan/admin/pakipenguin)
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12:28.46clive-mrchimpy , how different is your fastagi script to your agi script ?
12:29.51*** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br)
12:29.59*** join/#asterisk Vec (n=Vector@dsl-146-119-118.telkomadsl.co.za)
12:30.31clive-howzit vec
12:31.00MrChimpydifferent in that it's a tcpip server
12:31.11MrChimpyI/O is sockets
12:31.21MrChimpyarguments are passed differently too
12:31.34MrChimpyand you obviously don't have to deal with the HUP issue
12:31.46clive-is this for astcc?
12:31.54MrChimpyastcc?
12:32.15clive-calling card..uses HUP
12:32.50MrChimpymost DeadAGI apps should have a HUP handler if they want to do any cleanup when the caller hangs up mid execution
12:32.58clive-so fastagi can scale much more
12:33.24MrChimpyyep, should've used fastagi to start with but didn't know it existed
12:34.37clive-does fastagi end when the call ends, or does it continue like deadagi ?
12:37.18pdtmobileclive-: when the call ends the socket disconnects, so you can just (depending on the language) handle that happening and close down that session properly
12:37.44*** join/#asterisk pnlarsson (n=niklas@c83-248-2-120.bredband.comhem.se)
12:37.50pdtmobileaka, do any clean up your script needs
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12:42.04znoGhah, cool, didn't know you could have the "hotline" functionality in a PAP2
12:42.13znoGthat means no more dialplan configuration in the PAP2 !
12:42.36*** join/#asterisk gandhijee (n=gandhije@mail.win-ent.com)
12:42.46*** join/#asterisk andrejkw (n=andrejkw@c-71-57-143-216.hsd1.fl.comcast.net)
12:42.52andrejkwHey guys. I have a little problem. My provider requires me to make the trunk name of my SIP connection "i2telecom.com". Unfortunately, this name also becomes the identifier for the connection. Now, when I want to dial through it Asterisk think I am trying to dial through the domain "i2telecom.com" and not the actual connection. Is there anything I can do?
12:43.41*** join/#asterisk daysmen3_ (n=primus@host86-143-4-220.range86-143.btcentralplus.com)
12:44.44*** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com)
12:46.59znoGandrejkw: the trunk name *has* to be i2telecom.com? have you tried using [i2telecom] ?
12:47.11andrejkwYes I did.
12:47.19andrejkwIt won't Register if it's antyhing else.
12:49.41pdtmobileso you are saying Dial(SIP/i2telecom.com/5551234)
12:49.50andrejkwYes.
12:50.17andrejkwI tried both, Dial(SIP/i2telecom.com/5551234) and Dial(SIP/5551234@i2telecom.com).
12:52.13pdtmobileactually i'm not awake I meant the second one
12:52.51pdtmobilewhat host is your sip connection to?
12:53.02andrejkwThe host is i2telecom.com.
12:53.15*** join/#asterisk nortex (n=breeves@snapper.titanspecialties.com)
12:53.16clive-what is a "NMI"?
12:53.28PakiPenguin_<PROTECTED>
12:53.45*** join/#asterisk danalien (n=danalien@unaffiliated/danalien)
12:53.56clive-paki thanks, I am getting tons of NMI error messages when I try modprobe my wcfxo
12:53.57dpryooh oh, dazed and confused.
12:54.10clive-yes, dazed and confused..:((
12:54.25clive-any ideas on how to fix this ?
12:54.44dpryoIt happens when the hardware sends a message that the kernel doesn't understand, or handle.
12:54.56mutanyone here ever used any hyperlink/karlnet wifi router boards and/or AP PLuses or AP1000's??
12:55.12clive-that explains the 1000 times a minute it does this...
12:55.17clive-any way to fix this ?
12:56.09*** join/#asterisk NoRemorse (n=bah@210-84-10-183.dyn.iinet.net.au)
12:56.18clive-my alternative is to try the new version of ztdummy
12:56.29NoRemorsehi all, do I need the zaptel libraries and/or libpri if I dont have any physical E1 card in my box?
12:56.36dpryoclive-: Probably upgrade the kernel too?
12:57.12clive-dpyro its the new centos 4.3 ...  2.6.9-34.0.2.ELsmp
12:57.21dpryoheh
12:57.39dpryoonly 8 kernels old then :)
12:57.47NoRemorsewhat is ztdummy and do I need it ?
12:57.50pdtmobileNoRemorse: you need ztdummy for certain things to work
12:57.51dpryo2.6.17.6 is the latest.
12:57.52NoRemorsecos it is loaded
12:57.56clive-dpyro..lol...
12:58.13NoRemorsethanks pdtmobile, but do I need libpri or wont ztdummy compile w/o  it?
12:58.15clive-Noremorse you probably need ztdummy
12:58.32pdtmobileNoRemorse: you don't need libpri
12:58.41NoRemorseat some stage in the past I have had a digium E1 card in the system but it is gone now
12:58.41clive-dpyro, I think I am going to try the new ztdummy,....otherwise I am stuck
12:58.55clive-the old ztdummy sucked
12:59.17pdtmobileNoRemorse: it won't hurt to have it but you don't need it if you don't have a E1/T1 card
12:59.24*** join/#asterisk Greek-Boy (n=grb@193.220.93.162)
12:59.44NoRemorseok thanks
12:59.51*** join/#asterisk brad_mssw (n=brad@216.155.111.10)
12:59.58NoRemorsenow I gotta work out how to un install the zaptel module
13:00.34pdtmobilehuh?
13:00.43NoRemorsezaptel module is loaded
13:00.44pdtmobileztdummy is part of zaptel
13:00.57NoRemorseoh but will it compile w/o libpri?
13:01.07pdtmobileya
13:01.10moodpersonhm gayus where find "full" prise from www.sipnet.net ?
13:01.14NoRemorseok thanks
13:02.46*** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com)
13:05.50*** join/#asterisk tamp4x (n=tampon@64.201.13.51)
13:05.56tamp4xis it possbile to use the t1 card so that the machine is an IAD?
13:07.19pdtmobileYou want it to be Dulles Airport?
13:07.36CMike:)
13:08.34*** join/#asterisk Vorondil (n=jkiffmey@miro.voltaiccommerce.com)
13:09.46*** join/#asterisk |dennis| (i=dennis@200.32.215.84)
13:09.48Vorondilc:
13:10.07Vorondilbah, wrong window
13:10.19jbroomeblasphemy!
13:11.19CMikeprobably an symbolic link for /dev/null
13:11.23CMike-n
13:13.44Vorondilhehe
13:14.01VorondilCMike: that would make my job a little easier, now that i think about it
13:14.04Vorondil;)
13:14.29CMikeheh
13:17.27andrejkwSo, can anyone help please?
13:19.22effectiveapeanyone know what "1  Cause: Unallocated (unassigned) number (1), class = Normal Event (0) "
13:19.25effectiveapemeans?
13:20.21clive-ok,  new bie question ... will typing "svn update" bring me to the latest version ?
13:21.45effectiveapeit will bring the source to the latest version  as long as your not switched to a branch
13:21.45*** join/#asterisk saftsack (n=saftsack@p54A7E68B.dip.t-dialin.net)
13:22.21clive-effectiveape tahnks, ..how do I know if I am switched to a branch ?
13:22.36effectiveapeIf you don't know you're probably not ;)
13:23.34clive-effectiveape I think I may be because I first downlaoded with : svn checkout http://svn.digium.com/svn/zaptel/branches/1.2 zaptel-1.2.4
13:24.39effectiveapeaha ok. It's probably still being updated on the 1.2 branch anyway. (i don't know the * svn layout but i know svn)
13:24.48effectiveapeso doing an update you should be cool
13:25.20clive-I did and its gives me :
13:25.21clive-At revision 1242.
13:25.32clive-is revision 1242 the latest ?
13:25.38effectiveapewith no downloads?
13:26.06effectiveapemust be the latest on the branch you're on
13:26.35clive-seems like I am stuck in my branch...
13:26.54effectiveapeyeah but that's the current branch anyway so you should be cool
13:26.58*** join/#asterisk auralia (n=ehernand@207.71.51.162)
13:27.17effectiveapeunless you don't want to be on 1.2
13:27.19andrejkwOh come on, I am really desperate.
13:27.33effectiveapekinding. so am i! ;)
13:27.34andrejkwI've been sitting at this for 2 days now, and I can't figure it out.
13:27.34clive-mind you, I just found a way to look at the svn, and that is the latest revision, cool
13:27.49clive-whats it andre..maybe newbie me can help
13:28.00andrejkwHey guys. I have a little problem. My provider requires me to make the trunk name of my SIP connection "i2telecom.com". Unfortunately, this name also becomes the identifier for the connection. Now, when I want to dial through it Asterisk think I am trying to dial through the domain "i2telecom.com" and not the actual connection. Is there anything I can do?
13:28.22andrejkwI tried both, Dial(SIP/i2telecom.com/5551234) and Dial(SIP/5551234@i2telecom.com).
13:28.40andrejkwNeither one works.
13:28.51*** join/#asterisk DarKnesS_WolF (n=wolf@212.103.170.135)
13:30.06andrejkwNobody? :'(
13:31.41effectiveapeThere has to be some isdn people on here surely....
13:31.41moodpersonandrejkw: where your put sip resource ?
13:31.48clive-sorry andre no clue here...I am ure someone on this list can help, send out an email
13:31.58andrejkwmoodperson: what do you mean?
13:32.13moodpersonandrejkw: www.sipnet.net?
13:32.32moodpersonandrejkw: who sip provider ?
13:32.37andrejkwVoiceStick
13:32.48moodpersonplz home page
13:33.05andrejkwhttp://www.voicestick.com
13:33.18moodpersonok thx
13:34.09andrejkwHost is i2telecom.com, they require me to name the connection i2telecom.com, so it becomes [i2telecom.com]. And now I have no way to reffer to it without Asterisk thinking that I am trying to use the domain and not the connection.
13:35.29*** join/#asterisk heison (n=heison@ns.somanetworks.com)
13:36.10*** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com)
13:36.32heisondoes anyone know of a place where I can get a London number delivered via IAX?
13:38.36E-bolaDo anybody know a free sip client that works with asterisk and can call forward, for windows?
13:43.07jbalcombE-bola x-lite by eyebeam perhaps may work for you
13:43.22*** join/#asterisk Dovid (n=dovi5988@pool-71-250-2-14.nwrknj.east.verizon.net)
13:44.19Dovidcan anyone help me with polycom paging
13:44.20Dovid?
13:44.35E-bolajbalcomb: it dont work with call forward unless u pay
13:44.40jbalcombDovid you talking about auto-answer?
13:45.00*** join/#asterisk DarKnesS_WolF (n=wolf@212.103.170.135)
13:45.13*** join/#asterisk foRza (n=tMs@firewall.hikt.no)
13:45.14Dovidyes
13:45.38Dovidjbacomb: yes
13:45.51jbalcombDovid: have you seen the page on the wiki that specifically discusses auto-answer on the polycoms for paging?
13:45.56Dovidyes
13:46.05Dovidcant figure it out. its all startight forwars
13:46.11jbalcombwhat seems to be the trouble?
13:46.14Dovidbut still having issues. no clear enough
13:46.26Dovidi have to edit the image that goes on to the phone correct ?
13:48.16*** join/#asterisk lokkju (n=lokkju@unaffiliated/lokkju)
13:48.28*** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com)
13:48.28*** mode/#asterisk [+o anthm] by ChanServ
13:49.26Dovid?
13:50.40[TK]D-FenderDovid : yes.
13:51.56Dovidone sec, tryin the wiki again
13:53.24*** join/#asterisk trbldwine (n=trbldwin@adam.ur.northwestern.edu)
13:53.57*** join/#asterisk carl0s- (n=carl0s@compsup.demon.co.uk)
13:54.48*** join/#asterisk zotz (n=zotz@24.244.133.115)
13:54.55carl0s-ugh. how bizzare. I just rang my chick through my Cisco 7960 via Asterisk via the TDM400P, and I could hear my own voice coming back really loud and clear after a slight delay. It could have been a problem at her end, or it might be because I initiated the call with Speakerphone turned on the handset. What do ya think?
13:56.08Strom_Ccarl0s-: sounds like a far end echo problem
13:56.27Strom_Ccarl0s-: turn your echo canceller on :)
13:57.05[TK]D-FenderStrom_C : He can't cancel, he signeds a long term contract ;)
13:57.20Strom_Cheh
13:57.47carl0s-Strom_C: where?
13:57.57Strom_Ccarl0s-: in zapata.conf
14:01.55CMikespeaking of Cisco .. is there i working sip firmware for the 7971g ?
14:02.14carl0s-I just made another call and it was the same. Had to turn the volume right down. Could this be showing because I have had the line unplugged from the card all night? I only just plugged the line back in. Does some kind of line-normalisation happen only when the zap modules are loaded or something?
14:02.37Strom_Ccarl0s-: turn.  on.  echo.  cancel.
14:02.46carl0s-OK OK
14:02.56carl0s-but it was perfect yesterday. I'll check the conf now :D
14:03.04carl0s-;)
14:03.44carl0s-echocancel=yes
14:03.44carl0s-echocancelwhenbridged=no
14:03.44carl0s-echotraining=800
14:03.52carl0s-it's already on then!
14:05.00Strom_Cwell, try reloading the drivers then if you're convinced there's equalization that goes on
14:05.19carl0s-I'm not convinced of anything, was just wondered if that's how possibly what might happen.
14:05.21Strom_Cor try calling a different number
14:06.55*** join/#asterisk marv[work] (n=timr@64.89.118.139)
14:07.10carl0s-I called two different numbers. I've just restarted asterisk and zaptel, I'll see how it is now.
14:07.57*** part/#asterisk kmilitzer (n=km@office-gw.westend.com)
14:08.10heisonanyone know about toll free numbers in London?
14:09.45*** join/#asterisk DrMouse (n=noneof@62.164.176.20)
14:10.50carl0s-heison: what about them?
14:12.30DrMouseright, i have a problem. this is with a digium TDM400, I have 2 FXO modules and 2 FXS. ignoring the fxs, dialing out on either FXO i cant get the TX level to be around half. I am in the UK on BT and dont know of a milliwatt test tone number, so i am using the simple method of trying to get the VU to register about 50%.
14:12.55heisoncarl0s-: i'm looking for IAX delivered numbers in London... do you have someone to recommend?
14:13.08DrMouseeven at +20 gain the tx level is still too low, and people are complaining that the call is too quiet
14:13.25clive-does anyone know how to use ztdummy with the new rtc stuff in it ?
14:13.28heisoncarl0s-: rephrase, i'm looking for a London number delivered via IAX
14:13.44carl0s-heison: sorry, I can't help. I thought you were going to ask something like "do they start with 0800?"
14:14.04DrMouseyet it is fine with a std phone
14:14.16DrMousei am using a GXP-2000
14:14.23carl0s-heison: if it's freephone, it won't be a London number in particular will it, it'd just be a nationwide 0800 or 0500 number I should think.
14:14.24DrMouseany ideas?
14:14.28*** join/#asterisk Casterman (n=pierre@kasadsyed.net8.nerim.net)
14:14.43*** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1)
14:15.08clive-~seen tilghman
14:15.19jbotclive-: i haven't seen 'tilghman'
14:15.19jbalcombDrMouse
14:15.24jbalcombDrMouse: who is complaining the call is too quiet?
14:16.20DrMousejbalcomb - the ppl on the other end of the phone. and also on another site with similar setup
14:16.35effectiveapeany isdn junkies yet?
14:16.52*** join/#asterisk shadebob (n=chatzill@ll81-144-114-192-81.ll81.iam.net.ma)
14:16.54jbalcombDrMouse any connection with people on the calling end using speaker phone?
14:17.13DrMousejbalcomb - nope
14:17.19Strom_CDrMouse: where are you located
14:17.26Strom_CDrMouse: and who is your telephone company
14:17.49DrMouseStrom_C - Leeds, UK. Teleco is BT.
14:17.52carl0s-Strom_C: echo seems fine now I've reset the 7960 phone.
14:17.56jbalcombDrMouse: have you confirmed that raising the txgain has made any difference at all?
14:18.04effectiveapeoooh i'm in leeds too ;)
14:18.15Strom_CDrMouse: how many thousand feet long is your loop?
14:18.17shadebobhi, I try to write an AGI. I have to execute an application every 30s on a channel, during a communication. I don't known how I can send EXEC application during the dial... Someone can help me
14:18.40*** join/#asterisk cvv (n=cvv@212.8.35.34)
14:18.53cvvgood evening!
14:19.07Strom_Cevening?!  it's 7:20 in the morning! :)
14:19.48cvvat my clock I have 17:20 ;-)
14:19.52DrMousejbalcomb -  tx gain makes a difference, but when at an acceptable level to the callee it is 'scratchy' (clipping i think) and DTMF digits dont send properly
14:20.50Strom_CDrMouse: i think setting your txgain at 20dB is bonkers
14:21.04Strom_CDrMouse: how many thousand feet long is your loop?
14:21.18*** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net)
14:21.48DrMouseStrom_C - dunno, but we are about a mile from the exchange. i know +20 is bonkers, but even at that im only getting a peak of about 1/3 on ztmonitor
14:21.58cvvCan anyone talk me what I need for implamentation conference via VoiP?
14:22.18DrMouseanyone know if you can put a gain or AGC on the RTP stream?
14:22.19tamp4xis it possbile to use the t1 card so that the machine is an IAD?
14:22.25tamp4xis=as
14:22.59Strom_CDrMouse: what kind of telephone sets?
14:23.38DrMouseStrom_C - Grandstream GXP-2000's, shit just realised ive got a sipura thing somewhere i can test with, brb...
14:24.28shadebobCan I execute a Dial and immediately after, without wait for hangup, pass to next action in dialplan?
14:25.02[TK]D-Fendershadebob : Nothing happens DURING a dial.  What you'd need to do is take the channel # before you dial and have a background process poll for it constantly till its gone in order to do something like taht.
14:26.04DrMouseStrom_C - same with sipura (841 maybe? unsure what model)
14:26.15shadebobhi D-Fender. A background process can be an AGI?
14:26.22*** join/#asterisk iriga (n=adu@LSt-Amand-152-31-4-219.w82-127.abo.wanadoo.fr)
14:26.37effectiveapedoes noone use isdn?
14:26.39*** join/#asterisk _problem_ (n=lokesh_k@estrela.nortenet.pt)
14:26.51MikeJ[Laptop]yes.. lots of people use isdn
14:26.53shadebobD-Fender : because asterisk wait the end of AGI to pass to next action...
14:27.09Strom_CDrMouse: if you plug the line directly into an analog phone and make a call, does the caller think the line is too quiet?
14:27.15effectiveapesurely people must have come up with similar problems
14:27.20shadebobD-Fender : maybe it exist a tip to avoid the waiting of the AGI end?
14:27.26DrMouseit seams like asterisk is recieving a very low volume rtp stream from the phones
14:27.27carl0s-effectiveape: yeh lots of people. You can look at Junghanns BRIstuff if you want to use ISDN2e with a HFC-S cheapo PCI card, else it's ISDN30/PRI stuff.
14:27.46DrMouseStrom_C - no volume is fine from an analogue phone
14:27.50effectiveapei'm using a quadBri card at the moment.
14:28.00effectiveapeJust noone seems to be able to help me getting it working ;)
14:28.21Strom_CDrMouse: odd...id call digium support.  they're open now.
14:28.54znoGandrejkw: you'd think Asterisk would check first if a context by that name exists, else dial via DNS
14:29.03znoGandrejkw: that is, lookup host by DNS and connect directly
14:29.26DrMouseStrom_C - any chance you got the number?
14:29.26[TK]D-Fendershadebob : you need to initiate the process to be run in the background before the dial.
14:29.29Strom_Candrejkw: oh, please, just call the context something else and use username=
14:29.41Strom_CDrMouse: IAX2/guest@misery.digium.com/s
14:29.52znoGandrejkw: still, if you dial using Dial(SIP/i2telecom.com/5551234) it should still use the [i2telecom] info to dial out, IIRC
14:30.11shadebobD-Fender : create a new thread for exemple? I not a coding guru :s
14:31.03effectiveapeand bt don't seem to like using phones anymore. It's email or nothing for their ISDN support :/ Muppets
14:31.06andrejkw?
14:31.18Strom_CDrMouse: or +1 256 428 6000
14:31.19andrejkwHmm
14:31.21andrejkwAre you sure?
14:31.51*** join/#asterisk anthonyl (n=anthony@office.midphase.com)
14:33.02andrejkwThat means my problem must be somewhere else
14:33.21carl0s-effectiveape: well, I can't help as I don't have ISDN. I know there are some smart and helpful people on here but you have to be patient and wait for (a)the right people to show up and (b)them to have some spare energy :
14:33.23carl0s-)
14:33.34*** join/#asterisk Hmmhesays (n=Neg@24-117-135-28.cpe.cableone.net)
14:34.12effectiveapeyeah but it's know when to keep asking ;) - I've been on here for 24 hours now.
14:34.18carl0s-lol
14:34.34[TK]D-Fendershadebob : Something like that.  Call an "Exec" that will daemonize and poll * for the channel.
14:34.54effectiveapeThey probably heard i was on here and don't come on ;)
14:35.26shadebobD-Fender : thanks a lot. How I can daemonize an asterisk application?
14:35.53carl0s-just sent you a /msg effectiveape
14:36.01[TK]D-Fendershadebob : its not an * application, its one you'll be making yourself.
14:36.41*** join/#asterisk JohnJacob (n=dhorner@pool-71-127-102-43.aubnin.fios.verizon.net)
14:37.11shadebobProblem is I want to modify register of the FXS port (wctdm) and I cannot write on /dev/zap/xxx through an external application because asterisk use it :(
14:37.48*** join/#asterisk s0lid (n=s0lid@202.71.179.140)
14:37.49_MDC_hmm.. I get seg fault when doing exten => 889,1,Dial(OH323/mc:@10.1.210.24/0013111111), what could I be doing wrong?
14:39.44clyrrad1Hi Guys - I am having a really strange problem - all phones can dial each other and leave voice mail - only problem is the MWI is not comming on - howerver if each phone dials into Comedian Mail - it will tell them they have a new message - and idea what could cause this?
14:39.56*** join/#asterisk unixgeek (n=unixgeek@216-220-234-197.exploremaine.com)
14:42.14saftsackhi
14:42.23saftsackare there voip hardware telephones which can handle video calls?
14:43.47effectiveapethose '24' motorola's look tasty
14:44.28effectiveapehttp://broadband.motorola.com/consumers/products/ojo/
14:45.01Qwellsaftsack: several
14:45.12saftsackit looks great ;)
14:45.22saftsackwhat are the prices of those telephones? ^^
14:45.24effectiveapesip supported although i've never tried it
14:45.32Qwellsaftsack: ~$300
14:45.48saftsackif i use h 323 with asterisk it would be possible to do video calling with asterisk, or?
14:45.50clyrrad1Qwell - do you have any idea what could cause my WMI problem?  Been playing around with it for 2 days now :s
14:46.22carl0s-ugh. that echo is unbearable.
14:47.06carl0s-it's not exactly echo though, it's my own voice coming out of the earpeice.
14:47.13jbroomeugh. that echo is unbearable
14:47.14jbroomeugh. that echo is unbearable
14:47.21*** join/#asterisk ChrisDe3 (n=Chrisde3@port-87-234-141-161.dynamic.qsc.de)
14:47.32carl0s-:D
14:47.41jbroome:>
14:48.35ChrisDe3one question: If an absolutetimeout is set to 1 hour... and the call is still up in 1hour and 11 minutes... this is correct behaviour?
14:49.17*** join/#asterisk froguz (n=xxxxx@pc-95-155-104-200.cm.vtr.net)
14:50.09*** join/#asterisk lokkju (n=lokkju@unaffiliated/lokkju)
14:51.25*** join/#asterisk NoRemorse (n=bah@210-84-10-183.dyn.iinet.net.au)
14:51.42NoRemorsehi all, does this mean anything to anyone?! handle_request: Unknown SIP command 'SI16384P/2.0' from '202.161.21.211"
14:53.13DrMouseexcuse the language but #%$*! i switched from using GSM to PCMU and sunddenly, ppl can hear me fine!
14:53.23DrMouseunless its an intermittant thing
14:53.34*** join/#asterisk johnnyb (n=jonathan@adsl-38-9-196.tulsaconnect.com)
14:54.29DrMousethank you all for your help anyway.
14:55.00DrMouseat least i now have an extension for contacting digium support :)
14:55.01[TK]D-FenderNoRemorse : garbage packet.
14:55.36froguzi'm not good at english but i'll try to put it simple: i'm trying to make calls to the PSTN using a Micronet FXO gateway (in proxy mode), wich has the extension 1001 configured in line 1 so, when i dial 1001 it gives me PSTN dialtone, but then i dial the phone number and nothing happens, it continue giving dialtone. is this a DTMF detection problem?
14:55.45DrMouseafter all that, im off for a smoke. once again thanks for your help.
14:58.15andrejkwCan I have custom caller ID? Like when a known number calls the phone will show the name of the person?
14:58.25andrejkwAnd I can set the name in some configuration file?
14:58.35andrejkwAnything like that possible?
14:59.51*** join/#asterisk websae_ (n=websae@adsl-64-149-206-121.dsl.milwwi.sbcglobal.net)
15:01.36*** join/#asterisk carl0s- (n=carl0s@compsup.demon.co.uk)
15:02.24*** join/#asterisk DarKnesS_WolF (n=wolf@81.10.111.139)
15:03.06*** join/#asterisk Damin (n=damin@nucleus.nacs.net)
15:03.11*** join/#asterisk mfdutra (n=marlon@200.208.130.16)
15:03.26mfdutramy asterisk doesn't respect the absolute timeout
15:03.45*** join/#asterisk hohum (n=dcorbe@12.195.58.235)
15:04.25NoRemorseGarbage packet? it keeps happening, always the same
15:05.51*** join/#asterisk dlynes_ (n=dlynes@S0106001217014b92.vc.shawcable.net)
15:06.00mfdutrashow channel Zap/1-1
15:06.01mfdutraElapsed Time: 17h49m35s
15:06.01mfdutraTIMEOUT(absolute) = 120
15:06.31ChrisDe3yes I have the same problem
15:06.38*** join/#asterisk burnproof (n=jsharryp@210.213.242.145)
15:06.45ChrisDe3http://bugs.digium.com/view.php?id=7546
15:07.38*** join/#asterisk eKo1 (n=eKo1@190.4.7.90)
15:07.54*** join/#asterisk razu (n=razu@87-119-182-133.tll.elisa.ee)
15:08.03*** part/#asterisk websae_ (n=websae@adsl-64-149-206-121.dsl.milwwi.sbcglobal.net)
15:08.20*** join/#asterisk s0lid (n=s0lid@202.73.164.125)
15:08.54*** join/#asterisk dorphalsig (i=Dorphals@pcsp168-254.supercabletv.net.co)
15:09.02dorphalsigHey
15:10.42malverianAnyone had any issues with 1.2.10? I'd like to upgrade to it from 1.2.9.1 to fix some voicemail bugs and the CLI issues.
15:11.21dorphalsigI am running 1.2.7 but some zap channels will die ocassionally and wont come up unless I restart asterisk
15:11.29NoRemorsecant get DISA to work, it says unkown application
15:11.31dorphalsigwhen I call it sounds as if the extension was busy
15:11.49dorphalsigand when I pickup the phone I get nothing
15:12.36dorphalsigNoRemorse... there is a module you have lo load to get DISA working
15:12.41dorphalsigcheck the wiki for its name
15:12.45*** join/#asterisk mog (i=ejabberd@68.62.237.103)
15:12.45*** mode/#asterisk [+o mog] by ChanServ
15:12.47NoRemorseapp_disa.so?
15:12.55NoRemorsei'll put it in modules.conf ty
15:13.06dorphalsignp
15:13.17burnproofNoRemorse: check if you have actual app_disa.so on /usr/lib/asterisk/modules
15:13.33dorphalsigGood Point
15:13.59NoRemorseyeah its there
15:14.08NoRemorseand so was noload=
15:14.10NoRemorselol
15:14.12burnproofNoRemorse: show applications like disa ?
15:14.15burnproofwhat's the output
15:14.38burnproofon your CLI show applications like disa ? what's d output
15:15.37NoRemorsenah think I have to restart, reload doesnt fix it
15:15.45NoRemorseI'll wait till calls drop
15:16.31jbalcombAnyone know fairly well how long your data center will stay up on battery backup?
15:16.55*** join/#asterisk websae (n=websae@adsl-64-149-206-121.dsl.milwwi.sbcglobal.net)
15:16.55NoRemorse30 mins to 2 hours
15:17.00jbalcombI'm trying to decide how long ours should stay up. I'm think 15 minute minimum pushing for 30
15:17.14jbalcombNoRemorse: roughly how many servers do you have?
15:17.24NoRemorselong enough to ensure a generator startup :)
15:17.58NoRemorsewhat ups u got?
15:20.19andrejkwIs there anyw ay to have a custom caller ID name for every person that calls?
15:20.30dorphalsigI am running 1.2.7 but some zap channels will die ocassionally and wont come up unless I restart asterisk. when I call it sounds as if the extension was busy and when I pickup the phone I get nothing
15:20.55[TK]D-Fenderjbalcomb : I am in the planning phases for that here.  My take is 1 hour tops on the std servers (file, mail, etc), and 2 hours on phone (switching, etc).
15:21.43*** join/#asterisk DasTech (n=DasTech@c-67-176-28-65.hsd1.co.comcast.net)
15:21.54jbalcombNoRemorse: we have 3 triplite 2000s, 2 apc 2200s, and 3 apc 1400s.
15:21.58clyrrad1Hi Guys - I am having a really strange problem - all phones can dial each other and leave voice mail - only problem is the MWI is not comming on - howerver if each phone dials into Comedian Mail - it will tell them they have a new message - Asterisk also sends an email to the person saying someone has left a message - yet no MWI is lit up on the phones - I have tried the phones on another server and the MWI does wor
15:22.02DasTechGouten Morgen
15:22.09jbalcomb[TK]D-Fender : have you come up with anything yet?
15:22.12DasTechneed some input for dial logic
15:22.18[TK]D-Fenderclyrrad1 : pastebin your setup and describe your hardware
15:22.33[TK]D-Fenderjbalcomb : not model #'s offhand, but it was a BIG rackmount APC solution.
15:22.51[TK]D-Fenderjbalcomb : 10$K or so
15:22.53jbalcomb[TK]D-Fender: 1 UPS for each rack or just 1 UPS?
15:23.18jbalcomb[TK]D-Fender: that's nearly the price of a generator
15:23.25[TK]D-Fenderjbalcomb : Simpler intelligent UPS (networked), with massive battery.
15:23.33DasTechI am working on extension with web interface I want to put a dial button and have it dial me and then play back PLease hold while I connect your call then dial the other person
15:23.38clyrrad1The hardware is are GNET VP104s phones that have been tested on other servers - and have used alot of them - the server is remote / on a different network then the phones - they connect remotely - they are not having firewall issues as I have completely disabled the firewall to trouble shoot that
15:23.40[TK]D-Fenderits the remote start/;stop that also adds to the cost.
15:24.03DasTechhas anyone done this
15:24.08*** join/#asterisk salviadud (n=ralfalfa@dsl-201-128-132-150.prod-infinitum.com.mx)
15:24.12clyrrad1TKD-Fender can I PM you any config you want to seee?
15:24.17*** join/#asterisk SplasPood (n=jwb@206.252.198.101)
15:24.19salviadudpastebin
15:24.19[TK]D-Fenderclyrrad1 : pastebin yoru voicemail / SIP setup
15:24.53clyrrad1Voicemail is being left using a Macro and the system is recording and storing the messages becase they can be checked by dialing into comedian mail
15:25.01clyrrad1all voice mail stuff is working perfectly just no MWI
15:25.19DovidTK: i edited the polycom directory file but for some reason the phone wotn get ti
15:25.22Dovidor so it seems
15:25.26[TK]D-Fenderclyrrad1 : PASTEBIN.  I stop telling me everything is alright when its NOT WORKING.
15:25.43NoRemorsewhen I goto DISA it cerates a second CDR with the same uniqueid any way to kill the initial one?
15:25.45clyrrad1what would youl ike to see?  1 context and the Macro?
15:25.51dorphalsigI am running 1.2.7 but some zap channels will die ocassionally and wont come up unless I restart asterisk. when I call it sounds as if the extension was busy and when I pickup the phone I get nothing
15:25.53[TK]D-FenderDovid : that was remarkably unclear....
15:26.06[TK]D-Fenderclyrrad1 : sip.conf and voicemail.conf
15:26.14Dovidone sec
15:27.13mfdutraSet(TIMEOUT(absolute)=timeout) cannot have spaces around '='
15:27.16mfdutrashit bug
15:27.32mfdutrano warning, no error
15:28.07NoRemorsecant.. stay.. awake.... must.. sleep....
15:28.50*** join/#asterisk kindor (n=roy@office.open-ict.nl)
15:29.19clyrrad1Here is the pastebin http://pastebin.ca/91354
15:29.57DovidTK: i edited the MacID-directory.xml and when the phone boots i tell it to provision but whent he phone is on i dont see the numbers in the phone
15:31.03GerbilWrkAnyone know the keycombination to set the IP of a Linksys SPA1001?
15:32.17*** join/#asterisk enjay- (n=enjay@71.216.165.97)
15:32.21razuGerbilWrk : pap2 combination should work there ...
15:32.28enjay-morning
15:32.36clyrrad1[TK]D-Fender - did you get the pastebin?
15:32.37froguzi'm trying to make calls to the PSTN using a Micronet FXO gateway (in proxy mode), wich has the extension 1001 configured in line 1 so, when i dial 1001 it gives me PSTN dialtone, but then i dial the phone number and nothing happens, it continue giving dialtone. is this a DTMF detection problem?
15:32.53GerbilWrkrazu, pap2?
15:33.38*** part/#asterisk mfdutra (n=marlon@200.208.130.16)
15:35.11razuGerbilWrk : does ****111# work ?
15:35.18*** join/#asterisk Dibbler_ (n=Dibbler@dsl-217-155-254-174.zen.co.uk)
15:36.24GerbilWrkpossibly
15:36.28GerbilWrkit said to enter value
15:36.57razuGerbilWrk : then check out this manual ... there are some codes you need to know : http://www.freshtel.net/support/hardware/Linksys%20PAP2.pdf
15:37.36razuGerbilWrk : pap2, spa2100 and spa1001 ivr should use the same codes
15:37.37froguzGerbilWrk, ****111# your*desired*ip*number#
15:38.06*** join/#asterisk TrixVox (n=Spy007@c-69-248-121-104.hsd1.nj.comcast.net)
15:38.11froguzGerbilWrk do you read spanish?
15:38.32salviadudi read spanish
15:39.10froguzsalviadud, me too. i'm asking to GerbilWrk.
15:39.22clyrrad1[TK]D-Fender you still here?
15:39.29salviadudyeah, pinche cabron, lee español GerbilWrk
15:39.38froguzhahahaha
15:39.59salviadudwell, would someone be interested in listening to a prank call i made recently?
15:40.07salviadudi called a rehab clinic in florida
15:40.19salviadudpretending i was high on something
15:40.28clyrrad1lol
15:41.22Qwelllink? :P
15:41.28salviadudcould someone give me a name for this "upload shit" sites?
15:41.41Qwellgoogle video?
15:41.42salviadudi just need to upload it somewhere
15:41.49Qwelloh, audio
15:41.50salviaduddoes google video take mp3?
15:41.57froguzGerbilWrk: 100# say dhcp, 101# set dhcp, 110# say ip, 111# set ip, 120# say mask, 121# set mask, 130# say gateway, 131# set GW, 73738# reset
15:41.57Qwelldoubt it
15:42.55clyrrad1I am having a really strange problem - all phones can dial each other and leave voice mail - only problem is the MWI is not comming on - howerver if each phone dials into Comedian Mail - it will tell them they have a new message - Asterisk also sends an email to the person saying someone has left a message - yet no MWI is lit up on the phones - I have tried the phones on another server and the MWI does work - any id
15:43.21*** join/#asterisk boch (n=root@201.216.241.97)
15:43.33*** join/#asterisk dimitrich (n=dimitri@lns-bzn-49f-62-147-167-75.adsl.proxad.net)
15:45.51saftsackwill the future bring mass video telephony?
15:46.05anthonylsexxy ness
15:46.31salviadudhttp://d.turboupload.com/d/790589/donitadunes.mp3.html
15:46.59dorphalsigI am running 1.2.7 but some zap channels will die ocassionally and wont come up unless I restart asterisk. when I call it sounds as if the extension was busy and when I pickup the phone I get nothing
15:47.13eKo1saftsack: it will bring mass video telephony porno actually
15:47.30salviadudyeah, that damn porno
15:47.40saftsackthats right too ;) but do you think normal people will use video telephony?
15:47.47salviadudi just don't get it... where do they get their money from?
15:47.56salviadudi'm not buying pr0n...
15:48.03effectiveapei can't see people using it
15:48.16salviadudi'm boycotting pr0n, and it doesn't seem to work
15:48.26GerbilWrkrazu, that did it, thanks
15:48.40saftsacki cant see it too but maybe the reason is why nobody uses it
15:48.43eKo1I said for porno. The purpose of VoIP is to make another medium to deliver porno.
15:48.48saftsacka circle ;)
15:49.04salviadudvideo telephony will be availabe when the hardware gets cheap
15:49.14saftsackyes think so too
15:49.37saftsackbut video telephony can just be there if everybody has a broadband internet connection
15:49.47saftsackand this means a big upload too
15:51.28eKo1salviadud: I think webcams are cheap enough as it is.
15:51.54salviadudwell, you wouldn't expect some dude to be at his computer while using the phone all the time
15:52.18salviadudyou gotta integrate
15:52.20*** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net)
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15:52.30salviadudwebcam + iaxy
15:52.35salviadudiaxy-cam phone
15:52.40salviadudsomething like that
15:52.44_MDC_I keep on getting the following error message when doing a H323 call; H.323 call 'ip$localhost/29214-0c9250b7' cleared, reason 12 (Not enough bandwidth), could it be that asterisk announces the wrong ip?
15:54.20salviadudhey, i need some feedback with this prank call i made
15:54.32salviadudmostly, because i recorded it in wav format
15:54.47salviadudthen i used sox to compress it into mp3
15:54.56*** join/#asterisk DarKnesS_WolF (n=wolf@81.10.111.139)
15:55.05salviadudi don't have a soundcard available, so i don't know if it sounds ok
15:55.26*** join/#asterisk awe6 (n=lba@user-12lml5g.cable.mindspring.com)
15:55.36salviadudsomegeek, with that said, would anyone like to hear it?
15:55.46salviadudi meant, so
15:56.57eKo1_MDC_: the message clearly states: Not enough bandwidth
15:57.23_MDC_eKo1, but asterisk and the gk is on the same lan, 100Mbit
15:57.44*** join/#asterisk Gunnar (n=gunnar@62.97.242.6)
15:58.11*** join/#asterisk bkervaski (n=bkervask@adsl-072-149-159-016.sip.bhm.bellsouth.net)
15:58.13_MDC_eko1, using ekiga/ohphone on the same machine agains the gk is working just fine
15:58.49bkervaskiHi all.   Is there a way to get * to display a specific string of text when a call queue is ringing it's extensions?  i.e., poke a caller id string in there so the people in the queue know what queue is ringing there phone?
15:59.12eKo1then I guess this is another bug in the h.323 asterisk channel driver
16:00.16_MDC_eKo1, i'm using the oh driver
16:01.31eKo1Good luck then. If you do find a bug, let the developers know. I'm no H.323 expert so I can't help any further.
16:01.48*** join/#asterisk lokkju (n=lokkju@unaffiliated/lokkju)
16:01.55clyrrad1Can anyone help me out with the MWI problem?
16:02.02[TK]D-Fenderclyrrad1 : You config looks fine
16:02.26clyrrad1ok.... thats what I was thinking - so why the MWI problem?  Its very strange
16:02.33_MDC_eKo1, ok, thanks anyway
16:02.34[TK]D-Fenderclyrrad1 : I'm betting there's something with the phone if it working fine aside from that and you are able to enter you box only to see new messages upon entry
16:02.42clyrrad1is there any known bugs with it?
16:02.46dorphalsigI am running 1.2.7 but some zap channels will die ocassionally and wont come up unless I restart asterisk and when I pickup the phone I get nothing, when I call it sounds as if the extension was busy, and when I look at the cli I see the extension is onhook but has an owner. is there anyway to fix this?
16:03.10clyrrad1TKD - the phones work on the other Asterisk server
16:03.15eKo1dorphalsig: Post it as a bug.
16:03.42clyrrad1I am using Asterisk 1.2.9.1
16:04.09bkervaskiHi all.   Is there a way to get * to display a specific string of text when a call queue is ringing it's extensions?  i.e., poke a caller id string in there so the people in the queue know what queue is ringing there phone?
16:04.48droopshey bkervaski, where are you trying to display it, in the cli?
16:05.21bkervaskiNo, on the SIP phone's caller id display.. right now it shows the callerid of the person calling, I was hoping to have it display the name of the queue...
16:05.27eKo1Has anybody tried the chan_ss7 driver?
16:06.27dorphalsigI am running 1.2.7 but some zap channels will die ocassionally and wont come up unless I restart asterisk and when I pickup the phone I get nothing, when I call it sounds as if the extension was busy, and when I look at the cli I see the extension is onhook but has an owner. is there anyway to fix this? Or at least to restart the channel without having to restart asterisk?
16:07.32droopsbkervaski, have you played with Set(CALLERID)
16:07.37*** join/#asterisk saftsack (n=saftsack@p54A7E68B.dip.t-dialin.net)
16:07.58bkervaskiNope.  So you're suggesting handle it in extensions.conf?  That's a good idea.  I'll give it a whack.
16:08.11droopsthats the first thing i would try
16:08.40droopsno idea if that will do what you need
16:08.44droops=o)
16:09.33bkervaskiWhat's the format?  set(callerid|mystring) ??
16:10.04[TK]D-Fenderclyrrad1 : Wish I could give you some more insight.....
16:10.07clyrrad1[TK]D-Fender - any ohter idea what can be the problem?
16:10.12*** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com)
16:10.13*** join/#asterisk tdonahue (n=tdonahue@207.138.151.58)
16:10.17clyrrad1hrm - :(  This is so frustrating
16:11.05droopsSet(CALLERID(NUM)=123456767)
16:11.14[TK]D-Fenderclyrrad1 : If calls flow, then MWI indication should as well.
16:11.19droopsor
16:11.24droopsSet(CALLERID(number)=123456767)
16:12.02droopsthats how i notify a office cell phone, that the call is comming from the office, and isnt a personal call
16:12.56[TK]D-Fenderdroops : I have a magic 867-5309 dial option for comic releif ;)
16:13.17clyrrad1TKD - yea thats why I am so confused by this - do you know of any bugs with MWI on this version?
16:13.28droopsthat will be implemented today
16:13.34droopsawesome idea
16:13.49tzafrir_laptopclyrrad1, #asterisk-dev is not #asterisk 2-tier support
16:14.27fileunless we're in a nice mood
16:14.33fileor waiting for a build to finish
16:15.40clyrrad1sorry - just at a loss on this one i been messin with it for 2 days now
16:16.42clyrrad1file with the sip debug show "MWI"?
16:16.54clyrrad1will*
16:17.16fileit should show chan_sip sending a packet to your phone saying, "this is the current state of voicemail... this is how many new, and how many old"
16:17.34clyrrad1yea i dont see any of that
16:17.48*** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com)
16:17.58filewhen leaving a voicemail?
16:18.25clyrrad1have not tried while leaving voice mail - is that info not sent all the time?
16:18.33clyrrad1I will try now while leaving voicemail
16:18.49eKo1~centosbug
16:18.50jbothmm... centosbug is a problem with the latest Centos kernels (4.2 and 4.3).  To fix it, paste everything inside the quotes into a root shell:  "sed -i s/rw_lock/rwlock/ /usr/src/kernels/`uname -r`-`uname -m`/include/linux/spinlock.h"
16:20.38clyrrad1file - there is no such message being sent while leaving a message too....
16:21.16stoffell_hclyrrad1: and what does "show voicemail users" show in the cli? (to be sure the message is left correctly)
16:22.23clyrrad1it shows that that there are messsages I can pastebin if you like its 5 lines
16:22.47stoffell_hwe believe you :)
16:23.27clyrrad1http://pastebin.ca/91416
16:23.36*** join/#asterisk RoyK (n=roy@193.75.62.110)
16:23.37clyrrad1haha - Ok
16:24.27*** join/#asterisk CoffeeIV (i=rgr@cpe-70-112-100-20.austin.res.rr.com)
16:24.35stoffell_hclyrrad1: what type of phones are they?
16:25.18stoffell_hRoyK: on the ferry? ;)
16:25.52RoyKstoffell_h: i'm off the ferry....
16:25.57clyrrad1They are GNET VP104S phones
16:26.55stoffell_hclyrrad1: no experience on those, but as file says, you should see MWI messages with tcpdump or sip debug... (if your phone subscribes to MWI that is)
16:27.29clyrrad1yes the phones do work with MWI as I had said before they work on another asterisk server
16:28.00stoffell_hclyrrad1: ok, then try with tcpdump or sip debug to see what message you get "on the other" server, and see if you get these on "this" server
16:28.10stoffell_hthere 'has' to be a difference in config..
16:28.38clyrrad1Well is the MWI not set in sip.conf with mailbox=?
16:28.46wunderkinum i assume that someone has checked voicemail.conf and sip.conf?
16:29.23clyrrad1Yes many times here is the config http://pastebin.ca/91354
16:29.26wunderkinclyrrad1, are you specifying the context
16:29.53wunderkinclyrrad1, invalid id
16:30.35wunderkinmust have just expired
16:30.44*** join/#asterisk lunk (n=lunk@66.152.8.184)
16:31.07lunkis there a way to set the Ttr dial flags when using .call files?
16:31.15clyrrad1I will paste it again
16:31.51clyrrad1here it is http://pastebin.ca/91426
16:32.03*** join/#asterisk nortex (n=breeves@snapper.titanspecialties.com)
16:32.51*** join/#asterisk McLazarus (n=mcallist@72.78.49.117)
16:32.55stoffell_hclyrrad1: your "other" server, same version as your current?
16:33.14clyrrad1no different versions
16:33.31McLazarusHi, I am sure I am missing something obvious in the docs but is there a way to take the output of a "pri debug span n" and just have it written to a file instead of the console?
16:33.41McLazarusor at least have it do both?
16:34.35clyrrad1the working server is using HEAD and the non working one is using 1.2.9.1
16:35.12RoyKclyrrad1: you mean trunk?
16:35.28stoffell_hi'd say: compare sip debug.. to be sure they both sent the same info..
16:35.36clyrrad1I got it from CVS head awhile back
16:35.52clyrrad1stofeell_h - I see no MWI info being sent
16:36.16stoffell_hclyrrad1: and do you see it being sent on the working server?
16:37.06clyrrad1I have not checked - but it seems pretty clear it is being sent on that server since the MWI works on that server
16:37.38stoffell_hclyrrad1: try sip debug on current server and restart a phone, the phone should at least "try" to subscribe to MWI...
16:37.42froguzsomebody has worked with micronet fxo gateways??
16:37.47stoffell_h(sip debug or tcpdump ..)
16:39.41*** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com)
16:40.16RoyKstoffell_h: sip debug is usually good for a start
16:40.54clyrrad1stofell_h yes i have tried to restart the phones and watch the sip debug i dont see any MWI stuff
16:41.21stoffell_hclyrrad1: ok, are you in the possibility to try exactly the same on the other server?
16:41.37clyrrad1mean setup wise?
16:41.54stoffell_hno, just booting the phone and making sure you see anything on MWI passing by..
16:42.13clyrrad1yes - I have done that - on the other server the MWI works
16:42.18*** part/#asterisk iriga (n=adu@LSt-Amand-152-31-4-219.w82-127.abo.wanadoo.fr)
16:42.27McLazarusah, stupid.  I see I just have to do "pri set debug file"
16:42.28stoffell_hclyrrad1: so you see mwi messages when using sip debug?
16:42.39clyrrad1I see the MWI light turn on the phone
16:42.48stoffell_hMcLazarus: thanks for sharing, i didn't knew that either!
16:43.02stoffell_hclyrrad1: ok, but you didn't check the sip debug then..
16:43.13lunkdoes anyone know how to set trunk options from inside a .call file?
16:43.22clyrrad1nope - becase it was working on the other server there are so many phones it will fly by - i wouldnt see it anyway
16:43.54stoffell_hhm, that's why I prefer tcpdump :) you can select info from specific ip's
16:44.32*** join/#asterisk paryl (n=chatzill@209.236.78.59)
16:44.34Dr-Linux|workhi guys
16:44.35clyrrad1Yea - this is so strange - the phones work in every other regard - just not the MWI
16:44.39Dr-Linux|workstoffell_h: hey
16:44.39clyrrad1it makes no sense
16:44.52clyrrad1Hi Dr-Linux
16:45.01stoffell_hhey Dr-Linux|work
16:45.10clyrrad1Everyone who has looked at my pastebin configs has found no issue with them....
16:45.21parylclyrrad1: what model of phone?
16:45.25stoffell_hclyrrad1: it doesn't make sens now, but if you can, try to compare, make some tcpdumps
16:45.27Dr-Linux|workwell, i just saw someone is using those entries for a user in sip.conf:
16:45.28Dr-Linux|workcallgroup=1
16:45.29Dr-Linux|workpickupgroup=1
16:45.29Dr-Linux|workcall-limit=1
16:45.34stoffell_hthere 'has' to be a difference in info being exchanged..
16:45.42Dr-Linux|workanybody can guide me, what's those for?
16:45.56clyrrad1they only thing I can think is that they are different versions of Asterisk
16:46.14clyrrad1I dont know if there are bugs with my current version - but it was stable so I doubt it is a bug
16:46.40stoffell_hclyrrad1: yes, and then it would be great to know if the info being exchanged is the same or not. it could still be a bug, or something else that's being overlooked
16:46.53stoffell_hDr-Linux|work: http://www.voip-info.org/wiki/view/Asterisk+callgroups+and+pickupgroups
16:47.11paryli'm having issues with IAX calls going dead, sometimes in one direction, sometimes in both, for about 10 seconds.  the calls resume like normal after that.  it's totally random, and there are no errors in the logs... i can't figure out what could be causing it?
16:47.23Dr-Linux|workstoffell_h: ok, and any clue about "call-limit=1" ?
16:47.47stoffell_hDr-Linux|work: check it on voip-info, i blieve it's something with max. calls being sent to a device
16:47.53clyrrad1stofell_h - yea not sure how else to troubleshoot this - and if i do a show voicemail users it shows that there are new messages waiting
16:48.21Dr-Linux|workstoffell_h: i google it, but coudn't find, so came here
16:49.24stoffell_hclyrrad1: i vote for `tcpdump host <ip>` ... :)
16:49.48clyrrad1what do you want me to look for?
16:50.08clyrrad1I can only tcp dump on the server running asterisk though
16:50.10stoffell_hclyrrad1: and use ethereal to cmopare the 2 streams.. (ideal to see if MWI info is exchanged while booting the phone)
16:50.11*** join/#asterisk visba (n=dca[lapt@sta-208-139-193-162.rockynet.com)
16:50.17*** join/#asterisk |dennis| (i=dennis@200.32.215.84)
16:50.37stoffell_hclyrrad1: that's good, you see all info going from/to the asterisk server and your phone
16:51.05RoyKclyrrad1: that's ok. tcpdump -s0 -w somefile udp and host x.x.x.x
16:51.06RoyKfor instance
16:51.14*** join/#asterisk Tako-san (n=Tako-san@24.108.162.254)
16:52.27clyrrad1ok doing that what should i grep the file for?
16:53.00stoffell_hclyrrad1: use ethereal on your workstation to open the capture file.. it's easier
16:53.08stoffell_hor should I say wireshark... :)
16:53.24RoyK#include <pcap.h>
16:53.25RoyK:=
16:53.27RoyK:P
16:53.36stoffell_hlol
16:53.44clyrrad1well not too famaliar with etherreal - but i do have the dump in the file and I can see it - just need to know what to look for
16:54.30stoffell_hclyrrad1: the mwi info... (assuming you did this on the 'working' server)
16:55.00clyrrad1do you know if the dump will actually say 'mwi' or something else?
16:55.28stoffell_hthat I don't know for sure, with ethereal you might be able to "follow the stream"
16:55.54stoffell_hclyrrad1: can u put the dump file  online?
16:56.06ChrisDe3to use the function "call-limit" ... I will have to add an entry in sip.conf (respectively a column in the database), right? and set call-limit=1 e.g.?
16:56.48*** join/#asterisk think_ (i=think@noguff.net)
16:56.58ChrisDe3because... it doesn't work for me
16:57.00parylwhat can i do if an FXO card doesn't seem to be getting hangups very quickly?
16:57.13clyrrad1im checkign the dump file now
16:58.56rob0paryl: wouldn't happen to be an X10[01]P, would it?
16:59.07*** join/#asterisk Delta239 (n=adfadsf@200.124.18.171)
16:59.17parylrob0, it's a sangoma a200
16:59.17Delta239hey does the sendmail service affects asterisk?
16:59.30stoffell_hDelta239: n.o.t.
16:59.45rob0Ah, in that case I have no idea.
16:59.52[TK]D-Fenderparyl : Need to try and get your telco to enable disconnect supervision
16:59.59*** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net)
17:00.04clyrrad1Ok checked the dump file on the working server and see nothing about MWI....
17:00.05[TK]D-Fenderparyl : Either polarity reveral or cut.
17:00.08think_anyone know how to "busy out" a pots line through asterisk?
17:00.43rob0Delta239: if voice mail is to be emailed, you need a sendmail binary. Doesn't need to be a full MTA, could be something like nullmailer.
17:00.46paryl[TK]D-Fender: k, i'll take a look at that, thanks man
17:01.12burnproofparyl: if either of the two won't work on your end you could fiddle on busydetect parameter on zapata.conf
17:01.21Delta239the main problem is that i can't get asterisk to run
17:01.24ChrisDe3anyone familar to call-limits?
17:01.41Delta239while is booting up it says nAsterisk service started but then when i type in asterisk -r
17:01.42*** join/#asterisk mut (n=animenod@65.111.222.120)
17:01.55DovidTK: the file format changed for the pollycoms, since the page on the wiki for auto answer was created
17:01.57Delta239it comes with the error message saying that asteriskctl exist?
17:02.02Dovidany ideas ?
17:03.33*** join/#asterisk Bullseye_Network (n=info@216.143.192.69)
17:03.59*** join/#asterisk johnnyb (n=jonathan@adsl-38-9-196.tulsaconnect.com)
17:06.27clyrrad1stofell_h on the working server I just found this...
17:06.28clyrrad1Messages-Waiting: no
17:06.28clyrrad1Message-Account: sip:asterisk@
17:06.28clyrrad1Voice-Message: 0/5 (0/0)
17:06.35clyrrad1is that what you were looking for?
17:07.11clyrrad1becase if so - that information is not being sent on the NON working server
17:08.23stoffell_hclyrrad1: hm, okay
17:08.57clyrrad1that give any clues?
17:08.57stoffell_hclyrrad1: on cli; do "show modules", is "app_hasnewvoicemail.so" and "app_voicemail.so" loaded?
17:09.46clyrrad1on the working server yes
17:09.49clyrrad1checking the non working one
17:10.23clyrrad1yes they both seem to be there on both servers
17:10.31clyrrad1the all have a Use Count of 0
17:11.00clyrrad1they*
17:11.05stoffell_hhm, okay, then i'mout of thoughts..
17:11.22clyrrad1haha - yea thats what i keep runnign into - its so frustrating
17:11.22stoffell_hunless you can try up- and/or downgrading..
17:11.40clyrrad1I still wonder if its a bug
17:11.42stoffell_hto see if it's version-related
17:12.08clyrrad1guess I will have to try to upgrade :s
17:12.56Dovidcan anyone help me with polycom paging ?
17:12.59stoffell_hyes, see if it stays.. (still I wonder if the phone subscribes correctly to the MWI)
17:14.14clyrrad1it must since they work on the other server
17:14.31Cresl1nmaybe the MWI tube is full
17:14.33stoffell_hto be sure, tcpdump could tell you ..
17:14.51clyrrad1Cresl1n - how could i determine that?
17:15.00*** join/#asterisk lung (n=lung@24-148-96-186.ip.mhcable.com)
17:15.11Cresl1njust look
17:15.18Cresl1nmake sure there aren't any inflatable dolls in it
17:15.18clyrrad1look where?
17:15.27clyrrad1very funny
17:15.28stoffell_homg...
17:15.36stoffell_h;)
17:15.39eKo1I just configured my te410p and zttool is telling me they're all OK. shouldn't they be all on red though?
17:15.41Cresl1nthey seems to clog up the tubes a lot ;-)
17:16.25clyrrad1looks like gonna have to try the upgrade route
17:17.02clyrrad1no one can find config probs in this channel or in the dev chanel so must be version related....
17:17.43stoffell_hclyrrad1: indeed, tcpdump should show if the MWI of the phone subscribes though...:)
17:17.54clyrrad1I have tried it and found nothing
17:18.09clyrrad1both on the working server and the non working one
17:18.17clyrrad1I have only found it with sip debug on the working server
17:18.26clyrrad1the non working server is not sending any such messages
17:18.47stoffell_hhm, oke.. i thought the phone had to "subscribe" to * to see the messages
17:20.08lungdoes anyone know if this is a valid invite: "INVITE sip:1##########;npdi=yes@x.x.x.x SIP/2.0"
17:20.18*** join/#asterisk MatsK (i=MatsK@83.233.97.229)
17:20.41*** join/#asterisk RoyK (n=roy@193.75.62.110)
17:22.51lungit works in pre-1.2.7, but not after due to bug 6409
17:27.43*** join/#asterisk bitboy (n=amit@adsl-065-012-197-229.sip.bct.bellsouth.net)
17:28.56bitboyhello.
17:30.31bitboyAnyone know if following is possible: dial an extension---this dials a number, once that call is over, another number is autodialed.  So I dont want the channel closed after first call
17:33.25bitboyanyone?
17:42.37andrejkwI am having hard time getting FWD to work.
17:42.43andrejkwOutgoing calls just don't work.
17:42.56andrejkwI get that it's circuit-busy all the time
17:43.33*** join/#asterisk rene- (n=rene-@dsl-200-67-175-250.prod-empresarial.com.mx)
17:44.13rene-can PauseQueueMember be used from the CLI or the Manager interfase? because i can see it in show applications but i cannot execute it from the cli
17:44.26rene-or is it something i can accomplish via the asterisk db?
17:45.35rob0andrejkw: I can only share my own recent experience with FWD. SIP does not work at all, which might be because the FWD clock was about 6 minutes off the last time I called 613. IAX2 works fine. Also ...
17:46.15rene-the command is QueuePause
17:46.31rob0... I tried to sign up at their forums, but the autoresponder was broken, so I could not get the confirmation. "Bad address syntax," is what my Postfix said.
17:46.33andrejkwI can't get IAX2 to rgister.
17:46.45andrejkwIt just says Rejected.
17:46.45*** join/#asterisk auralia (n=ehernand@207.71.51.162)
17:46.50rob0did you just now sign up?
17:46.59andrejkwYes
17:47.05andrejkwwell 3 - 4 hours ago
17:47.13auraliashould there be an alarm in zttool if there is no phone line connected to the tdm and the PSTN?
17:47.21rob0I think it took about that long for it to register for me.
17:47.30*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
17:47.38andrejkwSo shoild I try again and see?
17:47.43andrejkw*should
17:48.00MrChimpy.-
17:48.08rob0Follow the instructions for FWD IAX2 at the wiki, and yes, try again.
17:48.44rob0fwd-out/783889   192.246.69.186  (S)  255.255.255.255  4569          OK (40 ms)
17:49.43*** join/#asterisk auralia_aNew (n=none@207.71.51.162)
17:50.26auralia_aNewsorry i got kicked off, i just asked if zttool should have  any alarms if there is no  line connecting my tdm2400 and the  PSTN
17:51.04*** join/#asterisk oceanlan|dustin (n=info@cpe-24-210-253-66.woh.res.rr.com)
17:52.08andrejkwDo I absolutely need 2 entires (one for incoming and one for outgoing)?
17:52.10*** join/#asterisk anonymouz666 (n=anonymou@20151155235.user.veloxzone.com.br)
17:52.39*** join/#asterisk Iam8up|lpy (n=iam8up@cpe-24-210-253-66.woh.res.rr.com)
17:53.00anonymouz666anyone use Dell PowerEdge 850 in here?
17:53.06rene-how do i send the parameters for Asterisk Manager's QueuePause
17:53.08rene-?
17:53.21rene-dial plan application works like (queuename|interfase)
17:53.34rene-but i dont know what the right syntax would be for AMI
17:53.55*** join/#asterisk dwmw2_gone (n=dwmw2@baythorne.infradead.org)
17:54.23*** join/#asterisk TESTER2 (n=Cyber@modemcable082.42-81-70.mc.videotron.ca)
17:54.45oceanlan|dustinAre T1 "smartjack" interfaces symmetric? Can I replace a telco smartjack with Asterisk and a quad-T1 card?
17:55.20andrejkwI still get Rejected :(
17:57.44*** join/#asterisk bitboy (n=amit@adsl-065-012-197-229.sip.bct.bellsouth.net)
17:58.59*** join/#asterisk klictel (n=klictel@207.107.208.137)
18:03.39*** join/#asterisk ayamkeren (n=makoata@bb219-74-196-86.singnet.com.sg)
18:05.19*** join/#asterisk saftsack (n=saftsack@p54A7E68B.dip.t-dialin.net)
18:08.03my007mshello any one use E1 there
18:08.45my007msafter i start asterisk with E1 card from digium i get in asterisk CLI
18:08.54my007msthis error
18:09.14my007mschan_zap.c:6337 handle_init_event: Detected alarm on channel 14: No Alarm
18:09.14my007msJul 18 14:04:35 WARNING[11738]: chan_zap.c:1435 zt_disable_ec: Unable to disable echo cancellation on channel 14
18:09.19rob0andrejkw: the advantage of 2 entries vs. a single "friend" entry is that you can use different contexts for incoming and outgoing calls. Seems important to me, but YMMV.
18:09.21my007msfor all chanels
18:10.17andrejkwook
18:10.40ayamkereni just finished installing fedora core 5, but when i try to compile zaptel, it's failed, any idea how to do it correctly ?
18:12.00jbroomeyou have the build tools installed?
18:13.13Dovidmy polycom all of a sudden has been taking for ever to boot after provisioning and it wont contact the FTP server anymore. anyonehave an idea ?
18:14.06Dr-Linux|workquestion, is there any command on CLI> could tell me that how many users are  registered with my asterisk?
18:14.23*** join/#asterisk unit (n=doom@Toronto-HSE-ppp3780161.sympatico.ca)
18:14.25*** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net)
18:14.26*** part/#asterisk ChrisDe3 (n=Chrisde3@port-87-234-141-161.dynamic.qsc.de)
18:15.36clive-did Nufone go bankrupt?
18:15.36*** join/#asterisk Coeus (n=Coeus@ip24-255-125-43.dc.dc.cox.net)
18:15.39my007msDr-Linux|work, yes there is
18:15.56Dr-Linux|workmy007ms: what's that?
18:16.03my007msDr-Linux|work, if u speek sip
18:16.06rob0Dr-Linux|work: perhaps "sip show peers" and "iax2 show peers", whatever other protocols you might be using.
18:16.42TrixVoxclive-: Yes
18:17.08clive-Trixvox wow, ...what a pity
18:17.36gandhijeeanyone here use a TDM400 in a RISC machine?
18:17.40Dr-Linux|workrob0: that doesn't show as i want
18:17.42TrixVoxnot really
18:18.01clive-Trixvox, I wonder what my balance was...:(
18:18.44andrejkwAnyone here using FWD with IAX. I can't get mine to register, it's been 4 hours since I enabled IAX support on the account and I am still getting "Rejected". Is there something else I need to do?
18:19.28DaminTrixVox: They are up and running.. w/ new partners.. NuFone isn't dead..
18:19.43DaminTrixVox: My 800 number stil works, and my other one is being ported...
18:20.40clive-one thing I can say about Nufone, is that he was always honest in the dealings I had with him, I do hope that they are up and running still
18:21.20clive-damin can one still get in touch with them ?
18:21.35Daminclive-: Yep.. support@nufone.net. (or is it .com?)
18:21.50clive-damin thanks
18:21.54Daminclive-: Your balance is probably still there.. go login and check..
18:22.23clive-my balance was always screwed up...:)
18:23.54*** join/#asterisk apocn (n=apo@225stb46.codetel.net.do)
18:24.50apocnCan I originate a call from one extension to another from the CLI?
18:25.49*** join/#asterisk [Airwolf] (n=airwolf@dsl51B67BC5.pool.t-online.hu)
18:25.50*** part/#asterisk lung (n=lung@24-148-96-186.ip.mhcable.com)
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18:29.47*** join/#asterisk g__ (n=g@itd01fw-fibre.itdepartment.com)
18:30.47andrejkwIs there any Phone Administration Menu available for Asterisk/
18:30.59*** join/#asterisk pdtmobile (n=ptinsley@209.12.249.243)
18:30.59*** join/#asterisk hi365 (n=hi365@bzq-167-158.dsl.bezeqint.net)
18:31.09clive-msg hi365 hi
18:31.14hi365HI!
18:31.26jbroomehai2U
18:32.41rob0heat index of 103F (39C) so far, forecast to go up to 108F (42C) ... whew
18:32.46*** join/#asterisk pdtmobile (n=ptinsley@209.12.249.243)
18:32.56rob0and I am thinking of going out into that heat
18:33.29rob0car AC isn't working and the engine itself is overheating.
18:37.04*** part/#asterisk apocn (n=apo@225stb46.codetel.net.do)
18:37.56[TK]D-Fenderandrejkw : vim, emacs, mc, gedit, kedit, and amillion others, take your pick.
18:38.03tessier_What do people recommend these days for interfacing asterisk with a T-1 that does not involve a PCI card?
18:38.15pdtmobileare there any linux supported USB sound cards?
18:38.21andrejkwlol
18:38.24andrejkwI mean an IVR.
18:38.26Nuggetha ha ha
18:38.27andrejkwa pre-made one
18:38.45pdtmobileI want to add sound to a couple of PBXs without cracking the case
18:39.35[TK]D-Fenderanderiv : IVR's are damn quick to make.... nothing to sweat over
18:40.00E-bolathe recoding is the sucky part of ivr's
18:40.03E-bolasetup is easy
18:41.34*** join/#asterisk TripleFFFF (n=TripleFF@147-102.mc.cite.net)
18:41.40TripleFFFFwhats lall this "Forcing Marker bit, because SSRC has changed" ?
18:41.58gandhijeetessier_: the only other option AFAIK is a phonebrige/redphone
18:42.01gandhijeewhatever its called.
18:42.05gandhijeebut that too has a PCI card
18:42.16gandhijeei guess it might be possible to use some cisco crap
18:42.29tessier_gandhijee: I'm thinking Cisco, Mediatrix, Rhino, something like that.
18:43.00jbalcomb[TK]D-Fender: did you test the info i sent you?
18:43.03gandhijeewhy not use a PCI card?
18:43.19tessier_gandhijee: No echo cancelling, no signal processing, every PC is different.
18:43.41gandhijeeumm, there is echo cancelling on the cards
18:43.43gandhijeejust more $$
18:44.04[TK]D-Fenderjbalcomb : Yes, haven't had a chance to set my system up for it yet though.  Also I'm a little unclear on the network map but I'll ask about that later.
18:44.06[TK]D-Fenderjbalcomb : PM
18:44.16gandhijeeyou could always dedicate a PC as a PSTN bridge and use the heavy echo cancellers on them
18:44.18tessier_I actually had decent luck with my digium PCI card but their POTS cards were a disaster and the support sucked.
18:44.39tessier_So I am reluctant to go with them again.
18:44.54tessier_Actually, the support on getting the PRI card going was great.
18:45.00gandhijeei haven't has any problems with the Digium TDM400
18:45.03tessier_The support on fixing all of my broken POTS cards was what was terrible.
18:45.07gandhijeethere is always the option of sangoma.
18:45.19tessier_Yeah, I have been looking at Sangoma.
18:45.38[TK]D-Fendertessier_ : they're GOLD
18:46.14[TK]D-Fendertessier_ : *0* echo with their DSP and no PCI issues.
18:46.14gandhijeei have them for for a PRI card, and use a digium for the POTS line
18:46.41gandhijeei think i've managed to port zaptel to ARM/Xscale i think
18:46.55tessier_[TK]D-Fender: You are saying that Sangoma is gold?
18:47.02hi365there really are good. if only i could get mine to work!
18:47.05gandhijeeyea, sangoma is very good
18:47.14[TK]D-Fendertessier_ : As is great quality and solid like a rock.
18:47.21gandhijeehi365: haven't you been trying to get them for like a week now
18:47.23tessier_[TK]D-Fender: What does it have over the digium PRI card?
18:47.25g__Sangoma is very good for digital interfaces.
18:47.34clyrrad1This is my first time to upgrade asterisk - what is the proper way to do it?  I am on digium ftp and see a few options for the 1.2.10 including some "PATCH" files - to do the upgrade do I just download the new source and rebuild asterisk?
18:47.35g__I have some reservations on their analogue products.
18:47.36gandhijeehi365: i told you that you should look at their wiki
18:47.51hi365gandhijee feels like a year. it worked till i tried to add more channels
18:47.58gandhijeeyeah Sangoma made the original T1/E1 cards
18:48.00*** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net)
18:48.01[TK]D-Fendertessier_ : 3.3V/5V interoperable, high quality EC onboard, Doesn't mind sharing interrupts
18:48.09[TK]D-Fendertessier_ : Fors starters
18:48.16hi365benn there done that more times then i remember
18:48.49clyrrad1TKD- Have you upgraded asterisk before?
18:49.17[TK]D-Fenderclyrrad1 : Plenty of times.
18:49.22*** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com)
18:49.25clyrrad1am I correct in my above statment?
18:49.40clyrrad1to just download the new source and rebuild?
18:49.53*** join/#asterisk lokkju (n=lokkju@unaffiliated/lokkju)
18:50.03gandhijeeg__: why not fix them yourself, and help those guys out
18:50.07[TK]D-Fenderclyrrad1 : download all the new and COMPLETE source files from FTP, wipe your old extracted source.  Wipe your modules folder.  Recompile everything.  DONE
18:50.46clyrrad1Excellent thanks :)
18:51.09*** join/#asterisk AlexCTI (n=alex@adsl-074-238-025-003.sip.mia.bellsouth.net)
18:51.14[TK]D-Fenderg__ : Funny that SUCCESS can be found in its entirely in the word SOURCE ;)
18:51.15g__gandhijee: I'm a sysadmin, so I have no free time..
18:51.37g__The thought had crossed my mind though.
18:51.42*** join/#asterisk johnnyb (n=jonathan@adsl-38-9-196.tulsaconnect.com)
18:51.52gandhijeeO
18:52.03*** join/#asterisk auralia_aNew (n=none@207.71.51.162)
18:52.33g__TKD: of course, just because it's open source, doesn't mean it's gold.
18:53.07[TK]D-Fenderg__ : I wasn't referring to open source so much as "compile the damn thing yourself you lazy bastard" ;)
18:53.48g__TKD: I have too many servers for that.. pre-packaging has saved my ass a couple of times allready.
18:53.58[TK]D-Fenderg__ : and its the fact is a solid product that makes it "gold".  Closed products can earn that rating from me as well.  I prefer "open" but it isn't the be-all and end-all.
18:54.29gandhijeeg__: then you should be doing it yourself if you need the prepackaged
18:55.30g__gandhijee: lovely idea.  Sangoma's build script is a bit warpped.  I started working on this and I asked them for help.. and they responded with "oh, the script can build debian packages itself"..
18:56.01gandhijeei wouldnt know about debian
18:56.02_MDC_where can I found a good explanaition (and definition of) FXO, FXS, ISDN, digital/analog PSTN?
18:56.04gandhijeei use gentoo on my stuff
18:56.17gandhijee_MDC_: voip-info.org probably
18:56.27g__Debian is just like gentoo.. except the patching and compiling happens ahead of time.
18:56.30*** join/#asterisk tsurk0 (n=tsurko@85.187.160.157)
18:56.41g__(doesn't know much about Gentoo)
18:56.54gandhijeei used debian and slack for about a whole 2 days for devel
18:57.02gandhijeethen i trashed it all and went back to gentoo
18:57.16g__reminds me of my experience with XP.
18:57.24gandhijeei never could get my cross-devel stuff to work right with debian
18:57.25[TK]D-Fender_MDC_ : Here.....
18:57.26[TK]D-Fender~book
18:57.28jbotfrom memory, book is a book called  Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
18:57.33gandhijeeso i just did it by hand on gentoo
18:57.50g__In the end, it's all the same software.. I don't really care where it comes from.
18:57.54gandhijeetrue
18:58.13*** join/#asterisk hi365 (n=hi365@bzq-167-158.dsl.bezeqint.net)
18:58.31_MDC_[TK]D-Fender, i'm reading the book right now, but just wanted a good explanation table... will read further
18:59.02[TK]D-Fender_MDC_ : I've read up on it in the book and the quality of the background info was pretty good and thorough
18:59.19_MDC_ok, thanks, will read on
19:00.18Dovidmy polycom wont prevision.
19:00.22g__Question for everyone: I'm having occasional one-way audio problems over phone calls from PRI <-> IAX2 <-> SIP.  (The callee can't hear the caller and eventually hangs up.)  What troubleshooting steps can I take?
19:00.37Dovidi change the directory.cfg file but the phone wont grab it, any ideas ?
19:00.55g__(There is no IAX2 trunking; debugging isn't enabled, but that could be done;.. )
19:01.12g__Dovid: check the ftp server logs
19:01.21g__See what files it *is* grabbing.
19:01.35Dovidi did
19:01.39g__And?
19:01.42Dovidsays there wre no changes
19:01.45Dovidthere wre
19:01.59g__the ftp server log says that?
19:02.08g__What files does it download?
19:02.15*** join/#asterisk apocn (n=Apo@225stb46.codetel.net.do)
19:02.19Dovidnone
19:02.31g__What ftp server are you using?
19:02.43Dovidvsftpd
19:02.47g__good!
19:02.47Dovidit was working b4
19:02.50*** part/#asterisk TripleFFFF (n=TripleFF@147-102.mc.cite.net)
19:02.54apocnIs it possible to originate a call from one extension to another from the CLI?
19:03.04Dovidapocn: yes
19:03.09g__How does it know which ftp server to access?  DHCP information?
19:03.16Dovidno
19:03.19Dovidi put it in
19:03.22Dovidnot using dgcp
19:03.24Doviddhcp*
19:03.37Dovidi went and got clean configs from digium
19:03.49Dovidand it should wipe it clean but it isnt
19:03.49g__And it's not even downloading 0000xxxxx.cfg?
19:04.03Dovidusing the default
19:04.13apocnDovid: where can I read about this?
19:04.26Dovidshould i make it its own cfg file ?
19:04.33Dovidapocn: voip-info.org
19:04.41g__Definately!  if you have more than one phone, this is a must.
19:04.57apocndoes it have a particular name?
19:05.12Dovidhttp://pastebin.ca/91586
19:05.18Dovidusing only one phone now
19:05.26clyrrad1TKD - Upgrade worked thanks bro :)
19:05.34Dovidapocn: u wana make ur own app ur do it from the CLI ?
19:05.47apocnyeah
19:05.57apocnI made a webapp similar to FOP
19:06.05g__You should see download attempts of 000f4xxxx.cfg, 000000000.cfg (if the former doesn't exist), bootrom.ld,  000f4xxxx-phone.cfg,  000f4xxxx-directory.cfg, 0000000000-directory.cfg (if the former doesn't exist).. etc
19:06.23Dovidcan u look at my pastebin ?
19:06.24apocnusing Ajax and PHP. Right now you can see queues/agents, and you can move an agent from one queue to another
19:06.27Dovidhttp://pastebin.ca/91586
19:06.34jbalcombDovid: it can get the ftp server name from dhcp
19:06.47Dovidi dont have dhcp now
19:06.49[TK]D-FenderDovid : You are modifying 00000000-directory.xml for your phone to use?
19:06.50jbalcombDovid: did you create the PlcmSpIp user?
19:07.11[TK]D-FenderDovid : you can hardcode the provisiong sever right on the phone.
19:07.31Dovidjbalcomb: no,i redid the phone and put in diff ftp log in into
19:07.33[TK]D-Fenderjbalcomb : And don't you tell me that you LEFT it at default!
19:07.52DovidTK: i am tryin to get paging working, from what i understand I have to edit the fiels and then have the phone grab it
19:08.14jbalcombDovid: my polycom phones provision right out of the box. i copy <mac>.cfg and phone<exten>.cfg and they work.
19:08.33Dovidwhat do u mean by copy ?
19:08.46apocnDovid, but now I want that when the supervisor clicks on the agent, communicate them both (like calling from his softphone to the agent)
19:08.53jbalcomb[TK]D-Fender: youre damn right i did. i'll fix it as i go.
19:09.04jbalcomb;)
19:09.05g__Dovid: your phone's logs are interesting, but your vsftpd logs would be more helpful.  Mine show up in /var/log/auth.log
19:09.09*** join/#asterisk NDT (n=nunya@cpe-24-195-66-214.nycap.res.rr.com)
19:09.15Dovidapocn: dont know the commands by heart. type in help and it will give em to u
19:09.33apocnok, thanks
19:09.53rene-can i unpause an agent from the cli??
19:10.04Dovidi dont have /var/log/auth.log
19:10.40[TK]D-Fenderjbalcomb : :O
19:10.41Dovidg__: what do u mean that they are interesting ?
19:10.50NDTmy dial command goes to a macro...when I run an agi script from inside the macro I am setting a variable in my agi script...when the call is completed and I return to after dial and go to h exten...I can't retrieve the contents of teh variable...so what happens to it? LOL Assuming this has something to do with setting it while I am in the macro
19:11.23DovidTK: did u look at my pb ?
19:11.43gandhijeeanyone here use and of the TDMoE stuff?
19:11.45[TK]D-FenderDovid : Yes, and it says nothing about picking up its directory files.
19:12.00gandhijeei get a TDMoX:no master, is that normal?
19:12.18Dovidhmm
19:12.20Dovidy would that be ?
19:12.43[TK]D-FenderDovid : What is the name of the directory file you modified and the phone doesn't see as different?
19:12.48clyrrad1TDK - I found the MWI problem :) :)
19:12.59[TK]D-Fenderclyrrad1 : And it was... ?
19:13.00Dovidalso if i wipe out all files in my ftp directory and copy all the files fresh from the polycom zip shouldnt the phone revert to out of the box ?
19:13.11[TK]D-FenderDovid : NO.
19:13.23clyrrad1TKD- notifymimetype=text/plain
19:13.25[TK]D-FenderDovid : Answer my last question please...
19:13.27clyrrad1I commented that out and it works
19:13.36[TK]D-Fenderclyrrad1 : Where was that occuring?
19:13.40clyrrad1sip.conf
19:13.46Dovid000000000000-directory.xml
19:13.51clyrrad1under [general]
19:14.16[TK]D-FenderDovid : That file WILL NOT WORK.  only on the FIRST boot up of a new phone....
19:14.21Dovidthe manual said i dont need to set it as <mac>-directory.xml if i am only using one phone
19:14.26Dovidoh ok
19:14.31clyrrad1how crazy - that one line was driving me nuts LOL
19:14.34*** join/#asterisk eliel (n=eliel@200.123.183.89)
19:14.44Dovidso if i wana prevision i have to use <mac>-file.cfg ?
19:14.57*** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net)
19:15.24*** join/#asterisk _4d4m_ (n=adam@62.69.102.99)
19:15.27NDTLet me rephrase that last question....If you set a variable in AGI "SET VARIABLE blahblah" but you are inside a macro when you call the AGI script...what happens to the variable when you come out of the macro? I can't retrieve it when it returns to the h exten in the contect the macro was called from
19:15.28*** join/#asterisk X-Rob_ (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au)
19:15.54NDTcontext even
19:17.24DovidTK: just fid <mac>.cfg and it still wont grab it
19:17.50Dovidone sec
19:18.08rob0Maybe a dumb question, sorry, but what happens if I SetCallerID on a call going out a residential PSTN line? Does the telco's Caller ID cancel it out? (This is BellSouth, if that matters.)
19:18.53rob0Kind of hard to test because I have no cell phone and I don't pay BellSouth for caller ID service.
19:19.39[TK]D-FenderDovid : pastebin this file you just mentioned (including its exact name)....
19:19.50NDTahh nevermind it is destroying that channel variable at hangup...
19:19.56syzygyBSDrob0: you have to have the ability to set the callerid, normal pots lines don't let you set it
19:21.52rob0Yeah, this is just a home telephone line, no extra services at all.
19:22.09*** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-155-138-106.red.bezeqint.net)
19:22.36*** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp)
19:22.36DovidTK: one sec, phone
19:23.05rene-rob0: apparently setoutgoing wiil only work on digital lines and not in every case
19:23.12rob0Phone?!? You mean people here actually TALK on those things? Ewwww.
19:23.12rene-setoutgoing callerid that is
19:23.27tzafrir_laptopwe had to move to a different room at work, and suddenly we have a severe echo problem
19:24.04tzafrir_laptop(not from the phone or anything. The achustics of the room itself. Go figure...)
19:24.08rob0tzafrir_laptop: put a big cork board on the wall :)
19:24.47DovidTK: http://pastebin.ca/91607
19:25.19trelane_anyone here that uses snom phones mind sending me a couple of their stock configs for mass deployment?
19:25.24trelane_or publishing them somewhere
19:26.33g__tzafrir_laptop: I recommend putting carpet scraps on the wall.
19:26.50g__..Or put cubical dividers around.
19:27.08g__Plants, bookshelves..
19:27.13rob0Crucify a boss. Nail him to the wall.
19:27.22g__Yup--people also work.
19:27.39rob0Believe it or not, they absorb a lot of sound while they're decomposing.
19:27.51stoffell_h2 will even work better...
19:27.54g__I didn't know that.
19:28.06rob0'Course the smell might get bad after a bit, but not as bad as while they were alive.
19:28.10DovidTK: Also alook here http://pastebin.ca/91612
19:28.31DovidTK: and the other is http://pastebin.ca/91607
19:28.48g__Pictures of your boss?
19:29.00trelane_tzafrir_laptop, http://froogle.google.com/froogle?q=acoustic+foam&hl=en&btnG=Search
19:29.23g__expensive, eh?
19:29.44trelane_g__: yeah acoustic foam is not cheap
19:29.50DovidTK: u there ?
19:29.57trelane_you don't need much of this stuff, get the heavy foam and do only one wall and it should kill the whole room
19:30.26trelane_g__, you ought to see how much the adhesive costs
19:30.34g__How much is one room's worth?
19:30.40trelane_g__, what room size?
19:30.53g__this room.
19:31.02trelane_this is a channel not a room, if you want a chat room try AOL People Chat?
19:31.06g__how about a small bedroom.
19:31.12trelane_say 10x14?
19:31.18g__Sure.
19:31.33stoffell_hg__: depends on what country you are in ... some small bedrooms can be pretty big..
19:31.59g__That's true.. North American standards.
19:32.26g__(We'll catch up with the rest of the world eventually.)
19:32.36stoffell_h;)
19:32.51stoffell_hthat's not always the best thing to do. lol
19:33.06Dovidwhere r my polycom people ?
19:33.34jbroomethey ran away, along with your "a" and "e"
19:33.47stoffell_hand they were fast..
19:34.03gandhijeeanyone here use TDMoE?
19:34.08gandhijeei am apparently having some problems with it
19:34.54Dovid[TK]D-Fender: please let me know when u return
19:35.07trelane_g__: the Sonomatt stuff is about $950 for the foam, assume another 50-100 for the adhesive for said room, it breaks down to roughly $18/linear foot with 8' ceilings
19:35.19jbroomeoh god, the "y" and "o" ran away too!
19:35.49g__Wow.. I'm going to stick with carpet scraps and towels myself, thanks.
19:36.20trelane_g__, right but with this stuff you really only SHOULD do one wall, I priced all 4 (As that's how I do recording studios which must be totally dead (high pile carpet and sonomatt'd ceiling as well
19:36.32*** join/#asterisk Zodiacal (i=hehehe@bdsl.66.14.242.199.gte.net)
19:36.32*** join/#asterisk Hmmhesays (n=Neg@24-117-135-28.cpe.cableone.net)
19:36.53Zodiacalcan differnt brands of phones and even different protocals view/update hints?
19:37.01Zodiacali.e. cisco sccp with a polycom sip?
19:37.27Zodiacalthe hints are managed by asterisk right so it should really mater. or am i dreaming?
19:37.42trelane_g__, consider what it'd cost to carpet all 4 walls in a room
19:37.46Zodiacalshould = shoudn't
19:38.05g__trelane: if their scraps, it would be a labour-only cust.
19:38.09g__cost, rather.
19:38.21Zodiacalwhy carpet a wall?
19:38.24trelane_g__, where do you know of to get free carpet scraps that's sufficient to do this?
19:38.30trelane_Zodiacal, acoustics
19:38.38trelane_Zodiacal, carpet kills echo
19:38.43Zodiacaltrelane i know someone that used egg cartens
19:38.48*** join/#asterisk Assid (i=assid@203.115.83.215)
19:38.49Zodiacalfor a band
19:38.59Zodiacalnailed them into plywood and hung that
19:39.00g__trelane: any carpet place throws out tons of the stuff during installation.  I'm sure if you ask around you could get enough.
19:39.02Zodiacalworked ok
19:39.38eKo1carpet scraps + plywood == sound proof wall?
19:40.00Zodiacalegg cartens + plywood
19:40.03Zodiacaltheres somthing about the shape
19:40.04g__Zodiacal: I've heard that one too.. I've also heard towels.. and someone recommended hanging a Duvet on a line.
19:40.05trelane_eKo1, not sound proof, echo canceling
19:40.19eKo1echo canceling?
19:40.31trelane_eKo1, I'm talking, it echos off the wall, makes the room fscking loud
19:40.50trelane_downsides: carpet retains heat, carpet looks odd on walls
19:40.54Hmmhesaysmattresses
19:40.59trelane_Hmmhesays, indeed
19:41.08Hmmhesaysno joke
19:41.21[TK]D-FenderDovid : thsoe files don't tell me anything bout the XML contact directory you are working on.
19:41.23Zodiacalany ideas about my 'hint' question?
19:41.39*** join/#asterisk m4rkl4r (n=markp@outboundemail.uneta.com)
19:42.14trelane_g__, one more note, carpet on the walls depending on zoning and fire code may not be legal, consult a construction/archetectural type before doing it
19:42.20stoffell_hZodiacal: u want to know if polycom's and cisco's work well in 1 setup ? (even with hints)
19:42.26DovidTK: so what do u need ?
19:42.31Zodiacalstoffell_h yep
19:42.31Dovidwhat files ?
19:44.03Dovid[TK]D-Fender: wut u need ?
19:44.04stoffell_hZodiacal: it'll work.. but... i wouldn't do it...
19:44.41Zodiacalwhy not
19:45.06Zodiacalthey will rarly talk with each other
19:45.14Zodiacalmostly to transfer calls etc..
19:45.27[TK]D-FenderDovid : You need to make your directory file per-phone as <mac>-directory.xml
19:45.30*** join/#asterisk [Airwolf] (n=airwolf@dsl51B67BC5.pool.t-online.hu)
19:45.33Dovidi am doing that
19:45.49stoffell_hZodiacal: if I want to 'manage' 100 phones.. i prefer them to be 100 phones of brand X, not 30 phones brand X and 70 brand Y.. just for simplicity.. and manageability...
19:46.00Zodiacalic
19:46.30stoffell_hZodiacal: standardizing is, in this case, a good thing to do.. :)
19:46.30trelane_Zodiacal, pick one, or the other, don't do both.  You will regret it later
19:46.39[TK]D-FenderZodiacal : Yes, * maintains status based on ITS awareness.  Meaning if you DIRECTLY call a SIP phone its "monitoring" then * will not know the call is in progress.
19:47.09Zodiacaldirectly call?
19:47.10[TK]D-FenderZodiacal : Anything * is responsible for it propagates status for as well as the channel driver permits (can't speak for SCCP, only SIP).
19:47.22Dovidfiles are called 0004f2052338-directory.xml
19:47.30[TK]D-FenderZodiacal : To which I can say that an IP 601 + 3 Attendent modules = really cool.....
19:47.33Dovidand 0004f2052338.cfg
19:47.40[TK]D-FenderDovid : then it should pick them up.
19:47.47Dovidit isnt, can u think of y ?
19:47.52[TK]D-FenderDovid : pastebin 0004f2052338-directory.xml
19:48.05stoffell_h[TK]D-Fender: lol, i just got my 430 (since a few days), can't wait to start playin' with it
19:48.42[TK]D-Fenderstoffell_h : I got mine last week.... yes, VERY cool, and the firmware is MCUH zippy-er than any of the others, and boots much faster as well.
19:49.05[TK]D-Fenderstoffell_h : I like the LCD skin they use and wish they ported its look&feel to the others.
19:49.11DovidTK: http://pastebin.ca/91639
19:49.16[TK]D-Fenderstoffell_h : Feels very CCS-ish
19:49.45[TK]D-FenderDovid : Looks OK, guess you should verify the user rights on it.
19:49.48eKo1What does this message actually mean: Got SIP response 488 "Not Acceptable Here" back from 192.168.53.29
19:50.02stoffell_h[TK]D-Fender: nice! i just had the time to power it up for 10secs, the blinkin' lights are cool, gotta look into the rest.. rrrr ;)
19:50.05eKo1Wh is it not acceptable?
19:50.10DovidTK: do the files have to be set as read only ?
19:50.55[TK]D-FendereKo1 : Codec mis-match
19:51.18[TK]D-FenderDovid : I suggest 644 to the user.
19:51.28Dovidhmm
19:51.31Dovidnow it just worked
19:51.33Dovidok
19:51.45Dovidthis time i did a hard reboot
19:51.47Dovidoh well
19:51.51Dovidlooks like its workin
19:51.52Dovidthanks
19:51.52[TK]D-FenderDovid : you DO have to reboot to take changes naturally.
19:52.03Dovidreboot ?
19:52.21Dovidturn on phone, get new files, turn off phone and then it will have it ?
19:52.33Dovidor turn on get files and then there ?
19:52.49Dovidi did hard reboot as oposed to just rebooting from phone menu
19:52.52[TK]D-Fenderturn off phone, mod the file, turn on phone, done
19:53.04Dovidok
19:53.22Dovidlet me try again, just to be sure its workin, thanks for all the time
19:53.44eKo1[TK]D-Fender: right. thanks
19:54.17file[TK]D-Fender: how are we today?
19:54.21[TK]D-FenderI should get a job as paid remote Polycom support :)
19:54.39[TK]D-Fenderfile : we are well....
19:54.46Dovidhehe
19:54.48Dovidyes u should
19:55.04[TK]D-FenderI mean I pimp them enough you'd THINK I was paid to....
19:55.10Dovidlol
19:55.16fileadmit it - you are!
19:55.19Dovidi learn from people like ,
19:55.21jbalcombdovid: you might need to reset the local config on the phone as well
19:55.38Dovidi pass around the luv. was in here 2 hours workin wit some one, givin em help
19:55.43[TK]D-Fenderjbalcomb : not for phone directory.  if it doesn't match it trusts the server.
19:55.48Dovidjbalcomb: what do u mean ?
19:55.58[TK]D-Fenderjbalcomb : not like a config override requires.
19:56.05jbalcomb[TK]D-Fender ah, ok. thanks polycom guy.
19:56.20[TK]D-Fender"pseudo"-Polycom guy!
19:56.25stoffell_h;)
19:56.31stoffell_hpseudo ? :)
19:56.38jbalcomb[TK]D-Fender how many shares of thier stock do you own?
19:56.39Dovidnow its workin :):):):)
19:56.43Dovidlol
19:56.48[TK]D-Fenderjbalcomb : none :(
19:56.58jbalcomb[TK]D-Fender LIAR!!
19:57.00Dovidok, now that i have that, anyone wana help with polycom paging ?
19:57.09*** join/#asterisk juanjoc (n=juanjoc@200.73.189.82)
19:57.10[TK]D-Fenderjbalcomb : HEY!  There's something for you to make!  Polycom Buddly list editor!
19:57.22jbalcomb[TK]D-Fender maybe that'll be next
19:57.27[TK]D-Fenderjbalcomb : That is worth GUI-ing
19:57.37jbalcomb[TK]D-Fender everything i do is worth doing pal.
19:57.51[TK]D-Fenderjbalcomb : As long as you're being PAID to ;)
19:58.15*** join/#asterisk Druken (i=Druken@CPE00121716da99-CM00159a090acc.cpe.net.cable.rogers.com)
19:58.46Drukenok... i feel dumb, but am i wrong in thinking that nat=yes is correct?
19:58.49syzygyBSDthat is the "worth" part
19:58.54*** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net)
19:59.01syzygyBSDdepends on if you have a nat...
19:59.12Drukensyntax....
19:59.27syzygyBSDI have seen that in the tutorials, but I prefer nat=1
19:59.34[TK]D-FenderDruken : Depends on the question ;)
19:59.53Druken[TK]D-Fender: i have nat=yes in my sip.conf
20:00.05[TK]D-FenderDruken : I believ it accepts either, but ask yourself this... what does NAT=3 mean? :)
20:00.08Drukenyet.. the damn sip show peers isn't showing it as nat
20:00.23[TK]D-FenderDruken : pastebin....
20:01.36[TK]D-FenderI need an IP 601 to keep up with all the customers I'm maintaining!
20:02.29Dovidi luv the 601
20:02.29Drukenhttp://pastebin.ca/91651
20:02.33Dovidi jus need to do paging
20:03.35[TK]D-FenderDruken :  WIP                        (Unspecified)    D   N      255.255.255.255  0        UNKNOWN
20:03.49Drukencorrect
20:03.56[TK]D-FenderDruken : I might think that it would not fill in that field becuase perhaps the phone isn't registered....
20:04.09[TK]D-FenderDruken : Reg it and see what happens
20:04.26Drukenwell of course it's not registered... cause it's behind the nat and needs the flag set
20:04.42*** join/#asterisk tsurk0 (n=tsurko@85.187.160.157)
20:04.51Drukendamn phone has been trying to register for hours now...
20:05.12*** join/#asterisk af_ (n=af@ip-170-209.sn1.eutelia.it)
20:05.30[TK]D-FenderDruken : Do other phones of yours work behind NAT currently?
20:05.50Drukendunno, don't have any at the moment...
20:06.01[TK]D-FenderDruken : pastebin your [general] section.
20:06.23Drukenit's only 4 lines...
20:06.27*** join/#asterisk iCEBrkr (i=icebrkr@69.9.167.70)
20:06.38*** join/#asterisk [Airwolf] (n=airwolf@dsl51B67CB3.pool.t-online.hu)
20:06.56Drukenhttp://pastebin.ca/91657
20:08.59[TK]D-FenderDruken : Does your * server have a public IP?
20:09.33Drukenuhmm... yes.. i'm not a complette moron :)
20:10.16stoffell_hhm, what's the capital of canada?
20:10.21*** join/#asterisk Unistim_junky (n=rover@c-71-56-28-13.hsd1.ga.comcast.net)
20:10.23mitchelocottowa!
20:10.27[TK]D-FenderDruken : on sip debug do you see the incoming registration attempt?
20:10.28Drukenuhmm... toronto? :)
20:10.35mitchelocquebec!
20:10.44Drukenon sip debug i see so much shit, who can read it?
20:10.48stoffell_hlol, 3 diferent answers... :)
20:10.49[TK]D-Fendermitcheloc : Actually the first capital of Canada was MONTREAL <-
20:10.54mitchelocbetter yet, canada has a government?
20:11.03jbroomesyrupistan
20:11.04stoffell_hi'm from europe, sorry i don't know...
20:11.21[TK]D-Fenderstoffell_h : Ottawa, Ontario
20:11.41stoffell_h[TK]D-Fender: thanks, a reliable source ;) (google confirms)
20:11.42[TK]D-FenderDruken : no excuses!
20:12.01mitchelocooh, was i right with ottawa?
20:12.02Drukenpfft... i think it's a great excuse... hehe
20:12.27*** join/#asterisk Dr-Linux (n=Linux@202.59.73.131)
20:12.28mitchelocDruken, i agree, good point, nobody *can* read it, yet everyone points to it...
20:12.41[TK]D-FenderDruken : BTW some routers screw the hell out of SIP regardless and you may be up a creek.  What kind of phone?
20:12.47mitchelocyou need special elite speed reading skillZ
20:13.00DrukenWIP300, behind an rt31p2
20:13.02[TK]D-Fendermitcheloc : I do OK on sip debug, and I'm far from experienced compared to many here
20:13.17*** part/#asterisk TESTER2 (n=Cyber@modemcable082.42-81-70.mc.videotron.ca)
20:13.24Unistim_junkyWhere can I see a list of commands I can do by calling asterisk with -rx?
20:13.30[TK]D-FenderDruken : OH!!!! WIP300 doesn't work behind NAT!  its a serious POS.  forget the router... its the PHONE.
20:13.50Dr-Linuxi'd agree SIP seems nothing, but it's a huge game
20:13.50Drukenhmm.. well it worked last night...
20:13.56[TK]D-FenderSIP WiFi = SUCK
20:14.15mitchelocsip is a virus
20:14.20[TK]D-FenderDruken : from behind NAT?  I've read up on their lack of NAT functionailty.
20:14.30mitchelocit's like a weed that grew up in the voip space and should be stamped out
20:14.36Drukenthat's the only way it could have run in my house...
20:14.42[TK]D-Fendermitcheloc : Life is a sexually transmitted disease which is in all cases FATAL.
20:14.45Drukeni don't have a block of ip's here...
20:14.55stoffell_hlol mitcheloc
20:14.56[TK]D-FenderDruken : Hrm.... COULD be the router then...
20:14.57*** join/#asterisk AJaymn (i=AJmn@70.59.126.197)
20:14.59mitcheloc[TK]D-Fender, thanks for ruining my day fender
20:15.19Drukenmaybe i'll just setup a server for my house...
20:15.28Drukenfucken pain in the ass
20:17.00Dr-Linuxlol
20:17.01[TK]D-FenderPITA.... good for sandwichs, bad for IT....
20:17.10eKo1hehehe
20:18.25jbalcombfeh, i now have several phones that only have one way audio.
20:18.38Unistim_junkycan a user be unregistered via the CLI?
20:18.48jbalcombthey can hear the person they call but the person cant here them. also, no one can call them.
20:18.49Assid[TK]D-Fender: how do you know which routers act stupid with nat?
20:19.03[TK]D-FenderAssid : Trial & Frustration.
20:19.14Hmmhesaysok this as5300 is still giving me hell
20:19.14AJaymnAnyone know of a VoIP provider that allows CID info (spoofing)  ???
20:19.17Dr-Linuxjbalcomb: codec's issue?
20:19.28Dr-Linuxjbalcomb: what CLI says?
20:19.30[TK]D-FenderAJaymn : Apparently Broadvoice....
20:19.35jbalcombDr-Linux: i'll double check but everything is ulaw here
20:19.55Unistim_junkyjbalcomb :  check your firewall
20:20.02Dr-Linuxjbalcomb: what's softphone/hardphones users have?
20:20.12[TK]D-Fenderjbalcomb : Would all of the defect phones happen to carry the name "Grandstream"?
20:20.20jbalcombUnistim_junky: no firewall internal and it just started happening about two hours ago.
20:20.40jbalcomb[TK]D-Fender: nope, one x-ten lite and one polycrap ip501
20:20.43Unistim_junkyjbalcomb: could be routing.
20:20.49[TK]D-Fenderjbalcomb ;)
20:20.51AJaymn[tk]d-fender  ive never been able to get it to work with them
20:20.53jbalcomb[TK]D-Fender ;)
20:20.57[TK]D-Fenderjbalcomb : All local LAN?
20:21.02jbalcomb[TK]D-Fender certainly
20:21.16[TK]D-Fenderjbalcomb : Hrm
20:21.24jbalcomb[TK]D-Fender i thought at first there was something about dtmfmode...
20:21.36[TK]D-Fenderjbalcomb : No, that should all be OOB anyways...
20:21.54jbalcomb[TK]D-Fender i swicthed a gxp-2000 with a polycom and had to update the sip.conf.. maybe thats when it starts.. with the 'sip reload'
20:21.59[TK]D-Fenderjbalcomb : 1 way typically runs to NAT... or perhaps a defective handset / headset
20:22.00Dr-Linux[TK]D-Fender: what a call listen, when call to those users?
20:22.24HmmhesaysAnyone good at cisco as5300's?
20:22.32[TK]D-FenderDr-Linux : Talks does funny Yoda... hhmmmMMMMMMM?!?!?!
20:22.46jbalcomb[TK]D-Fender: hey, maybe this is the first reload i've done since you had me change those settings?
20:23.13[TK]D-Fenderjbalcomb : pastebin your general section for me, and list the IP's of the devices involved.
20:23.13Dr-Linuxjbalcomb: wht asterisk version?
20:23.29jbalcombDr-Linux 1..5
20:23.45[TK]D-Fender1.5?!  OMG, he's ahead of us all!!!
20:23.53[TK]D-FenderEven DIGIUM!
20:24.00Dr-Linuxlol
20:24.13Dr-Linuxhe means 1.2.5
20:24.18Dr-Linuxi believe :P
20:24.41jbalcombDr-Linux: i developed my own source tree and moved on.. ;)
20:24.52[TK]D-Fenderjbalcomb : Probably more like POISON OAK ;)
20:25.07Dr-Linuxjbalcomb: then you maybe have alot of other problems as well :P
20:25.18jbalcombhaha.. yeah, perhaps my terrible code could be viewed as such
20:25.40Dr-Linuxjbalcomb: did you try restarting your asterisk?
20:26.04[TK]D-Fenderjbalcomb : pastebin qwuick before I'm outta here...
20:26.05jbalcomb[TK]D-Fender AH HA!! the receptionist PC running x-ten lite is on 192.168
20:26.18[TK]D-Fenderjbalcomb : you can add multiple
20:26.23[TK]D-Fender"localnet" clauses....
20:26.26jbalcomb[TK]D-Fender http://pastebin.ca/91685
20:26.36Dr-Linuxjbalcomb: there could be many reasons i.e NAT, Firewall, Qaulify, Codec uncompatibility ... and so
20:27.11[TK]D-Fenderjbalcomb : http://pastebin.ca/91688
20:27.19rob0Well, I have * set up as a softphone now ... works, but it's still a softphone :)
20:27.21[TK]D-Fenderjbalcomb : ywc
20:27.34jbalcombywc?
20:27.40[TK]D-FenderYou're WelCome
20:27.48jbalcombah, ty
20:27.58*** join/#asterisk TommyTheKid (n=tommythe@mpk-edge.cto.sunit.net)
20:28.01gandhijeeanyone know why when i run my ztcfg i only get 25 out of my 30 channels for TDMoE???
20:28.05rob0Really laggy even calling a Zaptel phone on the same Ethernet segment.
20:28.26Dr-Linuxgandhijee: ztcfg -vvv
20:28.27rob0like about 1 second lag times.
20:28.55gandhijeei know
20:28.59Dr-Linuxgandhijee: otherwise check your zaptel.conf
20:29.03gandhijeeit only configures 25 lines
20:29.06gandhijeeits standard.
20:29.14TommyTheKidDoes anyone have any advice on dual core opterons verses dual physical CPUs? does it matter? .. plan to use a quad PRI (maybe 2) in it as a conferencing server
20:29.15gandhijeeonly diff is its running on Xscale/IXP
20:29.28gandhijeebut that shouldn't matter for that driver, it doesn't interface to the PCI bus
20:29.33Dr-Linuxgandhijee: what about your zaptel.conf?
20:30.10Unistim_junkycan a user be registered and unregisted via the Asterisk CLI>
20:30.28[TK]D-Fenderok, heading home, BBIAB
20:30.31DovidTK: having problems with the wiki and on paging
20:30.44Dr-Linuxjbalcomb: you have only ulaw
20:30.47Dr-Linuxadd some more
20:30.57jbalcombDr-Linux all the phones are ulaw
20:31.00Dovidwhat do i need to edit in sip.conf to have the phone accept paging ?
20:31.09Unistim_junkybalcomb:  If its all internal why do you have an externalip setup
20:31.13Dr-Linuxilbc takes lot of CPU and it's slim, that's why skype uses
20:31.14Dr-Linux:)
20:31.41Unistim_junkyDr-Linux:  can a user be registered and unregisted via the Asterisk CLI>
20:31.42TommyTheKidso, do we have chan_skype.so yet? :)
20:31.43Dr-Linuxjbalcomb: that would be not a pain if you try more ..
20:32.05*** part/#asterisk JohnJacob (n=dhorner@pool-71-127-102-43.aubnin.fios.verizon.net)
20:32.24gandhijeenm, i figured it out
20:32.37Dr-LinuxUnistim_junky: hhm... does user's account axist?
20:32.47Unistim_junkyyes
20:33.02Unistim_junkyso they are in sip.conf
20:33.28Unistim_junkyso I want to be able to unregister / register via CLI
20:34.01gandhijeei had it as 2-31
20:34.14gandhijeei forgot that the TDM400 takes 1 to 4 regardless
20:34.28Unistim_junkyDr-Linux:  I figure if I can do it on CLI> then I can right a cron job to do it.
20:35.08gandhijeei dont need to load asterisk on the side that is exporting my TDMoE spans do it?
20:35.10gandhijee*i
20:35.21Dr-Linuxhhm..
20:35.32Dr-LinuxUnistim_junky: i'm trying but today my browsing sucks
20:35.42Dr-LinuxUnistim_junky: did you try google?
20:35.50tzafrir_laptopTommyTheKid, don't count on such a chan_skype any time soon
20:36.04TommyTheKidhehe
20:36.05tzafrir_laptopTommyTheKid, BTW: have you tried google talk?
20:36.20TommyTheKidnope, as far as I know, its d0ze only
20:36.27Unistim_junkyDr-Linux: google, voip-info, digium... I figured I must be searching wrong keywords
20:36.44tzafrir_laptopOn linux there are some other clients
20:37.06tzafrir_laptopFor chat alone you can use any jabber client
20:37.43TommyTheKidmy real question was regarding Dual Core CPUs, like the Opteron 180, does dual core vs dual physical CPU make a difference with asterisk/digium te412p? Also would it be better or worse to do 64 bit?
20:38.15TommyTheKidyea, I use Psi or ichat on my googleTalk account, but its not voice :)
20:39.12tzafrir_laptopdual core is basically dual CPU.
20:39.30tzafrir_laptop(not to be mixed withhh "twice as fast")
20:39.39TommyTheKidthat was my view on it.. its not like HT :)
20:39.47*** join/#asterisk sevard (n=sev@adsl-71-129-115-244.dsl.irvnca.pacbell.net)
20:39.54tzafrir_laptopTommyTheKid, latest development psi should support using Jingle
20:40.02tzafrir_laptopHaven't tried it
20:40.11clive-i was wondering that also, does a dual core 3 ghz compare with a xeon 2 x 3 ghz ?
20:40.33TommyTheKidI fon
20:40.35*** join/#asterisk pa (n=Paolo@unaffiliated/pa)
20:40.59TommyTheKiduh, I don't use Intel, but I think the Xeon CPUs tend to be a bit more "server class" (higher cache, etc) than the CoreDUO?
20:41.38*** join/#asterisk VoIPMasta (n=John@201.160.17.205.cableonline.com.mx)
20:41.47gandhijeethere is a CoreDUO based Xeon now
20:41.57TommyTheKidah
20:42.09lokkjuwell, the Core Duo is coming out in multiple versions - a mobile, a workstation, and a server class chip, if I remmember right
20:42.10gandhijeewe just devel'd a platform for EMC using that arch
20:42.10VoIPMastaHi there, I'm setting an IVR but for an unknown reason it's not detecting the numbers I dial to reach the extensions, any ideas?
20:42.20*** join/#asterisk tsurk0 (n=tsurko@85.187.160.157)
20:42.47rob0I must say though, I feel considerably geekier with a * installed on /dev/laptop. Highly recommended to anyone who feels the need for increased geekitude.
20:43.05Strom_Crob0: I have two laptops with asterisk installed
20:43.08Strom_Ci win
20:43.22rob0yes you do ... I wasn't trying to compete with you :)
20:43.26gandhijeei think i have managed to port zaptel to Xscale for this board we have at this place
20:43.32VoIPMastarob0: I 've had asterisk running on my laptop for at least 1 year
20:43.49gandhijeeboard has x86 and IXP/Xscale on the same PCB sharing the PCI bus
20:43.54rob0Now let's persuade Digium to make CardBus Zaptel adapters.
20:44.01Unistim_junkyrob0: I have asterisk running on a 3850 IPAQ
20:44.23TommyTheKidDigium specific Q: The te412p (we are testing) has quad T1/E1 capability with DSP/echo canceling/etc... so that must mean it can handle 30 line E1's which means that we have like 12 (actually 16) DSPs that are not in use, can we use those for other uses? :)
20:44.24Unistim_junkyI think I win
20:44.28VoIPMastaUnistim_junky: I have also asterisk running on an embedded WRAP
20:45.16Unistim_junkynice, what is WRAP
20:45.29VoIPMastahave you seen SOEKRIS embedded mother boards?
20:45.37rob0wireless router appliance?
20:45.38Unistim_junkyo... ok
20:45.41Unistim_junkysorry..
20:45.45VoIPMastasomething like that
20:46.00VoIPMastaa small Geode 266 Mhz CPU, with a 256MB CF card
20:46.06gandhijeeVoIPMasta: IXP465 gonna test out the TDM interfaces soon
20:46.19VoIPMastait does also have 128 Mb RAM and 2 100mbps NICs
20:46.37Unistim_junkyDo you guys have any comments on registering / unregistering a user via CLI
20:47.06*** join/#asterisk TripleFFFF (n=TripleFF@147-102.mc.cite.net)
20:47.12TripleFFFFwhats most stable version of asterisk ?
20:47.12VoIPMastaand does anyone have any ideas on why my IVR isn't detecting the digits I dial to access extensions?
20:47.21VoIPMastaTripleFFFF: go for the latest
20:47.23TripleFFFFthat could be prodction wised
20:47.28jbalcombVoIPMasta: perhaps a dtmfmode problem?
20:47.29TripleFFFFwell im getting weird shit
20:47.44Unistim_junkydtmf...
20:47.56VoIPMastaI have 1.2.9.1 in a production environment
20:48.01VoIPMastaand I haven't had any problems with it
20:48.17*** join/#asterisk piwi (n=piwi@80-43-91-224.dynamic.dsl.as9105.com)
20:48.18Unistim_junkyhave you changed the loc. your calling from.
20:48.20VoIPMastaUnistim_junky: Yup, I think it;s something related to dtmf modes or something
20:48.24piwihi there
20:48.33*** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239)
20:48.43VoIPMastaUnistim_junky: yes, I've tried from my cell phone and from a SIP device using a regular analog phone
20:48.50Unistim_junkyDepending on the where you call from they may have a different SBC
20:49.09Unistim_junkytry changing dtmf to auto  ....I think it is...
20:49.25piwihey, is anyone using asterisk with freephonie.net?
20:49.44VoIPMastahowever I'm setting up this IVR using a SIP-provisioned DID
20:49.48Unistim_junkya different gateway...
20:49.53VoIPMastaso I don't have a "context" in sip.conf for this DID
20:49.58VoIPMastajust a register line
20:50.10jbalcombhow does one reboot the cisco7940 from the keypad?
20:50.17Dr-Linuxhow many bucks a voip guy get paid for an hour in USA?
20:50.27Unistim_junky* + 6 + settings
20:50.38VoIPMastaDr-Linux: depends on the tasks and knowledge
20:50.44bkw_you guys want a laugh
20:50.44bkw_http://crazytelemarketer.ytmnd.com/
20:51.05Dr-LinuxVoIPMasta: normally?
20:51.41VoIPMastaDr-Linux: paid by the hour (something like a freelancer) or a signed contract?
20:52.00*** join/#asterisk ariel_ (n=ariel_@74.8.35.2)
20:52.04VoIPMastaUnistim_junky: where can I set up the dtmf mode considering that I'm having this DID forwarded by a DID provider?
20:52.21Dr-Linuxfreelancer
20:52.57Unistim_junkygeneral section of sip.conf
20:52.57Dr-LinuxVoIPMasta: also tell me contract
20:54.06Unistim_junkyVoIPMasta: What do you mean being forwarded?
20:54.16VoIPMastaUnistim_junky: it worked :)
20:54.22Unistim_junkyThe user is hitting your IVR right?
20:54.33znoGdo many people use call parking?
20:54.33Unistim_junkyVoIPMasta: Cool
20:54.55Unistim_junkyNow, If Someone would just help me........
20:55.17Dr-LinuxVoIPMasta:?
20:55.18NDTIs the uniqueid...call specific or channel specific?
20:56.09Unistim_junkyuser meaning user in sip.conf
20:56.11VoIPMastaDr-Linux: from what I know: freelancer ~30.00 / hour.
20:57.14*** part/#asterisk TommyTheKid (n=tommythe@mpk-edge.cto.sunit.net)
20:57.24VoIPMastaUnistim_junky: I haven't registered a user using the CLI, so I'm afraid I'm unable to help you
20:57.36Unistim_junkyNoProb.. me neither
20:57.52Dr-LinuxVoIPMasta: aww someone ofered me $12/hr
20:57.56VoIPMastaUnistim_junky: to be honest, I didn't think it could be done, but it also depends on which protocol you're using
20:58.09VoIPMastaDr-Linux: but what's your expertise/experience?
20:58.31*** join/#asterisk fulgas (n=fulgas@a81-84-116-1.cpe.netcabo.pt)
20:58.35daysmen3_HELP anyone see a problem with syntax here - exten => s,n,Set(EXTNUM=${IF($["${EXT:-1}" = "#"]?${EXT:1}:${EXT})})
20:59.18Dr-LinuxVoIPMasta: they need me for asterisk and hardphones setup ..
20:59.25Unistim_junky[TK]D-Fender: What do you think?  Can a sip user be registered via the CLI
20:59.31VoIPMastaDr-Linux: and how experienced are you at it?
20:59.32Unistim_junkysetVar
20:59.52Unistim_junkydaysmen3_:
20:59.55*** part/#asterisk fulgas (n=fulgas@a81-84-116-1.cpe.netcabo.pt)
21:00.17Dr-LinuxVoIPMasta: i'm already running 4 asterisk servers located in USA
21:00.44ariel_wow 4 servers.
21:00.44Dr-LinuxVoIPMasta: 2 years with network field and 8 months with asterisk
21:00.57HmmhesaysI need as5300 help
21:00.58Hmmhesaysargh
21:01.20Dr-LinuxVoIPMasta: they want me, bcoz i'm in/from Pakistan.
21:01.29Unistim_junkyVoIPMasta: You a Cisco man?
21:01.42VoIPMastaUnistim_junky: I've played with cisco for some time
21:01.46[TK]D-FenderUnistim_junky : Not AFAIK
21:01.50daysmen3_im getting a - expecting $end; """ = "#"
21:01.58ariel_Hmmhesays, what is the issue with the unit?
21:02.02VoIPMastaDr-Linux: Well, but you're overseas, that also means a lower wage
21:02.21Dr-LinuxVoIPMasta: overseas?
21:02.23VoIPMastaDr-Linux: a lot of companies in the US look for people to work from home (and from another country) because it's cheaper
21:02.33*** join/#asterisk ivanfm (n=ivanfm@201.52.129.236)
21:02.39rob0US$30/hour sounds low for professional * consulting. I charge much more for general Linux work.
21:02.40Dr-LinuxVoIPMasta: correct
21:02.56Dr-LinuxVoIPMasta: i'm already working for a US company
21:03.06VoIPMastarob0: I do also charge more for a lot of things, but a "voip guy" like Dr-linux described gets about 30/hr
21:03.10[TK]D-FenderCisco certs get $100/h on consulting.
21:03.20ariel_not many here in the US charge less then 50 per hour. Most are over 100.00
21:03.30[TK]D-FenderI typically charge $40/h
21:03.43ariel_[TK]D-Fender, wow your cheap.
21:03.44VoIPMastaariel_: there's a diff between a pro and an amateur consultant
21:04.06VoIPMastathere are literally thousands of geek kids that have been messing with asterisk for some months and offer "consulting services"
21:04.08rob0What's the difference? The amateurs know more? :)
21:04.09Dr-LinuxVoIPMasta: actually i'd be very happy if i get even $10/hrs in pakistan
21:04.30VoIPMastarob0: sometimes the certificate grants you the right to charge more
21:04.31ariel_VoIPMasta, hummm wonder if they people think that there only amateur.
21:04.46VoIPMastarob0: like [TK]D-Fender said, a cisco certified consultant charges over 100/hr
21:05.05VoIPMastathat's why certifications are so useful
21:05.11*** part/#asterisk sevard (n=sev@adsl-71-129-115-244.dsl.irvnca.pacbell.net)
21:05.14*** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl)
21:05.35VoIPMastaDr-Linux: don't worry, I'm from Mexico and wages here are also in the low side.
21:05.39[TK]D-Fenderariel_ : But not "easy" ;)
21:05.53ariel_[TK]D-Fender, I see.
21:05.55Dr-LinuxVoIPMasta: they also want me to configure webserver/iptables and trouble ticket system for them on linux
21:06.09Dr-LinuxVoIPMasta: hehe cool
21:06.15VoIPMastaDr-Linux: then charge them for ticket/event instead of charging by the hour
21:06.15ariel_VoIPMasta, I see, Mexico deals more in Peso then dollars...
21:06.30VoIPMastaariel_: not really, most of my rates are in dollars
21:06.59Dr-Linuxi see
21:06.59Dr-LinuxVoIPMasta: what you do?
21:06.59VoIPMastaariel_: on IT industries most rates are in US
21:06.59VoIPMastaariel_: on IT industries most rates are in USD
21:07.08lesouvage"show queues" output is : queue1       has 0 calls (max unlimited) in 'ringall' strategy (0s holdtime), W:0, C:1, A:0, SL:100.0% within 0s  What does the Sl:100% within Os means?
21:07.17VoIPMastaDr-Linux: I run 4 datacenters, plus manage some voip devices (gateways) and also do some coding
21:07.26Unistim_junkyservice level
21:07.32Dr-LinuxVoIPMasta: good good
21:07.34AJaymnAnyone know of a VoIP provider that allows CID info (spoofing)  ???
21:07.37Unistim_junkylesouvage:
21:07.53ariel_AJaymn, voipjet does
21:07.53VoIPMastaAJaymn: there are some in Europe that allow you to spoof it
21:08.01Unistim_junkyAjaymn:  telasip & voicepulse
21:08.04VoIPMastaariel_: but sometimes voipjet doesn't even pass on CID
21:08.15Strom_Chello
21:08.24*** join/#asterisk pbx1 (n=pbx1@58.69.92.3)
21:08.26ariel_VoIPMasta, yes if  you send them the name. If you just send them the numbers there fine with it.
21:08.28Dr-LinuxVoIPMasta: my later plans are outsourcing .. as i just started to make my a few sites i.e www.syednetworks.com , redhat.pk , tele.pk etc
21:08.55AJaymnAny with Unlimited US calling?
21:08.59Unistim_junkyslashdot
21:09.04lesouvageUnistim_junky: ?
21:09.07*** join/#asterisk Dibbler_ (n=Dibbler@dsl-217-155-254-174.zen.co.uk)
21:09.19Unistim_junkylesouvage: service level
21:09.27rob0Unlimited? Limited only by how much you're willing to pay?
21:09.58rob0or are you talking about a flat rate?
21:10.07*** join/#asterisk angler (n=angler@pdpc/sponsor/digium/angler)
21:10.18lesouvageUnistim_junky: you mean 100 % of the incoming calls has been answered without delay?
21:10.28rob0I think the per-minute plans are usually better.
21:10.38rob0but I don't know
21:11.43jbalcombUnistim_junky: thank you
21:12.59rene-are the grandstreams GSM capable?
21:13.03rene-that would be cool
21:13.42Unistim_junkyjbalcomb:  glad to help
21:13.58Unistim_junkylesouvage: Yeah, something like that
21:14.47Unistim_junkylesouvage:  If I remember correctly you say all calls should be answered within like 5 minutes, and that pecentage shows how many were actually answered within that amount of time
21:15.19Unistim_junkyso if a manager looks at it they can see whether or not the agents are jerking off or not
21:15.33rene-Voipmasta: do you work on your own or are you employed
21:15.37rene-I live in Mexico too
21:15.55VoIPMastaI am a shareholder of the company I work for
21:16.01VoIPMastatechnically I'm the CEI
21:16.02VoIPMastaCEO
21:16.09rene-CTO?
21:16.14rene-ah
21:16.16VoIPMastanope, CEO
21:16.23rene-where in mexico are you based?
21:16.27TripleFFFFman since when to PH sing ?
21:16.27VoIPMastaI have 51% of the shares
21:16.39VoIPMastarene- We have offices in Mexico City, and Morelos
21:16.40AzraelVoIPMasta: is your company public?
21:16.45rob0Whacha doing in IRC? You should be out making money? :)
21:16.49rene-hahah
21:16.58carrarman 1.2.9.1 is buggy
21:16.59VoIPMastaAzrael: by public do you mean shares offered in an open market?
21:17.00rene-he gets .01 per msg posted
21:17.00TripleFFFFirc is not CEO , only CTO 's ;)
21:17.04AzraelVoIPMasta: yes
21:17.05Unistim_junkyis voip restricted in Mexico?
21:17.17rene-yes
21:17.18TripleFFFFyes mexico sucks for voip.. ask telmex and fox
21:17.21VoIPMastaNot really
21:17.23VoIPMastait's not regulated
21:17.24rene-you can do it for your private use
21:17.26VoIPMastatelmex sucks
21:17.27AzraelVoIPMasta: publicly traded is what i mean
21:17.28TripleFFFFtelmex is ran by fox grey alien friends
21:17.32VoIPMastabut there's not a real regulation for it
21:17.38gandhijeehey anyone know what package newt.h belongs too?
21:17.38Strom_Cdo any of you regularly purchase polycom phones, and if so, where do you purchase them from?
21:17.44TripleFFFFhow real can a 9mm bullet be ?
21:17.56VoIPMastaTripleFFFF: nope, Telmex isn't run by Fox nor any of his friends, but by Carlos Slim (The richest man in Latin America)
21:17.56Unistim_junkyYEs I heard that the goverment sends folks to chat to find out who is illegally routing calls outside of the place.
21:18.06AzraelStrom_C: at the last place i worked i setup the VoIP system for the company.  i got the Polycom phones from voipsupply iirc
21:18.07rene-to say it simply, if they find out you have something as simple as a calling center doing voip calls you get busted
21:18.14NDTWhy the heck does a jumping to a macro from dial command change the uniqueid of the call?
21:18.19TripleFFFFand how you think Slim to Fox relates to our candadian Desmarais and harper/bush ?
21:18.20AzraelStrom_C: actually that was just one of the vendors.  i'd call a bunch out there and see if you can get quantity discounts.
21:18.24VoIPMastarene-: we haven't been busted, and we've been doing voip for over 3 years
21:18.32TripleFFFFhow can you be richest man without any politic connections ?
21:18.35Strom_CAzrael: I just want one for testing purposes
21:18.37TripleFFFFhence his friend
21:18.38TripleFFFFlol
21:18.42AzraelStrom_C: i worked with the Polycom 301 and 501 models - great phones
21:18.50AzraelStrom_C: you can probably call a vendor and setup a 30 day trial
21:18.50rene-voip for private usd is allowed
21:18.55Unistim_junkygandhijee:  lib-newt
21:18.59VoIPMastarene- VoIP for end users is allowed
21:19.02Strom_CAzrael: I've had a bad time trying to configure those phones
21:19.06TripleFFFFlike bush declaring wars everywhere caus the company making the tanks etc weapons is his fathers, and borthers;lol
21:19.09VoIPMastawhat is restricted is to provide VoIP services
21:19.12rene-not if terminates in PSTN
21:19.13TripleFFFFanyhow back to voip
21:19.20rene-ok
21:19.28rene-you can set up vonage for some friend
21:19.29Strom_CAzrael: so I figure I should buy one and learn how to do it properly
21:19.31AzraelStrom_C: yeah?  i set them up for automagical TFTP config, wrote the xml config files, and it all worked great
21:19.34VoIPMastarene- We have a company in the US, and we provide VoIP services "from" the US
21:19.39TripleFFFFim wondering if i should get BE edition of asterisk
21:19.44VoIPMastarene- our company in Mexico just sales devices ;)
21:19.45[TK]D-FenderStrom_C : My rates are very accessable ;)
21:19.54TripleFFFFand if its 1k per server or per company lol
21:20.04gandhijeeUnistim_junky: is that perl related stuff too?
21:20.21Strom_CAzrael: would you mind shooting over some sample xml configs?  all the ones I've run into have been so horribly formatted that they've been impossible to work with
21:20.24Unistim_junkygandhijee: what dow you mean?
21:20.27rene-VoipMasta: i think you walk a thin line
21:20.46AzraelStrom_C: private msg me your email.  i'll be able to get to it tonight.
21:20.53gandhijeewell everything i search for on that returns stuff saying those are perl binds to redhats newt library
21:20.59VoIPMastarene-: I have a "value added services" license from COFETEL as well as some pretty good lawyers
21:21.03gandhijeewhich i am assuming is newlib
21:21.12rene-value added service is not for voip
21:21.15Strom_C[TK]D-Fender: who do you like to buy polycom phones from?
21:21.18Unistim_junkynah I don't think so.
21:21.31gandhijeei've tried to compile and install just newlib (as i need it for an embedded system) but it doesn't seem to work
21:21.31Dr-Linuxhow someone can be Asterisk Certified ?
21:21.32rene-value added service is like $500 USD from COFETEL
21:21.41gandhijeeDCAP
21:21.45*** join/#asterisk teknoprep (n=chris@unaffiliated/teknoprep)
21:21.48rene-by taking DCAP exam and get over 70 in both the written and live test?
21:21.50VoIPMastarene-: but it's enough to provide IVR, voice messaging, and such services
21:21.51Strom_CI just got my dcAP cert today!
21:21.52[TK]D-FenderStrom_C : CCP (Canadian Communications Products) - www.ccpin.com
21:21.56teknoprephey all
21:22.09rene-congrats Strom_C
21:22.12teknoprepwith trixbox.. its basically redhat?
21:22.13Dr-Linuxrene-: how much fees?
21:22.22teknoprepi can use it for other small apps if need be?
21:22.25rene-i only paid for the exam and that was like 200-300 usd
21:22.25[TK]D-FenderStrom_C : They are an auth'd reseller though I know more about them than they do ;)
21:22.33Dr-Linuxrene-: can i take exams being in Pak?
21:22.44rene-i dont know abouit that Dr-Linux
21:22.53rene-ask oej the DCAP guy
21:22.57Dr-Linuxrene-: are you asterisk certified?
21:23.07rene-yes
21:23.12Strom_CDr-Linux: the exam has to be taken in person
21:23.27Strom_CDr-Linux: there is a practical and a written portion
21:23.50Dr-LinuxStrom_C: means i can't do? :(
21:23.58Dr-Linuxi can't go to USA
21:24.28Strom_CDr-Linux: i think there are tests scheduled at astricon europe
21:24.33ariel_Dr-Linux, he is in the EU most of the time.
21:24.37gandhijeeDr-Linux: i belive its offered in europe somewhere
21:24.53[TK]D-Fenderbkw_ : Thats some funny shit....
21:25.01Dr-Linuxaww
21:25.14Dr-Linuxwhat's the info site for DCAP?
21:25.33rene-jbot: google
21:25.34jbotextra, extra, read all about it, google is a search engine found at http://www.google.com/
21:25.34ariel_~ dcap
21:25.36jbotmethinks dcap is Digium Certified Asterisk Professional.  See http://www.voip-info.org/tiki-index.php?page=Asterisk+dCAP
21:25.38Dr-Linuxactually i wanna try Asterisk certification :S
21:25.57filedCAPitated!
21:29.09Dr-Linuxrene-: i can't do DCAP :(
21:30.02piwihey guys I was wondering, I have asterisk running as a daemon on my machine, can I login from ekiga to the asterisk pbx from this same machine?
21:30.56VoIPMastapiwi: yes you should be able to
21:31.09Dr-Linuxgandhijee: no problem maybe someday Digium know that there is a country pk, till then i'll learn some agi :)
21:31.23VoIPMastapiwi: just make sure that asterisk is listening for SIP (i assume you're using SIP in ekiga) on localhost (or your NIC IP)
21:31.51gandhijeelol
21:31.54[TK]D-FenderDr-Linux : Try working on the basics first before asking to be certified.
21:31.57[TK]D-Fender;)
21:32.18*** join/#asterisk SkramX (n=MarkS@admins.sentiensystems.net)
21:32.20Strom_CDr-Linux: it's a difficult test
21:32.23gandhijeeDr-Linux: why can't you go to Europe to take it?
21:32.26Strom_Chi SkramX
21:32.37droopshey stron
21:32.38SkramXHow would I specify a variable for an AGI script in a file that I put in var/spool/asterisk/outgoing/ ?
21:32.41VoIPMastaI can give you a AUFPUOC certificate for free
21:32.42piwiVoIPMasta: which config file do I have to go in to change that?
21:32.44Strom_Chi droops
21:32.46SkramXHiya, Strom_C
21:32.49SkramXHeya, droops too
21:32.53droopshey SkramX
21:32.56SkramXwe know eachother via that BinRev place, eh?
21:32.57droopslong time on see
21:32.57Dr-Linux[TK]D-Fender: well, i just wanted to try :)
21:33.03droopson = no
21:33.17Strom_Chaha yep
21:33.18SkramXdroops: Yeapps
21:33.22SkramXgoing to HOPE?
21:33.23Strom_Cthat good ol binrev thing
21:33.24VoIPMastapiwi: sip.conf
21:33.25Dr-Linuxgandhijee: can't go to Europe/USA/CA .. Visa and shit you know
21:33.29droopsnot me
21:33.43SkramXI think Strom_C already said he wasn't either
21:33.45Strom_CDr-Linux: guess what - I've got my dCAP cert now :)
21:33.49SkramXanyone wanna help me out though? :)
21:33.49Strom_Cer
21:33.52Strom_Cdroops:
21:33.54piwiVoIPMasta: cheers! Im sorry Im total newbie with asterisk, starting from scratch
21:34.00droopsgood deal Strom_C
21:34.01*** join/#asterisk TommyTheKid (n=tommythe@mpk-edge.cto.sunit.net)
21:34.06Strom_CSkramX: whats teh problem?
21:34.14SkramXHow would I specify a variable for an AGI script in a file that I put in var/spool/asterisk/outgoing/
21:34.17*** join/#asterisk teknoprep (n=chris@unaffiliated/teknoprep)
21:34.22droopsdid strom just misspell something
21:34.24VoIPMastapiwi: great, keep up the good work
21:34.46TommyTheKidHow many quad PRI (te411p or 412p) cards can I put in say a quad 2.4GHz Opteron? are there sizing guidelines anywhere?
21:34.53SkramXi have a php script that places a call by putting the .call file in the proper /var directory.. but I want to make a string for Cepstral/an agi script to recite..
21:34.56AJaymnAnyone using ShellShark for VOIP ????
21:34.56Dr-LinuxStrom_C: well, actually certification doesn't matter in my country as very rare guys know here "what is Asterisk"
21:35.08Strom_CDr-Linux: that comment was supposed to be for droops
21:35.11gandhijeeTommy: i with the digium stuff a max of 2
21:35.12Strom_CDr-Linux: damned autocomplete
21:35.19gandhijeei mean i heard
21:35.24TommyTheKidok
21:35.33Strom_CTommyTheKid: I've seen more than two cards in a single box
21:35.38gandhijeesangoma makes an 8 port T1/E1 card
21:35.43Strom_CTommyTheKid: but that can get hairy
21:35.46*** join/#asterisk jm|work (n=jamiem@dsl-217-155-242-137.zen.co.uk)
21:35.46bkw_yes they do
21:35.51SkramXheh
21:36.02SkramXthanks bkw_ !
21:36.07Strom_CSkramX: I asked you to tell me the problem :)
21:36.08droopsi dont know SkramX, but im waiting for the answer
21:36.10TommyTheKidI plan to use it for conferencing too, so how many say 15-20 person conferences can ride onm the same box
21:36.11jm|workhelo :)
21:36.23SkramXStrom_C: well, I need to pass the .call file some sort of variable
21:36.24TommyTheKidI assume with the onboard DSP cards, there isn't much load on the CPU just for the call itself
21:36.26jm|workI have Googled for this but I seem to be going in circles:
21:36.29jm|workapp_dial.c: Unable to create channel of type 'Zap' (cause 0 - Unknown)
21:36.31[TK]D-Fenderbkw_ : Basically the punchline was the *slick* at the end?
21:36.33jm|work:/
21:36.33SkramXI dont know have a specific error or something
21:36.40Dr-LinuxStrom_C: well, whenever i face any problem with any of my asterisk box i ask here, even i solved the problem.. but i wanna see other peoples veiws ..
21:36.43gandhijeeTommy: like i said sangoma makes an 8 port T1/E1 card
21:36.55VoIPMastajm|work: what card are you using?
21:36.59Dr-LinuxStrom_C: and sometime i get very good help
21:37.10jm|workwcfxo0: <Wildcard X100P> port 0xb000-0xb0ff mem 0xee030000-0xee030fff irq 16 at device 8.0 on pci0
21:37.12SkramXbkw_: nice
21:37.13TommyTheKidgandhijee: I saw that, how many sangoma cards can we put in?
21:37.16jm|workFound a Wildcard FXO: Wildcard X100P
21:37.16jm|workZapTel device loaded.
21:37.16jm|workRegistered tone zone 4 (United Kingdom)
21:37.18SkramXi recently got a macmini core duo
21:37.20TommyTheKidwe are looking for density here :)
21:37.27Strom_Cjm|work: please use pastebin
21:37.30Strom_C~pb
21:37.34jbotit has been said that pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/
21:37.34VoIPMastajm|work: is it a digium x100p or a "clone"?
21:37.35bkw_TommyTheKid, sangoma cards do not have IRQ issues
21:37.43bkw_so you can put many in the same box
21:37.44jm|workVoIPMasta: no sure .....
21:37.47jm|worknot*
21:37.56VoIPMastajm|work: where did you buy it?
21:38.02jm|workrecently
21:38.02TommyTheKidwith quad CPU's I seriously doubt I would have IRQ issues anyhow :)
21:38.05[TK]D-FenderTommyTheKid : Not that you'd NEED to at 8ports / card :)
21:38.09VoIPMastawhere, not when
21:38.12jm|workit was quite cheap and "off eBay" so I daresay it's a clone
21:38.17*** join/#asterisk znoG (n=gs@162-148-235-201.fibertel.com.ar)
21:38.19VoIPMastaunder 15?
21:38.24jm|worker
21:38.25Dr-Linux[TK]D-Fender: did you remember my asterisk box? you compiled 7 months ago? :)
21:38.36VoIPMastahow much did you pay for it?
21:38.39bkw_if you come to cluecon you'll get a free T1 card :P
21:38.41[TK]D-FenderDr-Linux : Yup (somewhat)
21:38.46jm|work"It's Real X100P FXO card for Digium Asterisk pbx OEM"  according to the speil
21:38.49jm|workyeah <15
21:39.00bkw_its a freakin INtel 537 Modem
21:39.04*** part/#asterisk Unistim_junky (n=rover@c-71-56-28-13.hsd1.ga.comcast.net)
21:39.04Dr-Linux[TK]D-Fender: heh that server uptime is 212 days :)
21:39.04bkw_the same thing digium was selling for 99 bucks
21:39.10VoIPMastaI'm pretty sure it's a clone with one of the components busted so that it's detected as a "real X100p"
21:39.18jm|work:(
21:39.22bkw_I really like how digium was selling 6 dollars modems to everyone for so long at 99 bucks a pop
21:39.24*** join/#asterisk kcortez (n=kcortez@208.49.103.100)
21:39.24VoIPMastajm|work: use pastebin and paste your zaptel.conf file
21:39.35bkw_they pop two resistors off and slap on a heat sink
21:39.37Dr-Linux[TK]D-Fender: but now i have 4 servers in US datacenter
21:39.44jm|workit only has one line, really
21:39.50mishehubkw_: I'm in the wrong business
21:39.50jm|workfxsks=1
21:39.52VoIPMastabkw_: When I found out that I started buying those modems wholesale
21:39.59bkw_I did too
21:40.02VoIPMastaright now I have over 500 "clone" x100p cards
21:40.06Dr-Linux[TK]D-Fender: i never bother to upgrade 1.2.0 to latest one
21:40.13NDTFor awhile Newegg had em for $5
21:40.20piwiVoIPMasta: sorry but Im still not making it :( I put in general defaultip=localhost and when I launch ekiga it says port for sip is in use and it doesnt even try to login to asterisk and says it failed to register
21:40.29bkw_VoIPMasta, but most people feel that if you're not buying hardware from digium you're morally in the wrong
21:40.41VoIPMastapiwi: tell ekiga to listen on a different port (other than 5060)
21:40.47jm|workVoIPMasta: am I screwed :(
21:40.51bkw_like if you use Asterisk with say OpenVox cards... you're evil!
21:40.57jm|work"if it looks too good to be true: it probably is"
21:41.10VoIPMastabkw_: I'm not selling them, I use them for my company
21:41.27VoIPMastabkw_: and when I bough those modems it was almost impossible to buy digium hardware here in mexico
21:41.29Dr-LinuxStrom_C: i confess i'm not good, but imagine my country. i got award for 2006
21:41.33VoIPMastaso I take no guilt for using them
21:41.44*** join/#asterisk saftsack (n=saftsack@p54A7E68B.dip.t-dialin.net)
21:41.47bkw_I use sangoma hardware for all new projects
21:41.54[TK]D-FenderDr-Linux : and you're PROUD of that?!  Wak up to the "now".  You should fix your setup to 1.2 spec and upgrade for security alone.
21:42.04mishehuI will be using sangoma for all future projects.
21:42.13*** join/#asterisk lokkju (n=lokkju@unaffiliated/lokkju)
21:42.27saftsackbkw_, whats the reason for this decision?
21:42.36bkw_at least with sangoma I can move to freeswitch once we have the functionality complete
21:42.45VoIPMastajm|work: hold on
21:42.46saftsackwhat is freeswitch?
21:42.50jm|workVoIPMasta: okies thanks :)
21:42.53bkw_saftsack, www.freeswitch.org
21:42.55[TK]D-Fenderbkw_ : No zaptel channel driver?
21:42.59SkramXStrom_C: I think I found the solution... ill speak up later
21:43.10Strom_CSkramX: uh, ok
21:43.11bkw_[TK]D-Fender, if you really knew how zaptel worked you would not be using it either
21:43.13Corydon-wsaftsack: because bkw_ initially forked Asterisk, then moved to a completely rewritten project when that fork died
21:43.20[TK]D-Fenderbkw_ : Or is it so far off the beaten track for your uniform structure to be "feasable"?
21:43.23Cresl1npssh.... you're just bitter bkw_
21:43.28Cresl1nyou don't know how it works either
21:43.28bkw_No i'm not
21:43.30[TK]D-Fenderbkw_ : I'll take that as a "yes"
21:43.35bkw_Cresl1n, yes I do
21:43.39Dr-Linux[TK]D-Fender: you guided me alot and i followed you, if you remember i asked you "i hve no problems with current version should i upgrade" you said ...new versions new problems...
21:43.42Cresl1nwhateva foo
21:43.48Cresl1nnot like I do
21:43.50saftsackCorydon-w, is this forked project openpbx?
21:43.53jm|workVoIPMasta: the guy that was selling it has sent me a zapata.conf
21:44.00Corydon-wsaftsack: yeah, openpbx
21:44.08jm|worknot sure if that helps
21:44.16mishehuopenpbx died because it was too hard to maintain I bet.
21:44.17[TK]D-FenderDr-Linux : That doesn't carry when there are KNOW full-on crash exploits and major versions that threaten maintainability.
21:44.17piwiVoIPMasta: cant change the port with ekiga, do you know any good voip client easy to install with ubuntu?
21:44.24bkw_OpenPBX is still alive
21:44.25*** join/#asterisk anthm (n=anthm@adsl-69-211-76-234.dsl.milwwi.ameritech.net)
21:44.25*** mode/#asterisk [+o anthm] by ChanServ
21:44.27Cresl1nbkw_: you just talk about things that you don't know adequately about
21:44.28mishehufreeswitch is completely different.
21:44.30VoIPMastapiwi: linphone
21:44.33[TK]D-FenderOMGZ, teh man!
21:44.34bkw_Cresl1n, not true
21:44.35saftsackCorydon-w, why did this fork died?
21:44.55VoIPMastapiwi: it's in the synaptic packet manager (if you add all the branches)
21:45.01saftsackor are there any news belong this diieng?
21:45.01Corydon-wsaftsack: because not enough people thought it was worthwhile and the reasons that they forked were addressed
21:45.02mishehubkw_: I thought they stopped trying to maintain it.
21:45.08*** join/#asterisk MRH2 (n=Mr_happy@host-83-146-30-242.bulldogdsl.com)
21:45.19VoIPMastajm|work: use a pastebin and paste your zapata.conf there
21:45.25VoIPMastajm|work: not here, in a pastebin
21:45.27[TK]D-Fenderbkw_ : Why fork * when you can just build FreeSwitch from the ground up right?
21:45.28jm|workok ok!
21:45.29jm|work:)
21:45.32saftsackok, thats not good because i thought openpbx was a good idea
21:45.49bkw_[TK]D-Fender, and actually build it correctly
21:45.56[TK]D-Fenderbkw_ : Hail Mary
21:45.58Dr-Linux[TK]D-Fender: ok, my other 3 servers have 1.2.9.1 but THIS one 1.2.0 i'll upgrade it on weekend
21:46.14bkw_I just have differing views on how things should be done vs what Digium has on the roadmap for Asterisk
21:46.15Corydon-wsaftsack: at its peak, it had maybe 5 developers, 4 of which have since left for greener pastures
21:46.19mishehu[TK]D-Fender: a house on a bad foundation will always sink into the ground.
21:46.22VoIPMastaI have some very old asterisk versions running on some servers
21:46.26jm|workVoIPMasta: http://pastebin.ca/91754
21:46.36VoIPMastabut since those servers are on a lan I'm not really concerned about security issues
21:46.52[TK]D-Fenderbkw_ : To me, as much as I love *, its greatest value to me is its REPLACABILITY.  Thats why I'm all for Polycom / Sangoma.  The "core" doesn't own me, and never will.
21:46.56mishehudoesn't matter if you try to move the house and keep the foundation.
21:46.58saftsackCorydon-w, thats not good :(
21:47.05TommyTheKidwow, sangoma has Solaris support.. hmmmmm :)
21:47.06jm|workVoIPMasta: and here is zaptel.conf http://pastebin.ca/91756
21:47.15Corydon-wsaftsack: and really, the problem with openpbx was that the prima donnas left the Asterisk project and tried to collaborate... and prima donnas can only collaborate for so long before they start to bicker amongst themselves
21:47.15Dr-Linuxanybody ever luck to configure Sphinx voice recognition system with asterisk?
21:47.16[TK]D-Fenderbkw_ : So when Freeswitch hits the threshold point we'll see where the dust settles.
21:47.17bkw_[TK]D-Fender, I can agree with that one
21:47.42bkw_[TK]D-Fender, I like options
21:47.52VoIPMastajm|work: try running ztcfg -vv
21:47.55MRH2TK  - speaking of polycom do you happen to know how far off SIP2.0 is from a release date?
21:48.03Corydon-wsaftsack: so basically, openpbx imploded when they couldn't agree on a path forward
21:48.11[TK]D-Fenderbkw_ : Yup, standards first, control second, source third...
21:48.14saftsackCorydon-w, hmm ok sounds logical
21:48.16jm|workVoIPMasta: http://pastebin.ca/91757
21:48.29[TK]D-FenderMRH2 : I haven't touched base with them lately.  I'm runing 2.0 beta right now.
21:48.37saftsackCorydon-w, but the reasons why openpbx was forked were logical in my opinion
21:48.45Corydon-wsaftsack: freeswitch is basically a project from the ground up by one of the former openpbx developers
21:48.48MRH2aye ok
21:48.54bkw_Corydon-w thats wrong
21:49.04bkw_Tony was never a developer on OpenPBX he just gave advice
21:49.08mishehuCorydon-w: fud
21:49.26*** join/#asterisk opus_ (i=opus@pabstblueribbon.net)
21:49.28[TK]D-Fendersaftsack : it wasn, but many of their goals will be acheived under FreeSwitch, so it became a moot point I would imagine.  So much more productive and fun building something nice than fixing up something broken.
21:49.38mishehujm|work: ewww!  no colonscopies in the channel please!
21:49.42jm|workewww
21:49.44jm|workstop it!
21:49.58jm|workI meant instead of a semi-col .... no, that's as bad
21:50.01*** join/#asterisk Scrye (n=ryan@2001:470:1f00:2514:280:c8ff:fec9:96d8)
21:50.02xachenhrm. I seem to sell lots of Openvox cards. Not many Digium though :(
21:50.09Scryeis there a url for supported phones?
21:50.12Dr-Linuxopus_: your caps lock :)
21:50.13VoIPMastajm|work: remove the "if you have one Wildcard..." text
21:50.18jm|workVoIPMasta: just doing that :)
21:50.37Corydon-wmishehu: so, you're still working hard on openpbx?  When's the milestone that was due 6 months ago going to be released?
21:51.07*** join/#asterisk h0 (n=h0@unaffiliated/fakhir)
21:51.17jm|workVoIPMasta: same error :(
21:51.36mishehuCorydon-w: you REALLY need to screw your head on a bit tighter.  you haven't gotten a single fact right yet.  I *never* was a developer for openpbx.  Neither was anthm.  And I *am* a developer for a module in freeswitch.
21:51.51piwiVoIPMasta: I installed linphone, changed its port but I cant login:'( SIP Identity is sip:username and SIP Proxy is sip:localhost isnt it?
21:52.02VoIPMastayup
21:52.12*** join/#asterisk SplasPood (n=jwb@206.252.198.101)
21:52.12mishehuand I'm doing it in C++ and maybe I'll dedicate it to kram, since he was rather insulting to me at last year's cluecon.
21:52.14VoIPMastapiwi: have you created a user in your sip.conf?
21:52.23piwiVoIPMasta: yes I did
21:52.35VoIPMastapiwi: what does asterisk say when you try to login?
21:52.39Corydon-wmishehu: contact me when freeswitch is finally ready in about 5 years.
21:52.39VoIPMastaor to "register"
21:52.54mishehuCorydon-w: that's about how long it took asterisk, no?
21:52.56Scryehas anyone used a UTStarCom F1000 with any success?
21:53.00piwiVoIPMasta: shall we go /msg?
21:53.05Cresl1nmishehu: I'm sure you NOTHING provocate it either
21:53.20VoIPMastasure
21:53.22mishehuCresl1n: I honestly did not understand what you said.
21:53.22Corydon-wmishehu: No, I learned about Asterisk after about 3.
21:53.38moghey now everybody lets settle down a bit
21:53.41mishehuCorydon-w: learned about, but didn't do anything for about 5.
21:53.44Cresl1nheh
21:54.02moggonna start kicking people, im looking at you Cresl1n ^_^
21:54.02Corydon-wmishehu: No, I started contributing to Asterisk after 3
21:54.17hadsYay. It's #basheveryone
21:54.18mishehuCorydon-w: so, does freeswitch step on your ego?  because you're quite critical about a project that is around 6 months old.
21:54.23MikeJ[Laptop]ding ding ding.. corners everyone..
21:54.39jm|workVoIPMasta: you want an account?
21:54.41moghey everybody i have a problem with my x100p, discuss
21:54.43jm|workwould that help?
21:54.44Strom_Cmy penis is bigger than your penis
21:54.49MikeJ[Laptop]mog, heh...
21:54.55hadslol mog
21:55.00Cresl1nStrom_C: oh no you didn't!
21:55.02Corydon-wNo, I simply find it interesting that the freeswitch developers hang around the Asterisk channel trying to poach users
21:55.11*** part/#asterisk h0 (n=h0@unaffiliated/fakhir)
21:55.16mogyes its all interesting, but back to my x100p
21:55.16mitchelocmog, i have an x100p, what's the problem?
21:55.27mishehumog: Hi, this is Eddie, your shipboard computer...
21:55.28MikeJ[Laptop]caller id problems?
21:55.29mogwell when i plug my phone into it
21:55.32VoIPMastahold on, I'm looking for one of my old zapata.conf files
21:55.35mogit doesnt give me dial tone
21:55.36Cresl1nit is quite funny how they can't seem to spend any of their time in their own chat rooms
21:55.47jm|workVoIPMasta: if you mean me: thanks.
21:55.53Strom_Ci have an x100p that a dog pissed on and then a truck ran over and now it doesnt seem to work !!!! help plzkthx
21:55.57Strom_Cmog mog mog
21:56.00mishehuCorydon-w: Just wait until we start launching missiles and declaring jihad on you.
21:56.02mitchelocmog, i think, you are supposed to plug your phone service into the X100P, i'm not sure, turn to your right and ask kp, he might know ;)
21:56.04anglermog, lol
21:56.10VoIPMastajm|work: yes, that was for you
21:56.13mishehuwe will wipe out those asterisk infidels.  ;-)
21:56.19mishehu</sarcasm>
21:56.21moghmm im not sure
21:56.28mogi think it should work
21:56.31Cresl1nmishehu: oooh.... I'm sure all the secret NSA IRC spies just loved hearing that from you
21:56.33mogi mean it worked on my last x100p
21:56.38MikeJ[Laptop]mog.. I think that's a #asterisk-dev question no...
21:56.53mogheh
21:56.53MikeJ[Laptop]wait.. is it a digium x100p?
21:56.57mogwell they sent me here
21:57.06mogits a clone ,,,,, err digium one
21:57.09mitchelocCrestln: you didn't know that #asterisk is a sleepr cell?
21:57.17hadsAsk jbot, he might know. Someone was trying to talk to him yesterday.
21:57.22mishehuCorydon-w: I am critical of asterisk because I use it and have experience with it.  If you have such a problem with freeswitch users using asterisk as well, and mentioning the f word on the channel, maybe you should see if you can ban us all.
21:57.24MikeJ[Laptop]yeah.. brand new from digium last month right?
21:57.27mitchelocCresl1n, supposedly we are all going to take over the telecom industry...
21:57.37Strom_Ci thought mine was an official digium clone of the clone of the official digium card
21:57.42mitcheloc* in the name of Allah
21:57.47Dr-Linuxmishehu: how you know this word "Jihad" ? :S
21:58.01MikeJ[Laptop]mog :P
21:58.08mogokay seriously
21:58.14Strom_CDr-Linux: I don't think there's a single person in north america that DOESN'T know what a jihad is :)
21:58.17mogeveryone drop it
21:58.33mogthere is another channel for everyone involved
21:58.34mog#drama
21:58.35MikeJ[Laptop]mog, I tried too.. no one listens to me either
21:58.38mogtake it there
21:58.43Corydon-wmishehu: and I'm critical of freeswitch because it's vaporware.  Free speech
21:58.53MikeJ[Laptop]mog.. join me?
21:58.53Dr-LinuxStrom_C: aww but does it mean in english? i thought it's my language word?
21:59.09jm|workVoIPMasta: here's some dmesg shizzy if it helps: http://pastebin.ca/91763
21:59.13Strom_CDr-Linux: it's a loanword
21:59.24mishehuCorydon-w: hahaha and opinions are like assholes, and all stink except for mine.
21:59.28mitchelocis a jihad a type of drink?
21:59.35moganyone who wants to discuss this further go to #drama
21:59.38mogplease
21:59.49TommyTheKidthanks for the sangoma reference guys! Solaris drivers, 8 port, 2U PCI cards.. I am in heaven :)
21:59.50mishehuCorydon-w: if anything, you only make us strive harder to make sure that freeswitch has a long, hardy life.
21:59.53MikeJ[Laptop]all the cool people are there!
21:59.59MRH2this channel has been flagged for analysis following a keyword occurence ;)
22:00.08MikeJ[Laptop]heh
22:00.08Corydon-wmog:  okay, I'm done now
22:00.14mogk
22:00.16mognext person
22:00.17mitchelocbut only MikeJ and mog are in #drama, meh
22:00.19mogis getting kicked
22:00.21mishehumog: do I get the part?  *grin*
22:00.28mishehuI've been working on my acting skills.
22:00.34TommyTheKidmmm lunch
22:00.36*** part/#asterisk TommyTheKid (n=tommythe@mpk-edge.cto.sunit.net)
22:00.43Dr-Linuxmitcheloc: Jihad is a kinda in the way of Allah
22:00.48andrejkwIs there any way of getting the amount of money in my account using Asterisk?
22:01.03bkw_FreeSwitch IS NOT vaporware
22:01.10Cresl1npsssh
22:01.14Cresl1njust like openpbx
22:01.15*** kick/#asterisk [bkw_!i=ejabberd@68.62.237.103] by mog (mog)
22:01.18Cresl1nand 16khz audio
22:01.22*** kick/#asterisk [Cresl1n!i=ejabberd@68.62.237.103] by mog (mog)
22:01.22*** join/#asterisk bkw_ (n=bkw_@asterisk/friend-and-developer/bkw)
22:01.25mogseriously
22:01.26mogstop it
22:01.28mogeveryone
22:01.33mogim not kidding
22:01.35mogim tired of it
22:01.36bkw_I just corrected the facts
22:01.37MikeJ[Laptop]yay..
22:01.39andrejkwCan someone help, please?
22:01.43MikeJ[Laptop]sure
22:01.46mishehumog: can I have one too?  I feel left out.  *grin*  j/k
22:01.51mogsure
22:01.52MikeJ[Laptop]what ya need andrejkw
22:01.58*** join/#asterisk Cresl1n (n=matt@gateway.digium.com)
22:01.58*** mode/#asterisk [+o Cresl1n] by ChanServ
22:01.58*** kick/#asterisk [mishehu!i=ejabberd@68.62.237.103] by mog (he asked for it)
22:01.59*** join/#asterisk mishehu (i=mishehu@cshells.shavedgoats.net)
22:02.01[TK]D-Fenderok, heading off to class, back in several hours
22:02.04*** mode/#asterisk [+b %bkw_!*@*] by Corydon-w
22:02.04znoGmishehu: i can provoke you if you like ;)
22:02.06mishehumog: I was just kidding.  ;-)
22:02.10mog^_^
22:02.13mogjust a joke
22:02.18mogbut seriously no more drama here
22:02.21andrejkwIs there any way to find out the amount of money in my SIP account using Asterisk?
22:02.22mogwe have a new channel for that
22:02.24mog#drama
22:02.25*** part/#asterisk bkw_ (n=bkw_@asterisk/friend-and-developer/bkw)
22:02.27mogfeel free to join
22:02.29mogall welcome
22:02.29Cresl1n#asterisk-drama
22:02.32Cresl1n:-P
22:02.32znoGandrejkw: eh!?
22:02.34*** mode/#asterisk [-b %bkw_!*@*] by Corydon-w
22:02.38*** join/#asterisk AvoidingDeadlock (n=bkw_@adsl-70-143-63-127.dsl.tul2ok.sbcglobal.net)
22:02.38mogno just #drama
22:02.40MikeJ[Laptop]no.. #drama
22:02.41andrejkwI am making an IVR.
22:02.41mogis generic
22:02.42Dr-Linuxmog: why you are kicking ???
22:02.53mogtoo much drama
22:02.55drrayyou could do like the list, and say that question belongs on the drama list
22:02.56mogeveryone was warned
22:02.59xachenop abuse imho :(
22:03.00mogseveral times
22:03.11VoIPMastajm|work: can you hold on for a couple minutes, let me boot up my laptop, I'm sure I have some conf files there
22:03.24jm|workVoIPMasta: of course! Thank you very much  :D
22:03.27andrejkwIs it possible to get the credit in the account?
22:03.32andrejkwSomehow?
22:03.38VoIPMastaandrejkw: what kind of account?
22:03.46andrejkwVoipBuster
22:03.46jm|workI should have said   sulks _patiently_  :)
22:03.51andrejkwSIP
22:04.17*** join/#asterisk foo (n=foo@unaffiliated/foo)
22:04.19VoIPMastammm you have to figure out a way to connect to voipbuster's database or to read data directly from their website
22:04.19Dr-Linuxmog: you was also involved in asterisk/freeswitch discussion .. and when you didn't like you warned and kicking ..
22:04.23fooWhat is the status on 911 calls with asterisks?
22:04.30fooThat is, knowing the location.
22:04.52Corydon-wfoo:  it's not something we can know
22:05.09MikeJ[Laptop]Cresl1n, where are you....
22:05.12MikeJ[Laptop]come play!
22:05.15MRH2anyone got issues with zaptel 1.2 since approx week commencing 3rd July?
22:05.17fileit's a trick
22:05.18filedon't do it
22:05.23TripleFFFFhmm good idea ill add this api to us
22:05.29MikeJ[Laptop]DO IT!
22:05.33fooCorydon-w: Hm, so there is no solution if someone calls 911 from a VoIP? That is, there is no location-specific notification 911 call centers get...
22:05.44TripleFFFFyeah
22:05.51Corydon-wfoo:  well, there is, but it's probably wrong
22:05.56fooahh, I see.
22:06.02TripleFFFFcosts a leg and an arm + an eye if you roll a 1 or a 2... ;)
22:06.12Corydon-wfoo:  you're need to set up something specific to handle that
22:06.31fooI see, but there is a way to handle that kind of situation?
22:06.36MRH2foo:  I route emergency calls over the pstn voip for everything else
22:06.44TripleFFFFMRH2
22:06.51TripleFFFFthen you will get the bills if one dial 911 to fuck around
22:06.56Corydon-wfoo:  there are various SIP providers who will complete the 911 call appropriately, but you must register the location of each phone in advance
22:06.59TripleFFFF50 to 150 per call..
22:07.00MikeJ[Laptop]Cresl1n, come back!
22:07.09fooCorydon-w: I see
22:07.23[andromeda]Is there a VoIP provider that will offer a free inbound PSTN phone number, that i will be able to use with asterisk?
22:07.24mishehufoo: where's your friend bar?
22:07.39foomishehu: We were separated at birth. If you find him let me know.
22:07.40MRH2<PROTECTED>
22:07.58*** join/#asterisk Qwell[] (i=north@unaffiliated/qwell)
22:07.58*** mode/#asterisk [+o Qwell[]] by ChanServ
22:07.59mishehufoo: heh, good one ;-)
22:08.22foo:)
22:08.30MRH2each company site has a line for it
22:08.34fooI see.
22:08.41fooHm, it would be awesome if I could set up asterisk at home.
22:08.42piwi[andromeda]: I dont think such a provider exists... if it does please let me know!!
22:09.26VoIPMastajm|work: /msg me
22:09.36VoIPMastajm|work: so that we don't flood the chnnl
22:09.39jm|work:)
22:11.36MRH2anyone using a 4xx card on a dell 2650 per chance?
22:12.33NDT2 x 410s on a 2850
22:13.12MRH2cool ever get that flashing orange pci error on the box?
22:13.15*** part/#asterisk Scrye (n=ryan@2001:470:1f00:2514:280:c8ff:fec9:96d8)
22:13.48NDTjust when I had to disable one of the E1000 onboard nics to get the second card to work...cause they are an issue...
22:14.18MRH2hmmm is that normally an irq type issue then?
22:14.20anthmI gotta go but before i do, corydon, say anything you like about freeswitch because it's your right, but do not pass me off as an ex-openpbx developer when you know for a fact that my name appears like 90 times in the asterisk code.  You can not undo all the work i put into making asterisk get to where it is now. I am entitled to use it, work on it still, and to sit in this channel all i wish.
22:14.21Dr-Linuxwhy asterisk doesn't have chan_sccp? :S
22:14.43Strom_CDr-Linux: /me points at qwell
22:14.54anthmI still recall you deeply insulting kpflemming when pissed you off then the next day you are a cheerleader cos they give you commit access that is not very cool and i would have expeceted better.
22:15.05NDTMRH2: Normally...
22:15.22MRH2reason being i get that orange pci error all the time with the recent zaptel 1.2
22:15.22Corydon-wanthm: eh?
22:15.36anthmi've been reading this flame war
22:15.45NDTwhats lspci say?
22:15.52anthmand i am not impresses with the ex-openpbx business
22:15.53MRH2had to downgrade to svn from 2nd July to get rid of it
22:16.17NDTWHen I had the issue the 410 would show up as unknown communication device
22:16.20anthmany i am late, bbl
22:16.25Corydon-wI told mog I'm done, so there's nothing more to say on channel
22:16.33*** join/#asterisk enjay- (n=enjay@71.216.165.97)
22:16.51MRH2yep me too
22:17.12Qwell[]Well that was...interesting
22:17.18Qwell[]What DID I miss?
22:17.25NDTGoing to have to play in the BIOS then with the IRQs heh
22:17.33fileQwell[]: lots
22:17.34mishehuQwell[]: nothing worth the read.
22:17.40filethe world exploded...
22:17.46Qwell[]file: ooo
22:17.47fileI became the overlord of the planet...
22:17.57Qwell[]and then?
22:18.05mishehufile: and then I deposed you and reinstated the previous world order
22:18.12MRH2yep i just couldn't seem to get rid of it with recent zaptel
22:18.13opus_hey i'm an "ex" openpbx developer! :)
22:18.14mishehujust by overwriting you
22:18.30MikeJ[Laptop]opus_, there a lot of those?
22:18.31Qwell[]jbot: logs
22:18.32jbotapt/ibot/jbot/purl all log to http://ibot.rikers.org/<channelname>/ where channelname is html encoded ie: %23debian | lines that start with a space are not shown | some channels have stats at http://ibot.rikers.org/stats/<channelname>.html.gz, or updated "nightly"
22:18.38opus_so osoma yo momma or whatever
22:18.42Qwell[]:p
22:18.48MikeJ[Laptop]hehe
22:18.56drraybarrack yo momma oboma?
22:19.40mishehudrray: heh
22:20.07Dr-Linuxwhy JerJer don't come to this channel?
22:20.15MRH2I don't suppose that is bug report worthy is it?
22:20.21Dr-Linuxeven every asterisk user loves him? :)
22:20.55MikeJ[Laptop]#drama !
22:21.02NDTMRH2: No...Dell says it's not their problem...Digium will say it isn't theirs, Sangoma will say you won't have the issue heh
22:21.10MRH2lol
22:21.56mogwhats the problem MRH2 ?
22:22.09*** join/#asterisk AJaymn (i=AJmn@70.59.126.197)
22:22.40MRH2looks like wierd irq conflict with a 4xx card and  zaptel1.2 > 3rd July
22:22.55Corydon-wMRH2: is the board still taking interrupts?
22:23.34*** join/#asterisk mog_home (n=mogorman@68.62.237.103)
22:23.34*** mode/#asterisk [+o mog_home] by ChanServ
22:23.54jm|workI need to back up some config files
22:24.21MRH2well it seemed to work just came up with a "dell orange flashing pci error" on the chasis
22:24.26*** join/#asterisk lullabud (n=lullabud@12.24.42.67)
22:24.38*** join/#asterisk |dennis| (i=dennis@200.32.215.84)
22:24.54MRH2since moved back to the 2nd July version and that works as it always has done
22:25.02*** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com)
22:25.14Corydon-wOn the motherboard or on the Digium card?
22:25.14lullabuddoes anybody know of a download link for the Polycom 3.1.2 bootrom?
22:25.25MRH2on the motherboard
22:25.39*** join/#asterisk Qwell[] (i=north@unaffiliated/qwell)
22:25.39*** mode/#asterisk [+o Qwell[]] by ChanServ
22:26.07Corydon-wDoes Dell have a diagnostic code for what that means?
22:26.44type0you talking about a poweredge?
22:26.55type0we had some blinking orange lights with our 4 poweredges
22:27.19MRH2i'll check with dell on that - money seems to be on an irq issue
22:27.25NDTIt's the E1000s and the Digium cards...they don't like to share interrupts
22:28.24Corydon-wMRH2: typically when we have an IRQ issue, we try either moving the card to a different slot, or failing that, we set noapic in the Linux kernel boot options
22:28.27*** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp)
22:28.43jm|workoh; it not existing might be it
22:28.50NDTlol
22:29.26Strom_Chaha
22:29.34VoIPMastajm|work: however don't get too excited with chan_h323 since it isn't THAT good
22:29.42MRH2yep tried the ususal stuff and couldn't get rid of it - just weird it happens only after a recent change to zaptel
22:29.47jm|workis it a m$ thing?
22:29.48VoIPMastaif JerJer reads this he'll start flaming me ;)
22:30.10VoIPMastajm|work: no, $ms has nothing to do with h323
22:30.12lullabudnevermind on the polycom boot rom.  amazing what a google search including "index of" will get you. ;-)
22:30.32jm|workVoIPMasta: no I just read that: amazingly they conformed to a RFC for once :O
22:30.48jm|workwell; an  h. at least
22:32.38Corydon-wMRH2: dell.com suggests that you're experiencing a low power condition.  Have you checked to make sure that you're getting sufficient voltage from both power supplies?
22:32.39MRH2I'll try disabling everything i can on the board and work up from there.
22:33.16NDTMRH2: Heh ain't many things to disable in that bios heh
22:33.34MRH2lol
22:33.36NDTSwicth the IRQ on the card in the bios to one the NICs aren't using
22:33.47MRH2yep did that
22:33.51jm|workso apart from pulver who else should I subscribe to?
22:34.02NDTWorked for me...until I added another card heh
22:34.06MRH2even tried moving the card
22:35.03Corydon-wMRH2: are there any events in the BIOS event log?
22:35.04drraymove it to another PC?
22:35.06*** join/#asterisk teknoprep (n=chris@unaffiliated/teknoprep)
22:35.09teknoprephi all
22:35.21NDTBottom slot worked perfectly for me...until I added that second card and had to disable one of the onboard NICS...needless to say...Anything I do with these poweredges we have again...will be with a sangoma card
22:35.45jm|workbah
22:35.56jm|workthis was not the best day to leave my mobile phone at work by mistake
22:36.02MRH2croydon i'll have to retest it with a later zaptel again - anything of note round about the 7th July that would mess with it?
22:36.14VoIPMastajm|work: do you want to test your "incoming" calls?
22:36.18jm|workfor some reason my land line is engaged every time I try to phone it ....
22:36.23jm|workVoIPMasta: perhaps :)
22:36.31NDTYou will see references to those NICS and 405/410/411 cards in a few places...
22:36.37jm|workit's 11:40pm here and the GF is bed and the extensions will probably ring
22:36.41VoIPMastajm|work: pm me your number and I can dial it (so that you can see if the call hits your asterisk box)
22:36.49VoIPMastaI see
22:36.53jm|workGFs, eh.
22:36.57jm|workWho'd 'ave 'em.
22:37.13jm|work"Still doing that boring phone stuff?  I'm going to bed."
22:37.18jm|work(:
22:37.47VoIPMastaStart worrying when she decides to go to bed with someone else, since you are more into voip than into her
22:37.50VoIPMastahehehe
22:37.54jm|workhehehe
22:38.30jm|workI'm pleased that I can dial out now :)
22:38.48MRH2ok gtg - thanx all - i'll report back ;)
22:38.52jm|workthat means that when I'm in New York on business (HAH!) I can call UK at national rates :)
22:38.57NDTlater MRH2
22:39.23jm|workI need more POPs!
22:39.29jm|workor PsOP
22:39.31jm|workor whatever
22:40.17Dr-Linuxi'm sorry friend, i passed you wrong comments about kicking.
22:41.36Dr-LinuxQwell[]: you come late today? :)
22:41.42Qwell[]Dr-Linux: yeah :p
22:41.53Qwell[]had to get around the firewall again...
22:42.44Dr-LinuxQwell[]: i like some opss here, they never talk, but when they talk they give some good appriated help
22:43.12Dr-LinuxQwell[]: but today i saw something different
22:43.24jm|work'mailbox' ... sheesh
22:43.30Qwell[]Dr-Linux: it's best to leave it alone...
22:43.50Strom_Cjm|work: i've never heard it called "answerphone" in the U.S.
22:43.59jm|work"voicemail"
22:44.08Strom_Cexactly :)
22:44.12jm|workforgive me for spawning you a language 1000 years ago ;)
22:44.33*** join/#asterisk bkw_ (n=bkw_@asterisk/friend-and-developer/bkw)
22:44.42Dr-LinuxQwell[]: actually everyone have respect in his/her own hand, some people dont understand ..
22:44.52Strom_CEnglsh in 1006 didn't exactly resemble modern English :)
22:45.01Dr-LinuxQwell[]: so when you are going to dedicate chan_sccp for asterisk? :P
22:45.11jm|workNay, Siree. I bequeath thee, such!
22:45.17Qwell[]Dr-Linux: Never.  I didn't write it
22:45.51Dr-LinuxStrom_C: i don't know english, but today i say .. i'm better :P
22:46.02jm|workI wish the analogue phone converters were as cheap as my dog-pissed-on-van-run-over-card :(
22:46.15Dr-LinuxQwell[]: why asterisk doesnt'have chan_sccp?
22:46.35Qwell[]Strom_C: ?
22:46.37Dr-LinuxQwell[]: Strom_C pointed me to you
22:46.42Corydon-wBecause skinny is the same thing
22:47.04Corydon-wand chan_skinny predates chan_sccp
22:47.06Dr-LinuxCorydon-w: Qwell[] don't think
22:47.07Qwell[]trying not to say unkind words about (the MIA, possibly dead) Sergio...
22:47.11Dr-Linuxdoesn't think the same
22:47.41Dr-LinuxQwell[]: i think Sergio has died :S
22:47.51Qwell[]I'm sure he has :p
22:48.04*** join/#asterisk hads (n=hads@mail.nice.net.nz)
22:48.12Dr-LinuxQwell[]: also i guess he had some problem with Sourceforg ...
22:48.15Qwell[]would be pretty hard to get a disclaimer from a dead man...
22:48.23Corydon-win which case the copyrights don't revert to the public domain for another 75 years
22:48.39Qwell[]Corydon-w: longer, if Disney has anything to say about it
22:48.56Corydon-wcorporations get 95 years, period
22:49.02Dr-LinuxQwell[]: sorry i don't understand difficult english :S
22:49.04Qwell[]fun
22:49.06Corydon-windividuals get life plus 75
22:49.11Dr-Linuxthe copyrights one
22:49.13Qwell[]life PLUS 75?!
22:49.23Corydon-wYep
22:49.26Qwell[]pfft
22:50.01Qwell[]so, like...
22:50.09Qwell[]maybe somebody should find out if he is still actually alive
22:50.25*** join/#asterisk Skarmeth (n=Skarmeth@201009018188.user.veloxzone.com.br)
22:50.35Corydon-wYou could also get his heirs to disclaim the copyright
22:50.38Dr-LinuxQwell[]: if redhat wants, can they take my domain from me?
22:50.45Qwell[]Dr-Linux: this is true
22:50.48TripleFFFFhey
22:50.48Qwell[]erm
22:50.52Qwell[]Corydon-w: this is true
22:50.57Qwell[]Dr-Linux: What domain?  Probably not
22:51.05Dr-LinuxQwell[]: redhat.pk
22:51.09Qwell[]heh
22:51.16Dr-Linuxtele.pk
22:51.20Qwell[]possible
22:51.22Dr-Linuxi don't think tele.pk
22:51.25Corydon-wDr-Linux: it depends.  How much money do you have to spend on a lawyer?
22:51.26jm|workVoIPMasta: is there an GUI softphone that doesn't have a bazzilion R-DEPS like gnome and shizzy?
22:51.29Dr-Linuxbut i'm asking about redhat.pk?
22:52.02Qwell[]Dr-Linux: That is likely a trademark issue (assuming they have a trademark there)
22:52.03Dr-LinuxCorydon-w: how can i take stand with red hat? huh i'm a poor pakistani
22:52.50Dr-LinuxQwell[]: i'm asking bcoz i just got this domain, if they will take it from me, then i'm not gonna work on it
22:52.51Corydon-wThey could try, but unless you're doing something that violates their trademark, they won't succeed unless they can successfully make you spend enough money that you just give up
22:52.56Dr-Linuxthen i'll prefer tele.pk
22:53.06*** part/#asterisk hohum (n=dcorbe@12.195.58.235)
22:53.19Corydon-wAnd that's for any corporation, not just RedHat
22:54.20Dr-LinuxCorydon-w: so will they give me money that i'd have spent on site, or i'll be just removed?
22:55.57*** join/#asterisk dlynes_laptop (n=dlynes@S0106001217014b92.vc.shawcable.net)
22:56.19Dr-Linuxbad bad :( i have about 15 trade mark domains
22:56.46[andromeda]Has anyone here used Gizmo's inbound service with asterisk?
22:57.18Corydon-wDr-Linux: generally not, but talk to a lawyer to be sure
22:58.35Dr-LinuxCorydon-w: we don't have tht system here, remember i'm a tribal guy .. no police, no courts, nor govt. etc rulse
22:58.41Dr-Linuxrules
22:59.54opus_yeah you should be fine in pakistan
23:00.03Corydon-wDoes the tribal chief like you?
23:00.12Qwell[]more importantly...
23:00.16Qwell[]Does the tribal chef like you?
23:00.24opus_why are so many VOIP people from pakistan :)
23:00.30Dr-LinuxCorydon-w: sorry i didn't understand friend?
23:00.36dlynes_laptopopus_, call centers
23:00.38drraycheap
23:00.39drrayer
23:00.40dlynes_laptopopus_, same for india
23:01.02Dr-Linuxopus_: i think i'm only guy in tribals who knwos a bit Linux/voip etc
23:01.08Corydon-wDr-Linux: if the system of justice is in your favor, then you have little to worry about
23:01.43Dr-LinuxCorydon-w: but domains case is international
23:02.10Dr-Linuxif they take my domain, i'll do again bad bad stuff as before
23:02.11opus_red hat might not be registered to do business in your country :) so it might be fair game
23:02.26Corydon-wDr-Linux: they still have to contact you.  If the lawyer is strung up in the town square before he serves you with papers, there's not much they can do
23:02.29opus_just like going to mcdonalds in Iran I think? :) Or Star burger in mexico
23:02.45Dr-Linuxtribal people only know guns etc stuff
23:02.45*** join/#asterisk gaupe (i=rmo@slogen.sunnmore.net)
23:02.59opus_Dr-Linux is bin laden really dead? :)
23:03.09dlynes_laptopopus_, didn't you know?
23:03.15dlynes_laptopopus_, bin laden is Dr-Linux's cousin
23:03.24Dr-Linuxopus_: if they can have after somethime ..
23:03.38Dr-Linuxopus_: who says ben laden dead??
23:03.44opus_the president of pakistan said he was dead
23:03.50Dr-Linuxi'm from the same town
23:03.52NDTHas anyone looked at the UNIQUEID of a call after it has returned from a macro?
23:03.56dlynes_laptopSee????
23:03.58drrayI believe he's been dead for a year or two
23:03.59Dr-Linuxopus_: lolzz
23:04.03dlynes_laptopHe's Bin Laden's cousin!!!!!
23:04.06drraybut I could be wrong
23:04.08Dr-Linuxhe is just a shit liar
23:04.15opus_bin laden runs asterisk, i knew it.
23:04.31opus_Dr-Linux the president?
23:04.44Dr-Linuxhe says, bcoz he don't want US army to get in tribals area of pakistan to search him
23:04.47dlynes_laptopDr-Linux, he just wants the americans out of pakistan and afghanistan, so he can get on with his life :)
23:04.49opus_Dr-Linux how tall is bin laden?
23:05.03Dr-Linuxopus_: and US army make him an issue to try to get in tribals
23:05.23opus_huh?
23:05.39*** part/#asterisk foo (n=foo@unaffiliated/foo)
23:05.40Dr-Linuxthat hows they bombing my area on my Eid day :( my baby cousins were died :(
23:05.52Qwell[]Eid day?
23:06.02dlynes_laptopmuslim holiday near christmas time
23:06.07Dr-Linuxdlynes_laptop: there is no American army in tribal dude.
23:06.10Dr-Linuxand can't be
23:06.11dlynes_laptopa time of celebrating by eating lots after a time of fasting
23:06.24dlynes_laptopeids
23:06.25Dr-LinuxTribals are fighting with Pakistan army
23:06.54dlynes_laptopright, Dr-Linux ?
23:07.11Dr-Linuxopus_: actually Laden was not that much as US mentiond
23:07.21Dr-Linuxdlynes_laptop: Eid is our holiday
23:07.29dlynes_laptopthat's what i said
23:07.35dlynes_laptopIt's like the muslim version of Christmas
23:07.45dlynes_laptopA day to pig out on lots of food
23:07.47Dr-Linuxdlynes_laptop: imgine .. how US planes without Poilot bombing my area ..
23:07.49dlynes_laptopand exchange presents
23:07.52Dr-Linuxyes yes
23:08.00Dr-Linuxpig? aghhh
23:08.44Corydon-wDr-Linux: it's a euphemism, meaning "to eat like Americans"
23:09.02Dr-Linuxwe don't eat pork
23:09.11dlynes_laptopnobody said anything about eating pork
23:09.14Corydon-wDr-Linux: as in, a large quantity
23:09.47Dr-Linuxsorry, dlynes_laptop some time i don't understand english but always try though
23:09.53Qwell[]"pig out" == "gorge"
23:09.55Corydon-wdlynes_laptop: you're talking to someone without much knowledge of euphemisms
23:10.24dlynes_laptopDr-Linux, nod...thought you were jsut reading it wrong
23:10.49Corydon-wHence, the English => Russian => English translation into "The vodka is good, but the meat is rotten." for "The spirit is willing, but the flesh is weak."
23:11.04NDTQwell: You have any idea how I can work around the uniqueid of a call changing when returning from a macro? I need the original uniqueid for use again...but when you return from the macro it changes it
23:11.36Dr-LinuxCorydon-w: it's not about Knowledge, it's bout culture and langauge.
23:11.48*** join/#asterisk heliosj (n=jeff@pdpc/supporter/active/xheliox)
23:11.56Dr-Linuxi'm sure you'd not know even a word of my langauge except Jihad
23:11.59dlynes_laptopDr-Linux, yes, americans love to eat :)
23:12.13*** part/#asterisk ariel_ (n=ariel_@74.8.35.2)
23:12.32dlynes_laptopDr-Linux, jihad's an arabic word, not urdu, isn't it?
23:12.37Dr-Linuxdlynes_laptop: that's how i can't go to USA/UK :)
23:13.18Dr-Linuxdlynes_laptop: it's the Same, it's a Holly word
23:13.53hadsdlynes_laptop: Got an encouraging email back from Sangoma today about organising telepermits for NZ. At this stage they are interested in getting it done.
23:14.08dlynes_laptopcool
23:14.37Dr-Linuxdlynes_laptop: what you say, if i do somethign wrong like cybercrime, what FBI can do with me?
23:14.53Corydon-wDr-Linux: you don't really want to know
23:15.01Qwell[]federal PMITA prison
23:15.12dlynes_laptopthey'll sick the cia on you and supply you with kalishnikovs
23:15.14Corydon-wIn any case, it's not the FBI that would do it to you
23:15.37Dr-Linuxwho will do then?
23:15.47Corydon-wDr-Linux: even in the US, people disappear without a trace
23:16.09*** join/#asterisk anthm (n=anthm@000-439-099.area4.spcsdns.net)
23:16.09*** mode/#asterisk [+o anthm] by ChanServ
23:16.21Dr-LinuxCorydon-w: correct
23:16.39Corydon-wDr-Linux: I suspect the same is true of most countries
23:17.02Dr-LinuxCorydon-w: but i dont think they can do anything in tribals
23:17.33Corydon-wDr-Linux: so nobody you've ever known has ever disappeared?
23:17.55kcortez]
23:18.03Dr-LinuxCorydon-w: here? no
23:18.14AndyCaphads: they'll get to you when they're done with Australia. :P
23:18.29hadsAndyCap: Heh :)
23:18.35Dr-LinuxCorydon-w: i know a number of peoples doing carding/hacking etc and nothing happend
23:18.35Corydon-wDr-Linux: I'm surprised, actually.  I've known a few people who have vanished
23:19.17gaupeit happened a lot here in Norway too, but that was mostly between 1940 and 1945 ;)
23:19.25Corydon-wDr-Linux: they may have disappeared of their own volition, or they may have been disappeared by someone else, and we'll probably never know the truth
23:19.48Dr-LinuxCorydon-w: well, in the tribal there is some different case
23:19.57Dr-LinuxCorydon-w: do you know about tribals?
23:20.05Corydon-wgaupe: these are all in the last 10 years, for people who, as far as I'm aware, have never left the USA
23:20.26Dr-LinuxCorydon-w: even Pakistan police can't enter in the tribals area
23:20.27*** join/#asterisk kfudge (n=kzymxa@24-217-137-21.dhcp.stls.mo.charter.com)
23:20.30AndyCapCorydon-w: at least not before they disappeared.. ;)
23:20.47kfudgehow would I make an extension say 99 go to an ivr
23:20.54AndyCapextraordinary rendition and all that
23:21.00Corydon-wAndyCap: well, they could be anywhere.  That's part of the point.
23:21.17gaupeDr-Linux: we've got pakistani tribes in oslo too
23:21.45Corydon-wAndyCap: the two people I'm thinking of in particular are both white adults, one Christian, one Jewish.
23:22.07Dr-Linuxgaupe: oslo?
23:22.13dlynes_laptopDr-Linux, norway
23:22.26Dr-Linuxawww
23:22.54Dr-Linuxeverything is broken here related to norway/danmark
23:23.06gaupeDr-Linux: gangs made of pakistanis battling for the control of the illegal business
23:23.09Strom_Cgood things from
23:23.25Strom_Cer
23:23.30Strom_Cgood things from Norway: a-ha :)
23:23.35Dr-Linuxgaupe: we are really very unlucky, you know why?
23:23.47gaupeDr-Linux: no?
23:23.59Dr-Linuxgaupe: my few of family in Pakistan army,
23:24.11Dr-Linuxgaupe: and my rest of family is tribals
23:24.47Dr-Linuxand currently pakistan army and tribals are fighting bad, killing each other, tribals have power..
23:25.00*** join/#asterisk lyroy (n=lyroy@modemcable146.87-83-70.mc.videotron.ca)
23:25.20Dr-Linuxthen imagine .. an army guy is attacking at his own home
23:25.32lyroyDoes someone here ever configure a Cisco ATA 186 v.3.1.. SIP with Asterisk?
23:25.32Dr-Linuxgaupe: why? do you know
23:27.38*** join/#asterisk zotz (n=zotz@24.244.133.115)
23:32.02*** join/#asterisk jbot_ (i=ibot@pdpc/supporter/active/TimRiker/bot/apt)
23:32.02*** topic/#asterisk is Asterisk: The Open Source PBX -=- http://www.asterisk.org -=- Asterisk 1.2.10 and Zaptel 1.2.7 released! (July 14, 2006) -=- FreePBX/AMP/Asterisk@Home/Trixbox support in #freepbx
23:32.14Dr-Linuxdlynes_laptop: you are good in work, or good in making deals with customers? :)
23:32.27*** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com)
23:32.31dlynes_laptopI'm good at the work; the owner is good at finding customers
23:33.01Dr-Linuxgood match
23:33.19Dr-Linuxdlynes_laptop: how many peoples are working at your work place? :)
23:33.21dlynes_laptopwell, the better match is that he's a traditional telephone guy, and i'm a computer guy
23:33.27dlynes_laptopso it covers all the bases
23:33.39dlynes_laptopright now, two of us for the most part
23:33.42riddleboxwhat could cause my asterisk to not give me ringback some of the time(mostly to pstn) cell phones work fine?
23:33.48dlynes_laptophowever, we do subcontract once in a while
23:34.12dlynes_laptopriddlebox, mind rephrasing that?  it didn't make any sense
23:34.14Dr-Linuxdlynes_laptop: what you guys offer?
23:34.25dlynes_laptopDr-Linux, everything but the kitchen sink
23:34.32Dr-Linux:S
23:35.27riddleboxdlynes_laptop, when I call someone lets say a home phone, I have no ringback, but if I call a cellphone I get ringback
23:35.53dlynes_laptopby ring back, you mean a ringing sound while you're waiting for the other end to answer?
23:36.00*** part/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net)
23:36.36dlynes_laptopriddlebox, ?
23:36.40riddleboxdylnes_laptop,yes
23:36.52dlynes_laptopriddlebox, is the call to the cellphone going through a gsm gateway?
23:37.01riddleboxdylnes_laptop,yes
23:37.14dlynes_laptopriddlebox, and the call to the phone is going through BRI ISDN?
23:37.18Dr-Linuxdlynes_laptop: he means,  >> dialing number >> dialed >> now listening and waiting >>> toooooon tooooooon .... tooooooooon ... toooooon
23:37.19Dr-Linux:)
23:37.23lyroyDoes someone here is using a Cisco ATA 186 if so what version are u using (SIP)?
23:37.38wunderkintooon?
23:37.44wunderkinsounds different there
23:37.56riddleboxdlynes_laptop, it is through broadvoice
23:37.57dlynes_laptopwunderkin, the ringing in india sounds super weird, too
23:38.13Dr-Linuxwunderkin: what's there sound?
23:38.16dlynes_laptopriddlebox, maybe they're not passing progression
23:38.22dlynes_laptopriddlebox, try complaining to them
23:38.36dlynes_laptopriddlebox, i.e. call progression
23:44.24*** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt)
23:44.24*** topic/#asterisk is Asterisk: The Open Source PBX -=- http://www.asterisk.org -=- Asterisk 1.2.10 and Zaptel 1.2.7 released! (July 14, 2006) -=- FreePBX/AMP/Asterisk@Home/Trixbox support in #freepbx
23:48.29*** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net)
23:54.53*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)
23:55.32*** join/#asterisk carl0s- (n=carl0s@compsup.demon.co.uk)
23:55.47carl0s-msg nickserv identify abracadabra
23:56.10dlynes_laptopcarl0s-, thanks for telling us what your nickserv password is
23:56.11carl0s-:p
23:56.18Dr-Linuxcarl0s-: lolz
23:56.29carl0s-:D
23:56.36Dr-Linuxcarl0s-: do like this:
23:56.38carl0s-I've been wanting to do that for a while :D
23:56.47Dr-Linux/ns identify abracadabra
23:57.21dlynes_laptopDr-Linux, /ns boot dr-linux
23:57.29carl0s-Dr-Linux: what does that do?
23:57.52carl0s-It wasn't really my nickserv password. I just pretended for a laugh.
23:57.58Dr-Linuxcarl0s-: same that you were trying to doing, but in short way
23:58.01carl0s-(yeah like people are going to beleive me)
23:58.07carl0s-Dr-Linux: ah, thanks :)
23:58.21carl0s-I think I probably wouldn't have been able to join the channel if I hadn't already identified anyway
23:58.32Dr-Linuxcarl0s-: like what it does:
23:58.39Dr-Linux//say $OS
23:58.57carl0s-yep, I've seen that stuff before. a long time ago.
23:59.08dlynes_laptop$OS
23:59.10dlynes_laptopand?
23:59.30Dr-Linuxdlynes_laptop: you are not on windows :)
23:59.38dlynes_laptopyes, i am
23:59.42dlynes_laptopwindows xp
23:59.49carl0s-hmm I thought it'd report the O/S. I remember people trying to get newbies to do things like "/exec $decrypt[a83hs83hfd893h93]" where it decrypted to "rm -rf /"
23:59.54carl0s-XP

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