| 00:00.03 | hhkahya | yes, installed and configured step by step in the inst. but it is not working |
| 00:00.10 | darkdrgn2k | hhkahya: freepbx? |
| 00:00.16 | hhkahya | probably centos problem (or apache ) |
| 00:00.18 | ManxPower-work | hhkahya: Then you should ask on the #AsteriskNow or AsteriskGUI |
| 00:00.19 | equijada | i have the same problem with extension 8 but 8 it is not in my extensison |
| 00:00.26 | hhkahya | darkdrgn2k : asterisk 1.6 |
| 00:00.34 | [TK]D-Fender | [18:57]<hhkahya>i have been created and included it to sip.conf sip forwarding settings to |
| 00:00.40 | [TK]D-Fender | hhkahya: there is no such thing as forwarding |
| 00:00.46 | ManxPower-work | Because you'll never understand what we are saying and what we tell you won't apply to a GUI configuration |
| 00:01.00 | equijada | http://pastebin.com/m71417bf6 |
| 00:01.04 | equijada | this is my error |
| 00:01.07 | [TK]D-Fender | hhkahya: sip.conf points to a dialplan conectext. Go look where your CALL is looking for that match and SHOW US your dilaplan and the SIP DEBUG of your failed call |
| 00:01.09 | [TK]D-Fender | ~pb |
| 00:01.09 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
| 00:01.11 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^^ |
| 00:01.52 | equijada | Unknown extension '8' in context 'from-ptsn' requested |
| 00:02.06 | equijada | but i dont have any 8 in my extension |
| 00:02.07 | [TK]D-Fender | equijada: Trying to dial from a Zaptel FXS channel? |
| 00:02.38 | *** part/#asterisk ManxPower-work (n=EWieling@216.186.151.147) |
| 00:02.41 | equijada | [TK]D-Fender:No I have a T1 channel and I calling to this |
| 00:03.02 | [TK]D-Fender | equijada: What is it conencted to? |
| 00:03.14 | [TK]D-Fender | equijada: What kind of T1 signalling? |
| 00:03.23 | equijada | AMI |
| 00:03.27 | equijada | superframe |
| 00:03.38 | equijada | i show the zaptel too |
| 00:03.50 | equijada | in this file |
| 00:04.13 | *** join/#asterisk slinksh0t (n=slinksh0@c-76-108-31-194.hsd1.fl.comcast.net) |
| 00:04.19 | equijada | just i want is receive a call for this trunk and say a file |
| 00:04.24 | [TK]D-Fender | equijada: what DEVICE is * conencted to for this? |
| 00:04.36 | equijada | openvox d110p |
| 00:04.56 | [TK]D-Fender | equijada: THE OTHER SIDE DAMMIT |
| 00:06.04 | equijada | [TK]D-Fender: telco t1 |
| 00:06.10 | equijada | i mean |
| 00:06.40 | [TK]D-Fender | equijada: Looks like thy may be trying to send you a DID via FSK |
| 00:06.54 | [TK]D-Fender | equijada: Do you have DID's starting with 8? |
| 00:06.58 | equijada | nop |
| 00:07.08 | equijada | well, the pilot number |
| 00:07.13 | equijada | begins with 8 |
| 00:07.28 | equijada | the main number is 809.... |
| 00:08.23 | [TK]D-Fender | equijada: Then Go make an IVR to read in the full number. exten => _X.,NoOp(Telco is sending "${EXTEN}") |
| 00:08.29 | [TK]D-Fender | equijada: Then Go make an IVR to read in the full number. exten => _X.,1,NoOp(Telco is sending "${EXTEN}") |
| 00:08.32 | [TK]D-Fender | (forgot prio |
| 00:09.28 | *** join/#asterisk slinksh0t (n=slinksh0@adsl-233-203-182.mia.bellsouth.net) |
| 00:09.34 | equijada | i dont uunderstand :( |
| 00:11.28 | hhkahya | my main set-ups http://pastebin.com/m63890fa8 |
| 00:15.53 | *** part/#asterisk snadge (n=snadge@starbug.ugh.net.au) |
| 00:16.25 | [TK]D-Fender | equijada: I just gave you an exten to add to your inbound context so it doesn't crap out on the fact you're GETTING a number and can't see what it is in full |
| 00:16.49 | [TK]D-Fender | hhkahya: host=212.24.146.39&212.24.146.38&82.113.42.140<- can't have multiple hosts |
| 00:17.02 | [TK]D-Fender | hhkahya: And that pastebin doesn't show me a FAILURE |
| 00:17.16 | [TK]D-Fender | [18:56]<hhkahya>i have take a like this error [Jan 11 01:56:09] NOTICE[31986]: chan_sip.c:18002 handle_request_invite: Call from â to extension '226254282' rejected because extension not found. |
| 00:17.28 | [TK]D-Fender | hhkahya: I fail to see an EXTENSION to match that number |
| 00:18.28 | [TK]D-Fender | hhkahya: You have 1 exten in the context we only HOPE that your call is even LOOKING AT. It is "s". "s" is not a magic catch-all, and your call is failing because you don't have an exten that can match that number it is showing you |
| 00:18.59 | dlynes | [TK]D-Fender, testy testy... |
| 00:19.01 | dlynes | blinks. |
| 00:19.25 | hhkahya | i see, thanks, i am trying on it :) |
| 00:20.37 | hhkahya | <[TK]D-Fender> how can i enter the multiple hosts ? |
| 00:20.50 | [TK]D-Fender | [19:16]<[TK]D-Fender>hhkahya: host=212.24.146.39&212.24.146.38&82.113.42.140<- can't have multiple hosts |
| 00:21.01 | [TK]D-Fender | hhkahya: What part of NOT POSSIBLE are you not understanding? |
| 00:22.00 | hhkahya | i understand but should i enter different parts for that ips ? how can reach these hosts ? |
| 00:23.36 | [TK]D-Fender | hhkahya: You cannot have multiple specific hosts to CONTACT. It is not possible. NOT FUCKING POSSILBE. Am I clear now? |
| 00:23.57 | [TK]D-Fender | hhkahya: There is no "how" in doing this with *. |
| 00:24.25 | hhkahya | thanks |
| 00:28.47 | equijada | [TK]D-Fender |
| 00:28.49 | equijada | : |
| 00:28.56 | equijada | I get this |
| 00:28.58 | equijada | Executing [8092385690@from-ptsn:1] NoOp("Zap/1-1", ""8092385690"") in new stack |
| 00:29.23 | equijada | but that is the number that i call |
| 00:29.48 | [TK]D-Fender | equijada: Good. Then make an exten to MATCH it. |
| 00:30.22 | equijada | but the idea of "s" it is not that |
| 00:30.23 | equijada | ? |
| 00:30.57 | equijada | my extension just have 3 lines |
| 00:31.21 | voipmonk | grins |
| 00:31.34 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 00:31.42 | [TK]D-Fender | equijada: Another person who fails to READ |
| 00:31.54 | [TK]D-Fender | [19:18]<[TK]D-Fender>hhkahya: You have 1 exten in the context we only HOPE that your call is even LOOKING AT. It is "s". "s" is not a magic catch-all, and your call is failing because you don't have an exten that can match that number it is showing you |
| 00:32.14 | [TK]D-Fender | equijada: Because of the nature of your T1, there IS a DID dialed. Therefor * HAS a number to look for and the call does NOT go to "s" |
| 00:33.04 | equijada | exten => s,1,Answer() |
| 00:33.04 | equijada | exten => s,2,Playback(vm-Work) |
| 00:33.04 | equijada | exten => s,3,Hangup() |
| 00:33.43 | equijada | so we need to create an extension with this number |
| 00:34.06 | voipmonk | yes |
| 00:34.08 | voipmonk | you do equijada |
| 00:34.23 | voipmonk | points to [TK]D-Fenders post |
| 00:34.32 | equijada | ok |
| 00:34.37 | voipmonk | gets one of those big game fingers and points |
| 00:34.41 | equijada | let 's do it so |
| 00:34.59 | voipmonk | no lets, just do it |
| 00:35.08 | voipmonk | then retest |
| 00:35.08 | voipmonk | good luck |
| 00:35.09 | voipmonk | :) |
| 00:35.14 | p3nguin | like Nike |
| 00:35.43 | *** join/#asterisk Defraz (n=Defraz@c72co-edge-router.fuzecore.com) |
| 00:35.54 | equijada | so replace s by the number? |
| 00:36.17 | p3nguin | yes |
| 00:42.40 | equijada | Yeah!! Works!! |
| 00:42.56 | equijada | Thks!! 2 day without sleep for this |
| 00:43.09 | equijada | thks [TK]D-Fender!!!! |
| 00:44.21 | equijada | but I wanna know why? I have relaized anothers instalations and I never get this |
| 00:44.39 | [TK]D-Fender | equijada: This telco passes the number. End of story |
| 00:45.44 | equijada | [TK]D-Fender: so this number is the main number for my T1 |
| 00:46.28 | equijada | telco can I send another number for another channel |
| 00:47.31 | [TK]D-Fender | equijada: Looks lik |
| 00:48.14 | equijada | thks!! |
| 01:00.10 | rossand | This is odd: twinkle (on Linux) works out/in with SIP. Same account with my snom phone can call out but gets an error for incoming calls. Here's a partial quote: "Received response: "Forbidden" from ... is circuit-busy" Any suggestions? |
| 01:00.15 | *** join/#asterisk xuser (n=xuser@unaffiliated/xuser) |
| 01:02.06 | rossand | In the messages log, I see this: Forbidden - maybe wrong password on authentication for NOTIFY |
| 01:02.41 | [TK]D-Fender | rossand: Notify != call. And your partial info isn't useful. |
| 01:02.50 | [TK]D-Fender | rossand: Go look at your configs and complete call debug |
| 01:03.58 | rossand | [TK]D-Fender: That's what brought me here. That is all the information. unfortunately. |
| 01:04.21 | [TK]D-Fender | rossand: then go get more. |
| 01:05.16 | corretico | hello eveboyd |
| 01:05.29 | corretico | sorry people |
| 01:05.54 | corretico | I need some help with my fresh asterisk installation |
| 01:06.36 | corretico | I'm new with asterisk and I want to use my asterisk with a Cisco 7960 SIP Phone |
| 01:06.45 | corretico | Any suggest for that?? |
| 01:07.22 | *** join/#asterisk Kumbang (n=kumbang@rusnas.paume.itb.ac.id) |
| 01:07.51 | [TK]D-Fender | corretico: Go configure it |
| 01:08.17 | corretico | <[TK]D-Fender>sure... but I dont know how I can start on it |
| 01:08.19 | corretico | jejeje |
| 01:08.38 | corretico | sorry for my english |
| 01:09.08 | [TK]D-Fender | corretico: Go read the WIKI. Plenty of guides there. |
| 01:09.10 | [TK]D-Fender | ~wikis |
| 01:09.10 | infobot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 01:12.18 | rossand | [TK]D-Fender: Thanks anyhow - only 2 lines of output in the log & high verbose is all I've got. Since twinkle works for in/out, and snom out but not in, it gives me a thread to pull on and try experimenting to figure it out. |
| 01:12.49 | [TK]D-Fender | rossand: Logs are worthless. Go pay attention to * CLI & SIP DEBUG |
| 01:13.09 | rossand | [TK]D-Fender: Cool, thank you. That's helpful. |
| 01:14.39 | rossand | [TK]D-Fender: sip debug is the output from running asterisk with -v's or core set verbose #, correct? |
| 01:15.05 | [TK]D-Fender | rossand: No, sip debug is entirely different and has to be requested at CLI |
| 01:15.10 | [TK]D-Fender | rossand: "help sip" |
| 01:15.33 | rossand | I see it... sip set debug. Thanks again. |
| 01:29.22 | VxJasonxV | Anyone have any documentation on res_phoneprov? Google is turning up mostly checkin pages, mailing lists, ml resyndicates... etc. |
| 01:30.35 | *** join/#asterisk Badrobot- (n=badrobot@cpe-76-173-229-89.socal.res.rr.com) |
| 01:31.24 | carrar | corretico, also skim through the "Book" |
| 01:31.28 | carrar | ~book |
| 01:31.29 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 02:01.25 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 02:03.37 | [TK]D-Fender | VxJasonxV: Poorly documented, and probably only marginally useful. Go configure them yourself |
| 02:07.55 | *** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com) |
| 02:17.55 | *** join/#asterisk xuser (n=xuser@unaffiliated/xuser) |
| 02:28.53 | *** join/#asterisk pawz (n=pawz@210.56.81.88) |
| 02:33.59 | *** join/#asterisk chuckf (n=chuckf@ubuntu/member/chuckf) |
| 02:34.03 | *** join/#asterisk plut0 (n=cory@cpe-72-224-58-127.nycap.res.rr.com) |
| 02:34.32 | plut0 | having nat issues with SIP, i setup stun, can make calls but theres no voice |
| 02:36.26 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 02:36.53 | *** join/#asterisk Tech_Travis (n=Administ@cpe-76-168-191-127.socal.res.rr.com) |
| 02:43.45 | VxJasonxV | [TK]D-Fender, good answer :P |
| 02:44.47 | VxJasonxV | hmm. a call exiting "non-zero" is bad, isn't it? |
| 02:45.00 | VxJasonxV | Is there something I'm supposed to do after issuing a Hangup() in the dialplan? |
| 02:45.11 | ChannelZ | have a cookie |
| 02:45.37 | VxJasonxV | perhaps not "bad", but, wouldn't I rather calls exit zero? |
| 02:45.54 | *** join/#asterisk fofware (n=chatzill@190.7.25.160) |
| 02:48.04 | [TK]D-Fender | VxJasonxV: No |
| 02:48.16 | VxJasonxV | oh. well then |
| 02:48.23 | ChannelZ | It usually just means someone hung up |
| 02:49.10 | VxJasonxV | I got 30 SIP calls from some stranger, and when I looked back at their IP, it's a spammer. |
| 02:49.15 | VxJasonxV | SIP spam? seriously? |
| 02:49.29 | ChannelZ | made ya look! |
| 02:49.48 | ChannelZ | they'll do anything |
| 02:49.54 | VxJasonxV | indeed |
| 02:52.35 | *** join/#asterisk hluesea (n=hulusika@88.239.44.185) |
| 02:52.39 | hluesea | thanks |
| 02:52.52 | *** join/#asterisk ruben23 (n=AGENT@122.55.48.243) |
| 02:59.45 | *** join/#asterisk neurosys (n=neurosys@c-71-196-20-208.hsd1.fl.comcast.net) |
| 03:01.55 | *** part/#asterisk plut0 (n=cory@cpe-72-224-58-127.nycap.res.rr.com) |
| 03:10.08 | *** join/#asterisk angler (n=angler@pdpc/sponsor/digium/angler) [NETSPLIT VICTIM] |
| 03:10.08 | *** join/#asterisk hluesea (n=hulusika@88.239.44.185) [NETSPLIT VICTIM] |
| 03:10.08 | *** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com) [NETSPLIT VICTIM] |
| 03:10.08 | *** join/#asterisk binbash_ (n=peter@ip4da53781.direct-adsl.nl) [NETSPLIT VICTIM] |
| 03:10.08 | *** join/#asterisk toppe (n=root@cs181001225.pp.htv.fi) [NETSPLIT VICTIM] |
| 03:10.09 | *** join/#asterisk dandre (n=daniel@ble59-2-81-56-122-47.fbx.proxad.net) [NETSPLIT VICTIM] |
| 03:10.09 | *** join/#asterisk rossand (n=aross@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com) [NETSPLIT VICTIM] |
| 03:10.09 | *** join/#asterisk retentiveboy (n=pdugas@atl.pra-corp.com) [NETSPLIT VICTIM] |
| 03:10.09 | *** join/#asterisk hardwire (n=spencers@39.183.dowl.anc.borealisbroadband.net) [NETSPLIT VICTIM] |
| 03:10.09 | *** join/#asterisk ticoit (n=ticoit@201.191.151.155) [NETSPLIT VICTIM] |
| 03:10.09 | *** join/#asterisk dustybin (i=dustybin@thinkdebian.org) [NETSPLIT VICTIM] |
| 03:10.09 | *** join/#asterisk niekie (i=quasselc@CAcert/Assurer/niekie) [NETSPLIT VICTIM] |
| 03:10.09 | *** join/#asterisk edguy3 (n=edguy3@c-98-221-27-224.hsd1.nj.comcast.net) [NETSPLIT VICTIM] |
| 03:10.09 | *** join/#asterisk sbrath (n=sbrath@unaffiliated/sbrath) [NETSPLIT VICTIM] |
| 03:10.09 | *** join/#asterisk porter (n=terdon@unaffiliated/porter/x-000001) [NETSPLIT VICTIM] |
| 03:10.10 | *** join/#asterisk steve745 (n=W33DS@pool-96-243-156-36.tampfl.fios.verizon.net) [NETSPLIT VICTIM] |
| 03:10.10 | *** join/#asterisk Corydon76-dig (i=indigo@c-69-137-80-31.hsd1.tn.comcast.net) [NETSPLIT VICTIM] |
| 03:10.10 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) [NETSPLIT VICTIM] |
| 03:10.10 | *** join/#asterisk digitalirony (n=digitali@my.ass.looks.just.like.your-face.info) [NETSPLIT VICTIM] |
| 03:10.10 | *** join/#asterisk hohum (i=dcorbe@apollo.corbe.net) [NETSPLIT VICTIM] |
| 03:10.10 | *** join/#asterisk TSM (n=the_soft@fw-lon1.wenn.com) [NETSPLIT VICTIM] |
| 03:10.10 | *** join/#asterisk x86 (n=x86@p3m/member/x86) [NETSPLIT VICTIM] |
| 03:10.10 | *** join/#asterisk t (i=tom@freenode/staff/tomaw) |
| 03:10.10 | *** mode/#asterisk [+oo angler Corydon76-dig] by irc.freenode.net |
| 03:11.28 | *** join/#asterisk mog (n=mog@c-71-228-185-24.hsd1.al.comcast.net) [NETSPLIT VICTIM] |
| 03:11.28 | *** join/#asterisk neurosys (n=neurosys@c-71-196-20-208.hsd1.fl.comcast.net) [NETSPLIT VICTIM] |
| 03:11.28 | *** join/#asterisk ruben23 (n=AGENT@122.55.48.243) [NETSPLIT VICTIM] |
| 03:11.28 | *** join/#asterisk Tech_Travis (n=Administ@cpe-76-168-191-127.socal.res.rr.com) [NETSPLIT VICTIM] |
| 03:11.28 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) [NETSPLIT VICTIM] |
| 03:11.28 | *** join/#asterisk xuser (n=xuser@unaffiliated/xuser) [NETSPLIT VICTIM] |
| 03:11.28 | *** join/#asterisk Badrobot- (n=badrobot@cpe-76-173-229-89.socal.res.rr.com) [NETSPLIT VICTIM] |
| 03:11.28 | *** join/#asterisk Kumbang (n=kumbang@rusnas.paume.itb.ac.id) [NETSPLIT VICTIM] |
| 03:11.28 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) [NETSPLIT VICTIM] |
| 03:11.28 | *** join/#asterisk darkdrgn2k (n=darkdrgn@bas2-toronto44-1242515082.dsl.bell.ca) [NETSPLIT VICTIM] |
| 03:11.28 | *** join/#asterisk catojo (n=catojo@189.24.114.76) [NETSPLIT VICTIM] |
| 03:11.28 | *** join/#asterisk voipmonk (n=shido6@dsl-67-204-37-228.acanac.net) [NETSPLIT VICTIM] |
| 03:11.28 | *** join/#asterisk diatonic_afk (n=diatonic@mail.clearwater-research.com) [NETSPLIT VICTIM] |
| 03:11.28 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) [NETSPLIT VICTIM] |
| 03:11.28 | *** join/#asterisk sspreitzer (n=sspreitz@fedora/sspreitzer) [NETSPLIT VICTIM] |
| 03:11.28 | *** join/#asterisk devmod (n=devmod@c-76-100-208-204.hsd1.md.comcast.net) [NETSPLIT VICTIM] |
| 03:11.28 | *** join/#asterisk Iamnach0 (n=Iamnacho@ip174-70-146-115.ks.ks.cox.net) [NETSPLIT VICTIM] |
| 03:11.28 | *** join/#asterisk JAMMAN2110 (n=james@unaffiliated/jamman2110) [NETSPLIT VICTIM] |
| 03:11.29 | *** join/#asterisk geneticx (n=etnos@adsl-2-204-26.mia.bellsouth.net) [NETSPLIT VICTIM] |
| 03:11.29 | *** join/#asterisk jbw (n=jbw@dsl-105-162.cust.imagine.ie) [NETSPLIT VICTIM] |
| 03:11.29 | *** join/#asterisk steve (i=steve@bouncer.stephen.marsh.name) [NETSPLIT VICTIM] |
| 03:11.30 | *** join/#asterisk mnt_real (n=sinan@bas12-montrealak-1167974851.dsl.bell.ca) |
| 03:11.30 | *** join/#asterisk fish-bulb (n=cstewart@nat/digium/x-mfnyburdhuetlvef) [NETSPLIT VICTIM] |
| 03:11.30 | *** join/#asterisk kisom (n=kisom@c-4edde155.648-1-64736c11.cust.bredbandsbolaget.se) [NETSPLIT VICTIM] |
| 03:11.30 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) [NETSPLIT VICTIM] |
| 03:11.30 | *** join/#asterisk MatBoy (n=MatBoy@wiljewelwetenhe.xs4all.nl) [NETSPLIT VICTIM] |
| 03:11.30 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) [NETSPLIT VICTIM] |
| 03:11.30 | *** join/#asterisk viq (n=viq@unaffiliated/viq) |
| 03:11.30 | *** join/#asterisk angryuser (n=angryuse@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) [NETSPLIT VICTIM] |
| 03:11.30 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) [NETSPLIT VICTIM] |
| 03:11.30 | *** join/#asterisk DarkFibre (n=dmelouk@127.159.119.70.cfl.res.rr.com) [NETSPLIT VICTIM] |
| 03:11.30 | *** join/#asterisk CRCinAU (n=CRCinAU@irc.crc.id.au) [NETSPLIT VICTIM] |
| 03:11.30 | *** join/#asterisk tzafrir (n=tzafrir@local.xorcom.com) [NETSPLIT VICTIM] |
| 03:11.30 | *** join/#asterisk _bradk (n=brad@unaffiliated/-bradk/x-9249860) [NETSPLIT VICTIM] |
| 03:11.30 | *** join/#asterisk mchou (n=quassel@unaffiliated/mchou) [NETSPLIT VICTIM] |
| 03:11.30 | *** join/#asterisk Raden (n=Raden@71.89.121.119) [NETSPLIT VICTIM] |
| 03:11.30 | *** join/#asterisk dkirker-openmobl (n=dkirker@gateway0.openmobl.com) [NETSPLIT VICTIM] |
| 03:11.30 | *** join/#asterisk ectospasm (n=weechat@user-24-236-95-118.knology.net) [NETSPLIT VICTIM] |
| 03:11.30 | *** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) [NETSPLIT VICTIM] |
| 03:11.30 | *** join/#asterisk dzup (i=dzup@support.team.at.shellium.org) [NETSPLIT VICTIM] |
| 03:11.30 | *** join/#asterisk fnordus (n=dnall@70.70.0.215) [NETSPLIT VICTIM] |
| 03:11.30 | *** join/#asterisk Buklov (n=buklov@mail.abcmebel.ru) [NETSPLIT VICTIM] |
| 03:11.30 | *** join/#asterisk war9407 (i=war@liquidswords.org) [NETSPLIT VICTIM] |
| 03:11.30 | *** join/#asterisk dlynes (n=dlynes@bas6-hamilton14-1167992947.dsl.bell.ca) [NETSPLIT VICTIM] |
| 03:11.30 | *** join/#asterisk NET||abuse (n=l@188.141.22.178) [NETSPLIT VICTIM] |
| 03:11.30 | *** mode/#asterisk [+o mog] by irc.freenode.net |
| 03:11.30 | *** join/#asterisk sun28 (n=light@sun28.ipfw.su) [NETSPLIT VICTIM] |
| 03:11.30 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 03:11.32 | *** join/#asterisk denon (n=denon@synapse.subneural.net) [NETSPLIT VICTIM] |
| 03:11.32 | *** join/#asterisk louben (n=lou@212-70-216-131.ath.static.tee.gr) [NETSPLIT VICTIM] |
| 03:11.32 | *** join/#asterisk wimt (i=wimt@freenode/staff/wikipedia.wimt) [NETSPLIT VICTIM] |
| 03:11.32 | *** join/#asterisk mr_ian (n=mr_ian@S0106001b63f49383.du.shawcable.net) [NETSPLIT VICTIM] |
| 03:11.32 | *** join/#asterisk benngard (n=benngard@90-230-92-67-no148.tbcn.telia.com) [NETSPLIT VICTIM] |
| 03:11.32 | *** join/#asterisk evil_gordita (n=evilgord@ip70-188-50-186.rn.hr.cox.net) [NETSPLIT VICTIM] |
| 03:11.32 | *** join/#asterisk werdan7 (n=w7@freenode/staff/wikimedia.werdan7) |
| 03:11.32 | *** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd) [NETSPLIT VICTIM] |
| 03:11.32 | *** join/#asterisk ph8 (i=ph8@unaffiliated/ph8) [NETSPLIT VICTIM] |
| 03:11.32 | *** join/#asterisk waa (n=waa@balrog.credipar.com.br) [NETSPLIT VICTIM] |
| 03:11.32 | *** join/#asterisk chazzm (n=chazz@173-24-238-25.client.mchsi.com) [NETSPLIT VICTIM] |
| 03:11.32 | *** join/#asterisk MuffinMan (n=muffinma@asterisk/issue-tracker-bot/muffinman) |
| 03:11.32 | *** join/#asterisk eppigy (n=Dave@216-139-241-102.aus.us.siteprotect.com) [NETSPLIT VICTIM] |
| 03:11.32 | *** join/#asterisk RobH (n=RobH@cpe-173-169-30-118.tampabay.res.rr.com) [NETSPLIT VICTIM] |
| 03:11.32 | *** join/#asterisk slima (i=slima@unaffiliated/slima) [NETSPLIT VICTIM] |
| 03:11.32 | *** join/#asterisk _Raptor_ (i=raptorbl@andariel.informatik.uni-erlangen.de) [NETSPLIT VICTIM] |
| 03:11.32 | *** join/#asterisk Gido-E (n=gido@lounge.datux.nl) [NETSPLIT VICTIM] |
| 03:11.32 | *** join/#asterisk shafu (n=giany@83.169.0.238) [NETSPLIT VICTIM] |
| 03:11.32 | *** join/#asterisk vk2dgy (n=rossw@ali-syd-3.albury.net.au) [NETSPLIT VICTIM] |
| 03:11.32 | *** join/#asterisk wonderworld (n=ww@ip-62-143-22-226.unitymediagroup.de) [NETSPLIT VICTIM] |
| 03:11.32 | *** join/#asterisk ltd_wk (i=z@patwk.transact.net.au) [NETSPLIT VICTIM] |
| 03:11.32 | *** join/#asterisk afink (n=afink@204.26.87.226) [NETSPLIT VICTIM] |
| 03:11.32 | *** join/#asterisk oldhack (n=jfincher@cpe-24-27-56-221.austin.res.rr.com) [NETSPLIT VICTIM] |
| 03:11.32 | *** join/#asterisk MmixX (i=mixed@61.14.191.133) [NETSPLIT VICTIM] |
| 03:11.32 | *** join/#asterisk dahl (i=dahl@4.84-234-222.customer.lyse.net) [NETSPLIT VICTIM] |
| 03:11.32 | *** join/#asterisk bmoraca (n=bmoraca@66-242-174-254.ceres.bvn.net) [NETSPLIT VICTIM] |
| 03:11.32 | *** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) [NETSPLIT VICTIM] |
| 03:11.33 | *** join/#asterisk bpgoldsb (n=bpgoldsb@ip24-250-198-162.ga.at.cox.net) [NETSPLIT VICTIM] |
| 03:11.33 | *** join/#asterisk russellb (n=russellb@asterisk/digium-open-source-team-lead/russellb) |
| 03:11.33 | *** join/#asterisk bmoraca_work (n=bmoraca@66-242-174-254.ceres.bvn.net) [NETSPLIT VICTIM] |
| 03:11.33 | *** join/#asterisk ccesario (n=ccesario@189-19-6-236.dsl.telesp.net.br) |
| 03:11.34 | *** join/#asterisk pfn (i=pfnguyen@socal.hanhuy.com) |
| 03:11.34 | *** join/#asterisk IckyGone (n=magnus@ada-bcn-fw01.adamoeurope.com) [NETSPLIT VICTIM] |
| 03:11.34 | *** join/#asterisk _mwoodj_ (n=mwoodj@pdpc/sponsor/digium/hyper-eye) [NETSPLIT VICTIM] |
| 03:11.34 | *** join/#asterisk jblack (n=jblack@71.181.248.16) [NETSPLIT VICTIM] |
| 03:11.34 | *** join/#asterisk luckyaba (n=lucky@ip72-194-215-55.sb.sd.cox.net) [NETSPLIT VICTIM] |
| 03:11.34 | *** join/#asterisk securevoip (n=securevo@c-76-123-20-170.hsd1.va.comcast.net) [NETSPLIT VICTIM] |
| 03:11.34 | *** join/#asterisk nighty^ (n=nighty@x122091.ppp.asahi-net.or.jp) [NETSPLIT VICTIM] |
| 03:11.34 | *** join/#asterisk jarrod (i=anon@theos.org) [NETSPLIT VICTIM] |
| 03:11.34 | *** join/#asterisk Zhad (n=tom@host-1.art-it-services.co.uk) [NETSPLIT VICTIM] |
| 03:11.34 | *** join/#asterisk edwin_quijada (n=macaruch@200.26.172.50) [NETSPLIT VICTIM] |
| 03:11.34 | *** join/#asterisk darkskiez (n=dz@62-50-207-10.client.stsn.net) [NETSPLIT VICTIM] |
| 03:11.34 | *** join/#asterisk xpot (n=xpot@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net) |
| 03:11.34 | *** join/#asterisk friehmaen (i=freeman@xers.de) [NETSPLIT VICTIM] |
| 03:11.34 | *** join/#asterisk creativx (n=creadure@197.82-134-19.bkkb.no) [NETSPLIT VICTIM] |
| 03:11.34 | *** join/#asterisk mykhyggz (n=col@evolone.org) [NETSPLIT VICTIM] |
| 03:11.34 | *** join/#asterisk Slashman (n=Slash@ariane.fimasys.com) [NETSPLIT VICTIM] |
| 03:11.34 | *** mode/#asterisk [+ooo denon malcolmd russellb] by irc.freenode.net |
| 03:11.34 | *** join/#asterisk styelz (n=yoohoo@m0o0.mooo.com) [NETSPLIT VICTIM] |
| 03:11.34 | *** join/#asterisk voxter (n=voxter@S010600090f53ea17.vc.shawcable.net) [NETSPLIT VICTIM] |
| 03:11.34 | *** join/#asterisk high-freq (n=hfreq@adsl-99-41-143-147.dsl.ksc2mo.sbcglobal.net) [NETSPLIT VICTIM] |
| 03:11.34 | *** join/#asterisk mnicholson_ (n=mnichols@nat/digium/x-sydrdavitizqzvaa) |
| 03:11.34 | *** join/#asterisk mrbnet_ (n=mrbnet@74-95-100-233-Minnesota.hfc.comcastbusiness.net) [NETSPLIT VICTIM] |
| 03:11.34 | *** join/#asterisk xtrac020 (n=xtrac020@84-203-45-202.mysmart.ie) [NETSPLIT VICTIM] |
| 03:11.34 | *** join/#asterisk cnu (i=cnu@113.80-203-44.nextgentel.com) [NETSPLIT VICTIM] |
| 03:11.34 | *** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) [NETSPLIT VICTIM] |
| 03:11.34 | *** join/#asterisk miloux (n=KVIrc@milu.rit.se) [NETSPLIT VICTIM] |
| 03:11.34 | *** join/#asterisk felipe_ (n=felipe@my.nada.kth.se) |
| 03:11.35 | *** join/#asterisk Winkie (n=urmom@ur.fa.gs) [NETSPLIT VICTIM] |
| 03:11.35 | *** join/#asterisk Pan3D (n=Pan3D@streaming.sensoryresearch.net) [NETSPLIT VICTIM] |
| 03:11.35 | *** join/#asterisk ChanServ (ChanServ@services.) |
| 03:11.35 | *** join/#asterisk ecrane (n=ecrane@o1-69-19-166-10.static.o1.com) [NETSPLIT VICTIM] |
| 03:11.35 | *** join/#asterisk Kobaz (n=kobaz@its.kobaz.net) [NETSPLIT VICTIM] |
| 03:11.35 | *** join/#asterisk Qwell (i=north@pdpc/sponsor/digium/Qwell) |
| 03:11.36 | *** join/#asterisk Talkradio (i=talkradi@linuxgeneration.ca) [NETSPLIT VICTIM] |
| 03:11.36 | *** join/#asterisk b14ck (n=comradeb@cpe-24-24-136-239.socal.res.rr.com) [NETSPLIT VICTIM] |
| 03:11.36 | *** join/#asterisk yang (i=yang@freenode/sponsor/cacert.assurer.yang) |
| 03:11.36 | *** join/#asterisk astrutt (n=astrutt@67.93.60.229) [NETSPLIT VICTIM] |
| 03:11.36 | *** join/#asterisk dmz (n=dmz@64.203.207.101.dyn-cm-pool-54.hargray.net) |
| 03:11.36 | *** join/#asterisk tompaw (n=tompaw@slave20.tesserakt.eu) [NETSPLIT VICTIM] |
| 03:11.36 | *** join/#asterisk file (n=file@asterisk/developer-and-muffin-lover/file) |
| 03:11.36 | *** join/#asterisk ThoMe (i=tm@tm.muc.de) [NETSPLIT VICTIM] |
| 03:11.36 | *** join/#asterisk bmg505 (n=leon@196-209-77-212-rndf-esr-5.dynamic.isadsl.co.za) [NETSPLIT VICTIM] |
| 03:11.36 | *** join/#asterisk Nugget (i=nugget@carrera.macnugget.org) [NETSPLIT VICTIM] |
| 03:11.36 | *** join/#asterisk avleen (n=avleen@chiku.silverwraith.com) [NETSPLIT VICTIM] |
| 03:11.36 | *** join/#asterisk mazpe (n=mazpe@ec2-174-129-37-13.compute-1.amazonaws.com) [NETSPLIT VICTIM] |
| 03:11.36 | *** join/#asterisk SkramX (i=mark@phalse.2600.COM) [NETSPLIT VICTIM] |
| 03:11.36 | *** join/#asterisk mace (n=mace@debian/developer/mace) [NETSPLIT VICTIM] |
| 03:11.36 | *** join/#asterisk scardinal (n=supreme@90.184.100.119) [NETSPLIT VICTIM] |
| 03:11.36 | *** join/#asterisk linagee6 (n=linagee@barghest-1-pt.tunnel.tserv3.fmt2.ipv6.he.net) [NETSPLIT VICTIM] |
| 03:11.36 | *** join/#asterisk aiksa[LV] (n=aiksa[LV@mx.fiveplus.lv) [NETSPLIT VICTIM] |
| 03:11.36 | *** join/#asterisk drfreeze (n=Jim@207.191.114.82) [NETSPLIT VICTIM] |
| 03:11.36 | *** join/#asterisk apignard (n=apignard@arnaud.staff.frontier.fr) [NETSPLIT VICTIM] |
| 03:11.36 | *** join/#asterisk defswork (n=andy@mx2.3gcomms.co.uk) [NETSPLIT VICTIM] |
| 03:11.36 | *** join/#asterisk terracon (n=greisky@CPE001cf097a750-CM0012254076d6.cpe.net.cable.rogers.com) [NETSPLIT VICTIM] |
| 03:11.36 | *** join/#asterisk jdoe (i=ykk@falseprophet.ca) [NETSPLIT VICTIM] |
| 03:11.36 | *** join/#asterisk jackal (n=jackal@pool-96-247-205-46.clppva.fios.verizon.net) [NETSPLIT VICTIM] |
| 03:11.36 | *** join/#asterisk Dibbler (n=Dibbler@87-194-103-72.bethere.co.uk) [NETSPLIT VICTIM] |
| 03:11.36 | *** join/#asterisk vty (n=arlf@174.120.203.106) [NETSPLIT VICTIM] |
| 03:11.36 | *** join/#asterisk VxJasonxV (n=jason@xmms2/troll/VxJasonxV) [NETSPLIT VICTIM] |
| 03:11.36 | *** join/#asterisk carrar (i=tim@osburn.com) [NETSPLIT VICTIM] |
| 03:11.37 | *** join/#asterisk dmast (n=dmast@exchange.newpointe.org) [NETSPLIT VICTIM] |
| 03:11.37 | *** join/#asterisk `Sauron (n=sauron@dsl001-130-033.aus1.dsl.speakeasy.net) [NETSPLIT VICTIM] |
| 03:11.37 | *** join/#asterisk rbd_ (n=rbd@rrcs-98-101-33-14.midsouth.biz.rr.com) [NETSPLIT VICTIM] |
| 03:11.37 | *** join/#asterisk zamba (i=marius@flage.org) [NETSPLIT VICTIM] |
| 03:11.37 | *** join/#asterisk Woody2143 (n=Woody214@machine76.Level3.com) [NETSPLIT VICTIM] |
| 03:11.37 | *** join/#asterisk keith4 (n=keith@unaffiliated/keith4) [NETSPLIT VICTIM] |
| 03:11.38 | *** join/#asterisk Failrar (n=Failrar@2001:470:1f15:316:2a0:d1ff:fe4e:e802) [NETSPLIT VICTIM] |
| 03:11.38 | *** join/#asterisk Obeliks (i=obeliks@gentoo/contributor/Obeliks) [NETSPLIT VICTIM] |
| 03:11.38 | *** join/#asterisk Zambezi (i=Zulu@unaffiliated/zambezi) [NETSPLIT VICTIM] |
| 03:11.38 | *** mode/#asterisk [+ooo ChanServ Qwell file] by irc.freenode.net |
| 03:11.38 | *** join/#asterisk lost_sou1 (n=noymfb@cpe-74-71-234-100.twcny.res.rr.com) [NETSPLIT VICTIM] |
| 03:11.38 | *** join/#asterisk razu (n=razu@razu.data.ee) [NETSPLIT VICTIM] |
| 03:11.38 | *** join/#asterisk boch (n=fran@200.61.191.9) [NETSPLIT VICTIM] |
| 03:11.38 | *** join/#asterisk tris (i=tristan@207.241.238.17) |
| 03:11.38 | *** join/#asterisk Linuturk (n=linuturk@unaffiliated/linuturk) [NETSPLIT VICTIM] |
| 03:11.38 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) [NETSPLIT VICTIM] |
| 03:11.38 | *** join/#asterisk naxxfish (n=fish@barney.naxxfish.eu) [NETSPLIT VICTIM] |
| 03:11.38 | *** join/#asterisk teknoprep (n=Chris@unaffiliated/teknoprep) [NETSPLIT VICTIM] |
| 03:11.38 | *** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) [NETSPLIT VICTIM] |
| 03:11.38 | *** join/#asterisk micols (n=mio@rlogin.dk) [NETSPLIT VICTIM] |
| 03:11.38 | *** join/#asterisk memph (n=memph@sd1438.sivit.org) [NETSPLIT VICTIM] |
| 03:11.38 | *** join/#asterisk phix (n=threat@123-243-44-131.tpgi.com.au) [NETSPLIT VICTIM] |
| 03:11.38 | *** join/#asterisk skymeyer (n=skymeyer@mailout.itconnect.be) [NETSPLIT VICTIM] |
| 03:11.38 | *** join/#asterisk steliosk (n=Stelios@ipa107.2.tellas.gr) |
| 03:11.38 | *** join/#asterisk mbranca (n=matteo@host139-217-static.224-95-b.business.telecomitalia.it) [NETSPLIT VICTIM] |
| 03:11.38 | *** join/#asterisk dlu_ (n=quassel@83.125.8.7) [NETSPLIT VICTIM] |
| 03:11.38 | *** join/#asterisk [netman] (n=netman@80.31.164.91) [NETSPLIT VICTIM] |
| 03:11.38 | *** join/#asterisk Superbartt (n=bart@ipd50a21c9.speed.planet.nl) [NETSPLIT VICTIM] |
| 03:11.38 | *** join/#asterisk thehar (i=thehar@thehar.xmission.com) [NETSPLIT VICTIM] |
| 03:11.38 | *** join/#asterisk jql (n=jql@12.9a.344a.static.theplanet.com) [NETSPLIT VICTIM] |
| 03:11.38 | *** join/#asterisk dwayne (n=dwayne@c-71-207-214-190.hsd1.al.comcast.net) |
| 03:11.38 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-99-199-10.ph.ph.cox.net) [NETSPLIT VICTIM] |
| 03:11.38 | *** join/#asterisk Chinorro (n=Chino@19.226.117.91.dynamic.mundo-r.com) [NETSPLIT VICTIM] |
| 03:11.39 | *** join/#asterisk alfa202 (n=svelluto@dhcp-0-9-e8-4a-96-80.cpe.quickclic.net) [NETSPLIT VICTIM] |
| 03:11.39 | *** join/#asterisk Arsenick (n=rpurcell@modemcable022.82-21-96.mc.videotron.ca) [NETSPLIT VICTIM] |
| 03:11.39 | *** join/#asterisk ZenBSDi (n=ZenBSDi@unaffiliated/ZenBSDi) [NETSPLIT VICTIM] |
| 03:11.44 | *** join/#asterisk t (i=tom@freenode/staff/tomaw) |
| 03:15.18 | *** join/#asterisk sebbl (n=Momofu@HSI-KBW-078-043-193-153.hsi4.kabel-badenwuerttemberg.de) |
| 03:31.52 | *** part/#asterisk ruben23 (n=AGENT@122.55.48.243) |
| 03:35.51 | *** join/#asterisk Defraz (n=Defraz@c72co-edge-router.fuzecore.com) |
| 03:38.18 | *** join/#asterisk sjobeck (n=sjobeck@valdisere.sjobeck.com) |
| 03:39.04 | *** part/#asterisk sjobeck (n=sjobeck@valdisere.sjobeck.com) |
| 03:40.29 | *** join/#asterisk _bugz_ (n=bugz@adsl-76-194-247-113.dsl.lsan03.sbcglobal.net) |
| 04:00.25 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.229.58) |
| 04:03.03 | *** join/#asterisk Kumbang (n=kumbang@rusnas.paume.itb.ac.id) |
| 04:06.30 | sun28 | moin |
| 04:07.33 | *** join/#asterisk OrNix (n=ornix@l151-249-47.static.cn.ru) |
| 04:07.46 | *** join/#asterisk ruben23 (n=AGENT@122.55.48.243) |
| 04:08.45 | *** join/#asterisk sgimeno (n=santiago@226.Red-80-33-64.staticIP.rima-tde.net) |
| 04:10.17 | *** join/#asterisk wonderworld (n=ww@ip-62-143-22-226.unitymediagroup.de) |
| 04:28.21 | *** join/#asterisk reptiles (i=jim@198.96.210.227) |
| 04:31.23 | *** join/#asterisk GameGamer43 (n=GameGame@0101-46-27-72-DYNAMIC-dsl.cwjamaica.com) |
| 04:32.48 | *** join/#asterisk GameGamer43 (n=GameGame@CPE-65-27-76-78.new.res.rr.com) |
| 04:37.56 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 04:44.48 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat/x-yythlhuwknfnolfi) |
| 04:49.37 | *** join/#asterisk GameGamer43 (n=GameGame@CPE-65-27-76-78.new.res.rr.com) |
| 04:58.40 | *** join/#asterisk soman (n=somnath@118.102.130.6) |
| 05:05.38 | *** join/#asterisk brut- (n=brut-@h66-173-4-254.mntimn.dedicated.static.tds.net) |
| 05:08.50 | *** part/#asterisk ruben23 (n=AGENT@122.55.48.243) |
| 05:13.59 | *** part/#asterisk voipmonk (n=shido6@dsl-67-204-37-228.acanac.net) |
| 05:15.48 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 05:33.58 | *** join/#asterisk chendy (n=chatzill@119.137.95.228) |
| 05:39.41 | *** join/#asterisk sun28 (n=light@sun28.ipfw.su) |
| 05:41.16 | *** join/#asterisk ticoit (n=ticoit@201.191.151.155) |
| 05:41.37 | *** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano) |
| 05:42.16 | drmessano | does the Asterisk fist pump |
| 05:46.58 | *** join/#asterisk JAMMAN2110 (n=james@unaffiliated/jamman2110) |
| 05:58.16 | *** part/#asterisk benngard (n=benngard@90-230-92-67-no148.tbcn.telia.com) |
| 05:59.07 | *** join/#asterisk denon (n=denon@synapse.subneural.net) |
| 05:59.07 | *** mode/#asterisk [+o denon] by ChanServ |
| 06:02.22 | *** join/#asterisk brookshire (i=mbrooks@hijacked.us) |
| 06:21.40 | *** join/#asterisk Faustov (i=user@gentoo/user/faustov) |
| 06:22.25 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 06:41.42 | *** join/#asterisk cuco (n=Diego@local.xorcom.com) |
| 06:47.21 | ChannelZ | stumbles around dizzy |
| 06:58.21 | *** join/#asterisk Ad-Hoc (n=nimbus@62.169.216.185) |
| 07:00.19 | *** join/#asterisk |Cybex| (n=John@atwork-21.r-212.178.82.atwork.nl) |
| 07:14.53 | *** join/#asterisk smooth_penguin (n=smoove@59.96.95.187) |
| 07:26.53 | *** join/#asterisk sgimeno (n=santiago@226.Red-80-33-64.staticIP.rima-tde.net) |
| 07:32.21 | *** join/#asterisk lordmortis (n=lordmort@203-8-160-250.secure.com.au) |
| 07:59.28 | *** join/#asterisk sebbl (n=Momofu@HSI-KBW-078-043-193-153.hsi4.kabel-badenwuerttemberg.de) |
| 07:59.55 | *** join/#asterisk Defraz (n=Defraz@c72co-edge-router.fuzecore.com) |
| 08:02.54 | *** join/#asterisk lftsy (n=mlr@install.deckpoint.ch) |
| 08:10.12 | *** join/#asterisk icyValk77 (n=icyValk7@gateway.ash.thebunker.net) |
| 08:22.24 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 08:26.56 | *** join/#asterisk retentiveboy (n=pdugas@atl.pra-corp.com) |
| 08:31.24 | *** join/#asterisk smooth_penguin (n=smoove@59.96.95.187) |
| 08:34.05 | *** join/#asterisk tamiel (n=tamiel@213.30.183.226) |
| 08:34.27 | *** join/#asterisk sgimeno (n=santiago@226.Red-80-33-64.staticIP.rima-tde.net) |
| 08:34.37 | *** join/#asterisk Tim_Toady (n=moi@77.49.24.242.dsl.dyn.forthnet.gr) |
| 08:38.18 | *** join/#asterisk chendy_ (n=chatzill@119.137.94.8) |
| 08:43.55 | *** join/#asterisk TSM2 (n=the_soft@87-194-32-212.bethere.co.uk) |
| 08:52.59 | *** part/#asterisk Tech_Travis (n=Administ@cpe-76-168-191-127.socal.res.rr.com) |
| 08:53.51 | *** join/#asterisk devmod (n=devmod@c-76-100-208-204.hsd1.md.comcast.net) |
| 08:53.53 | devmod | night |
| 08:54.21 | devmod | anyone knows how can I record video prompts? or convert .avi to .h263/4 ? |
| 08:56.37 | tzafrir | devmod, ffmpeg ? |
| 08:57.26 | *** join/#asterisk mikkel (n=mikkel@130.226.36.170) |
| 08:58.47 | devmod | i thought there were some weird headers at the beginning of the files? |
| 09:03.22 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 09:24.05 | *** join/#asterisk d00gster (n=doughant@94.99.115.204) |
| 09:27.39 | *** join/#asterisk Da-Geek (n=Da-Geek@62.189.17.99) |
| 09:57.51 | *** join/#asterisk s0lid (n=s0lid@112.201.196.45) |
| 09:58.19 | *** join/#asterisk darkskiez (n=dz@62-50-207-10.client.stsn.net) |
| 10:00.15 | *** join/#asterisk pawz (n=pawz@ppp118-208-39-3.lns20.bne1.internode.on.net) |
| 10:00.54 | *** join/#asterisk darkskiez_ (n=dz@62-50-207-67.client.stsn.net) |
| 10:08.01 | *** join/#asterisk TommyBotten (i=tommy@broken.pipe.no) |
| 10:12.15 | *** join/#asterisk casix (n=casix@xenpbxedifici.adamvozip.es) |
| 10:12.17 | casix | hello |
| 10:13.50 | casix | I'm using the G option to play a dynamic playback to the called. After the playback I would like to bridge the channel but I don't know how to do it. Any ideas? thx |
| 10:21.25 | *** join/#asterisk ayrjola (n=ayrjola@gw.ccxtech.fi) |
| 10:24.06 | *** join/#asterisk bzing2 (n=dr105@dhcp-194-66-208-235.canterbury.ac.uk) |
| 10:24.20 | *** join/#asterisk waa (n=waa@balrog.credipar.com.br) |
| 10:28.06 | *** join/#asterisk shamelessn00b (n=chatzill@58-65-172-114.nayatel.pk) |
| 10:28.10 | shamelessn00b | hi guys |
| 10:29.50 | shamelessn00b | I need to do a simple modification in the mp3 applicationof asterisk 1.6.2.0 |
| 10:30.32 | shamelessn00b | whenever user presses a key whilst an mp3 is being played instead of jumping on the next line in the dialplan I want the user to land on a specific portion of the dialplan |
| 10:30.40 | shamelessn00b | like its done in the meetme application |
| 10:32.51 | *** join/#asterisk mpe (n=mpe@0x4dd624b2.adsl.cybercity.dk) |
| 10:34.45 | tzafrir | shamelessn00b, why not do it in the dialplan? |
| 10:35.06 | *** join/#asterisk Akiraa (n=Akira@92.81.199.70) |
| 10:35.19 | shamelessn00b | tzafrir I cant get the dtmf user pressed when it exits the mp3 application |
| 10:35.36 | shamelessn00b | it just gets the user into the next line in the dialplan |
| 10:36.05 | tzafrir | shamelessn00b, Background(some-sound) |
| 10:36.30 | shamelessn00b | im playing an mp3 stream |
| 10:36.36 | tzafrir | WaitForDigit() ; or whatever it is called |
| 10:36.59 | shamelessn00b | the application flw is something like this |
| 10:37.05 | shamelessn00b | user is listening to an mp3 stream |
| 10:37.14 | tzafrir | shamelessn00b, so what you need is an option to play it in the background |
| 10:37.22 | shamelessn00b | and he can press a defined digit say * |
| 10:37.29 | shamelessn00b | whenever the user presses * |
| 10:37.33 | shamelessn00b | the stream playback stops |
| 10:37.55 | shamelessn00b | and the song in the mp3 stream is set as the user ring back tone |
| 10:38.29 | shamelessn00b | and the next line in dialplan takes the user back to a set of menu choiuces |
| 10:39.14 | Akiraa | Anyone field tested Skype and Skype Out? |
| 10:39.36 | shamelessn00b | I was doing it in the meetme app (using musiconhold with custom option) but was having issues with it |
| 10:40.06 | shamelessn00b | whenever the user pressed a key and got out of the meetme app the stream playback stopped for all the users |
| 10:40.28 | shamelessn00b | the meetme app somehow terminated the process that was running the stream |
| 10:44.36 | shamelessn00b | tzafrir: the background solution would have worked but I am playing a live radio stream |
| 10:44.41 | shamelessn00b | from icecast server |
| 10:58.23 | shamelessn00b | tzafrir: any ideas?? |
| 10:58.29 | *** join/#asterisk mpe_ (n=mpe@gate.ipvision.dk) |
| 11:02.29 | *** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy) |
| 11:03.00 | *** join/#asterisk darkskiez (n=dz@62-50-207-10.client.stsn.net) |
| 11:03.09 | *** join/#asterisk GNU\colossus (i=anonymou@89.110.149.123) |
| 11:04.02 | *** join/#asterisk hluesea (n=hulusika@88.247.127.66) |
| 11:04.11 | *** join/#asterisk Akiraa (n=Akira@92.81.199.70) |
| 11:08.37 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat/x-wnjmwuwflfxlqboj) |
| 11:13.16 | GNU\colossus | hi folks |
| 11:14.29 | GNU\colossus | we're having rather wird problems with our asterisk server here. seemingly at random, we cannot hear the remote user (rx breaks), but they an hear us just fine (tx works) - any idea what could be the cause of this? |
| 11:15.31 | UQlev | GNU\colossus: is your * behind firewall? |
| 11:15.39 | Akiraa | GNU\colossus: NAT issues most likely |
| 11:16.00 | *** join/#asterisk ManxPower-work (n=EWieling@216.186.151.147) |
| 11:16.08 | GNU\colossus | UQlev: what's my *? Akiraa: we're behind NAT, but our asterisk box is connected to an external VOIP provider |
| 11:16.46 | GNU\colossus | Akiraa: please note that this happens in the midst of calls. |
| 11:17.08 | UQlev | GNU\colossus: does your asterisk (*) server have public IP or nutted private IP? |
| 11:17.59 | GNU\colossus | UQlev: ah, nifty shorthand there. yes, we're behind NAT. but connected to an external voip provider. |
| 11:18.22 | Akiraa | some ISPs can fuck with traffic intentionally, but that's hard to pinpoint |
| 11:18.46 | ManxPower-work | It means nothing that you are "connected to an external provider" |
| 11:18.48 | Akiraa | GNU\colossus: are you using SIP or IAX2? |
| 11:19.20 | casix | I'm using the G option to play a dynamic playback to the called. After the playback I would like to bridge the channel but I don't know how to do it. Any ideas? thx |
| 11:20.02 | GNU\colossus | Akiraa: our ISP is also our VOIP provider, and they insist it's a problem at our endpoint. Akiraa: please pardon my stupidity, but how can I quickly check that? |
| 11:20.18 | Akiraa | GNU\colossus: if SIP, then the VoIP provider may need to be aware you are behind NAT (some serverside adjustments may be needed) |
| 11:20.28 | UQlev | GNU\colossus: all VoIP servises are very sensitive to quality of NAT-router. The best solution is to run your asterisk on a host with direct public interface |
| 11:20.37 | ManxPower-work | ~answers |
| 11:20.38 | infobot | well, answers is Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 11:20.48 | ManxPower-work | There's how to set up Asterisk when it is behind NAT. |
| 11:23.15 | UQlev | GNU\colossus: at the beginning I had weird problem when after 2-3 days after router restart my asterisk got missing packets, and then I had to restart router. Later I have got rid of router and no problems arose within a year or so |
| 11:24.48 | ManxPower-work | casix: Your question requires too much effort to parse. |
| 11:26.24 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat/x-cuwxqjvsrtlkqxwt) |
| 11:27.44 | GNU\colossus | after fiddling with the * console, I think we're using iax2 |
| 11:28.14 | ManxPower-work | GNU\colossus: You THINK? We can't help you if you don't know your own system. |
| 11:28.26 | casix | ManxPower-work: with option G in the dial the called channel is redirected to x priority and the caller to the x+1 priority. There I playback some messages and after that I would like to bridge the calls like a normal dial. Is that possible? |
| 11:28.33 | shamelessn00b | I want to modify the mp3player application a bit |
| 11:28.39 | ManxPower-work | casix: that should happen automatically |
| 11:28.44 | Faustov | GNU\colossus: hi :> |
| 11:28.51 | GNU\colossus | Faustov: hi there :D |
| 11:29.25 | shamelessn00b | whenever the user presses a key, instead of simply exiting the application and jumping on to the next line in the dialplan, I want the user to go to a specific portion in the dialplan depending on which key is pressed |
| 11:29.56 | GNU\colossus | ManxPower-work: I'm terribly sorry, but I'm not our on-site * "expert". still, I've been told to tap into the community to look for suggestions on what could be wrong with our setup. |
| 11:30.28 | ManxPower-work | GNU\colossus: Unfortunately there is not much we can do without knowing anything. |
| 11:31.43 | GNU\colossus | `iax2 show registry` shows a host from the IP-range of our ISP we're supposedly connected to. so I guess we're using AXP2 as protocol. there are no listed SIP subscriptions on the server in question. |
| 11:32.43 | ManxPower-work | GNU\colossus: So nothing is shown in "sip show subscriptions"? How about "sip show registry" or "sip show peers". |
| 11:33.23 | ManxPower-work | GNU\colossus: Are you using a GUI version of Asterisk? |
| 11:33.25 | GNU\colossus | ManxPower-work: nothing either. |
| 11:33.28 | GNU\colossus | ManxPower-work: CLI only |
| 11:33.52 | casix | ManxPower-work: I have this dialplan, but after that the call is hangup: http://pastebin.org/74222 I'm using asterisk 1.4.26.2 |
| 11:34.04 | casix | it is not bridged |
| 11:34.20 | ManxPower-work | casix: that is not a dialplan. dialplan extensions start with priority 1 |
| 11:35.03 | ManxPower-work | GNU\colossus: pastebin the out out if "iax2 show registry" and "iax2 show peers" |
| 11:35.04 | casix | yes yes ok I've cutted it a little, I have more things before this part |
| 11:35.06 | ManxPower-work | ~pb |
| 11:35.07 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
| 11:37.27 | GNU\colossus | ManxPower-work: http://pasted.at/6760770f4a.html |
| 11:38.37 | ManxPower-work | GNU\colossus: What, exactly, is this server used for? You said you have no sip peers and you have only one iax2 connection. Are all your phones Zap or DAHDI?? |
| 11:40.06 | casix | ManxPower-work: yes yes ok I've cutted it a little, I have more things before this part, get some variables from mysql... |
| 11:41.25 | ManxPower-work | casix: show the CLI output of a failed call. NO CUTTING! |
| 11:42.07 | GNU\colossus | ManxPower-work: it's a virtual machine running ubuntu 8.04 LTS on a rather potent host running Linux-KVM. the only services it's running are openssh and asterisk. by phones, do you mean the physical/software IP phones our marketing dep. uses to actually make calls? |
| 11:42.51 | ManxPower-work | GNU\colossus: I'm sorry, I cannot help you futher. |
| 11:42.58 | ManxPower-work | further |
| 11:43.09 | GNU\colossus | thanks for your time, anyway :) |
| 11:47.47 | *** join/#asterisk _cgc (n=_cgc@94-193-99-128.zone7.bethere.co.uk) |
| 11:47.57 | _cgc | morning everyone |
| 11:48.16 | casix | ManxPower-work: here is: http://pastebin.org/74228 |
| 11:50.17 | _cgc | does anyone know why when making calls over a sip trunk the sound does not work 1 way? |
| 11:50.30 | _cgc | http://pastebin.ca/1746758 <---- a copy of the call with sip debug |
| 11:52.10 | *** join/#asterisk smooth_penguin (n=smoove@59.95.11.101) |
| 11:52.38 | Akiraa | Is there any instance where ISDN/DSL/ADSL would be useful if you're not a POTS phone company with existing infrastructure? |
| 11:52.55 | shamelessn00b | ManxPower-work: |
| 11:53.00 | shamelessn00b | can has halp? |
| 11:53.52 | *** join/#asterisk smooth_penguin (n=smoove@59.95.11.101) |
| 11:54.22 | tzafrir | what can? |
| 11:54.57 | *** join/#asterisk jkroon (n=jkroon@dsl-244-41-117.telkomadsl.co.za) |
| 11:55.27 | *** join/#asterisk smooth_penguin (n=smoove@59.95.11.101) |
| 11:56.12 | ManxPower-work | tzafrir: he wants to modify app_mp3playback or whatever the app is. |
| 11:57.04 | *** join/#asterisk smooth_penguin (n=smoove@59.95.11.101) |
| 11:58.46 | *** join/#asterisk smooth_penguin (n=smoove@59.95.11.101) |
| 12:00.21 | *** join/#asterisk smooth_penguin (n=smoove@59.95.11.101) |
| 12:02.40 | ManxPower-work | casix: looks like G wants G(context^exten^pri) and you are only doing G(pri) |
| 12:04.26 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 12:04.44 | *** join/#asterisk catojo (n=catojo@189.24.114.76) |
| 12:05.49 | casix | ManxPower-work: ok, I try it now, but the playbacks are doned ok, the problem is that after that the 2 legs are hangup, not bridged |
| 12:06.17 | casix | ManxPower-work: it don't work either |
| 12:06.56 | casix | s/either/nor/ |
| 12:08.04 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 12:08.51 | *** part/#asterisk CRCinAU (n=CRCinAU@irc.crc.id.au) |
| 12:09.01 | *** join/#asterisk CRCinAU_ (n=CRCinAU@irc.crc.id.au) |
| 12:09.08 | CRCinAU_ | Anyone know anything about T38 and asterisk? |
| 12:11.11 | ManxPower-work | casix: I have no more suggestions |
| 12:11.34 | casix | ok, thank you, i will keep searching |
| 12:15.26 | *** join/#asterisk mpe_ (n=mpe@gate.ipvision.dk) |
| 12:15.42 | *** join/#asterisk mpe (n=mpe@gate.ipvision.dk) |
| 12:17.23 | shamelessn00b | how can I get a dtmf using asterisk low level functions |
| 12:17.46 | shamelessn00b | from a data frame |
| 12:18.20 | shamelessn00b | ast_frame f; |
| 12:18.25 | shamelessn00b | I have this variable |
| 12:18.53 | tzafrir | shamelessn00b, generally it is in dsp.c |
| 12:18.58 | shamelessn00b | <PROTECTED> |
| 12:19.08 | shamelessn00b | if (f->frametype == AST_FRAME_DTMF) |
| 12:19.10 | *** part/#asterisk xtrac020 (n=xtrac020@84-203-45-202.mysmart.ie) |
| 12:19.17 | shamelessn00b | this says that if the frame contains a DTMF |
| 12:19.22 | tzafrir | Also: what do you need to do in case of a digit? |
| 12:19.26 | *** join/#asterisk xtrac020 (n=xtrac020@84-203-45-202.mysmart.ie) |
| 12:19.29 | CRCinAU_ | hmmm - it seems nobody knows squatt about T38 and asterisk :p |
| 12:19.33 | tzafrir | e.g.: can you use feature codes instead? |
| 12:19.35 | CRCinAU_ | even google knows squatt. |
| 12:19.40 | shamelessn00b | I'll just save it as an environment variable |
| 12:19.42 | tzafrir | (features.conf) |
| 12:19.47 | shamelessn00b | and access it in my dialplan |
| 12:19.56 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 12:20.05 | shamelessn00b | the application is going to exit as the dtmf is recieved |
| 12:21.00 | shamelessn00b | any idea? |
| 12:21.24 | shamelessn00b | how do I access the actual data in the frame |
| 12:21.36 | shamelessn00b | not just tell what type of data the frame contains |
| 12:22.31 | shamelessn00b | http://docs.freeswitch.org/structast__frame.html#428eb62de8abc2cb3612ea5dfa96a3d6 |
| 12:23.08 | ManxPower-work | CRCinAU_: T-38 implimentations are buggy, instable, and have major interop issues. You are correct, most people don't think it is worth the major amounts of work to make it work. |
| 12:23.22 | ManxPower-work | shamelessn00b: try #asterisk-dev |
| 12:24.16 | *** join/#asterisk Caplain (i=shayne@84-141.35-65.tampabay.res.rr.com) |
| 12:24.33 | Caplain | exten => _10|X.,1,Macro(dialprovider) |
| 12:24.36 | Caplain | is that correct? |
| 12:24.47 | Caplain | the extension part at least |
| 12:24.48 | *** join/#asterisk shamelessn00b (n=chatzill@58-65-172-114.nayatel.pk) |
| 12:24.56 | shamelessn00b | ok thanks ManxPower-work |
| 12:24.56 | ManxPower-work | Caplain: no. |
| 12:25.45 | Caplain | what would be correct to match something like 10*266300 and have it cut off 10? |
| 12:26.22 | ManxPower-work | exten => _10*XXXXXX,1,Something(${EXTEN:2}) |
| 12:26.58 | ManxPower-work | Caplain: extension patterns are NOT regex's |
| 12:27.14 | Caplain | good, regex hates me |
| 12:28.06 | Caplain | Executing [s@macro-dialsipbroker:2] Dial("Local/10*266300@default-0c25,2", "SIP/sipbroker-out/10*266300") in new stack |
| 12:28.14 | Caplain | yeah its still passing the 10 to it |
| 12:28.27 | ManxPower-work | Caplain: then you are doing it wrong |
| 12:28.35 | Caplain | obviously |
| 12:28.41 | Caplain | which is why i came here |
| 12:28.44 | ManxPower-work | maybe if you pasted the ACTUAL line? |
| 12:28.56 | ManxPower-work | ~pb |
| 12:28.57 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
| 12:30.30 | Caplain | http://pastebin.com/m3ee3192 |
| 12:31.04 | Caplain | zpaste never allows long enough postage :( |
| 12:31.09 | Caplain | err dpaste |
| 12:31.29 | casix | Caplain: line 33 you need a 2 not a 0 |
| 12:31.30 | ManxPower-work | What the hell is this: exten => _10*X.,1,Macro(dialsipbroker,${EXTEN:0}) ; SIP-Code dialing |
| 12:31.52 | ManxPower-work | ${EXTEN:0} != ${EXTEN:2} !!!! |
| 12:32.00 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 12:32.35 | Caplain | ManxPower-work, oh? well its copy pasted and i havent dealt with this stuff in years |
| 12:33.02 | ManxPower-work | Caplain: Maybe so but I GAVE you the correct EXTEN |
| 12:33.14 | ManxPower-work | (7:26:21 AM) ManxPower-work: exten => _10*XXXXXX,1,Something(${EXTEN:2}) |
| 12:33.14 | Caplain | yes, thanks |
| 12:33.27 | ManxPower-work | go fix it. |
| 12:33.52 | Caplain | works!!!! :) |
| 12:33.56 | Caplain | thanks :-D |
| 12:34.10 | ManxPower-work | Caplain: Don't worry, you'll have plenty of other problems. |
| 12:34.33 | Caplain | actually no, the rest is copying and pasting what i just did over and over |
| 12:34.58 | ManxPower-work | no, since you don't even know what the :number after a variable you're going to have many other issues |
| 12:36.21 | Caplain | ManxPower-work, i didn't know that elecromagnetism was radio waves a few months ago and now i have a homebrew 5 mile wifi antenna |
| 12:36.23 | Caplain | so yeah :/ |
| 12:36.26 | Caplain | i learn fast |
| 12:37.38 | ManxPower-work | Caplain: I wish you the BEST of luck. |
| 12:37.53 | Caplain | :-D thanks |
| 12:37.55 | Caplain | ill need it |
| 12:38.18 | CRCinAU_ | ManxPower-work: Digium have created a commercial fax thing for asterisk |
| 12:38.28 | CRCinAU_ | so it's obviously something someone wants.... |
| 12:38.36 | CRCinAU_ | but there's shit all in the way of docs :( |
| 12:38.49 | ManxPower-work | CRCinAU_: What makes you think they created it for T.38? |
| 12:39.01 | ManxPower-work | virtually all faxing in the world is NOT T.38 |
| 12:39.37 | CRCinAU_ | http://www.digium.com/en/products/software/faxforasterisk.php <<-- this. |
| 12:40.24 | ManxPower-work | CRCinAU_: then go contact them for support. |
| 12:40.53 | *** join/#asterisk Morlac (n=_morlac_@93.95.201.250) |
| 12:41.00 | Morlac | guys |
| 12:41.04 | Morlac | need help |
| 12:41.13 | ManxPower-work | ~ask |
| 12:41.14 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 12:41.34 | Morlac | I need to convince someone that the NEC univerge is no good...am trying to convince him of switchvox |
| 12:41.47 | Morlac | I have no knowledge about the univerge |
| 12:41.51 | ManxPower-work | Morlac: Maybe you could ask on a Switchvox channel? |
| 12:42.08 | Morlac | that would do... #switchvox? |
| 12:42.18 | ManxPower-work | Morlac: no idea. Nobody here uses it. |
| 12:42.45 | Morlac | I see, ok, ill look around |
| 12:43.51 | Morlac | thansk |
| 12:45.34 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 12:49.42 | *** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 13:02.34 | *** join/#asterisk ariel_ (n=chatzill@63.214.236.169) |
| 13:02.49 | *** join/#asterisk beek (n=klinebl@pdpc/supporter/bronze/beek) |
| 13:08.33 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 13:09.25 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 13:10.27 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 13:11.27 | *** join/#asterisk garymc (n=garymc@81.138.225.161) |
| 13:13.14 | *** join/#asterisk [TK]D-Fender (n=chatzill@216.191.106.163) |
| 13:13.18 | *** join/#asterisk SolmyrBhaal (n=louisruc@196-209-115-118-tpr-esr-2.dynamic.isadsl.co.za) |
| 13:16.24 | devmod | anyone knows how can I record video prompts? or convert .avi to .h263/4 to be played on asterisk ? |
| 13:16.25 | *** join/#asterisk NET||abuse (n=l@91.123.232.146) |
| 13:20.24 | SolmyrBhaal | Hey All |
| 13:20.29 | *** join/#asterisk guyvdb_ (n=guy@dsl-240-132-98.telkomadsl.co.za) |
| 13:20.54 | SolmyrBhaal | I am trying to get into developing in Asterisk for an embedded platform. We have downloaded Astfin which works great however - for ever change I want to make to the Asterisk Source code we need to rebuild Linux and then copy the uImage on the board - this also doesn't allow us to debug the board. |
| 13:21.23 | SolmyrBhaal | We have plug-ins for the board (it is BlackFin 537) linked to Eclipse that we can build C and C++ projects to the board. Is there an easy way to link Asterisk to Eclispe so I can build and make any changes needed to the source? Or any other program that you can easily debug and make changes to asterisk (other than using test editior and the make files included?) |
| 13:21.49 | ManxPower-work | SolmyrBhaal: Try asking on #asterisk-dev |
| 13:22.04 | SolmyrBhaal | KK thanks, |
| 13:22.08 | guyvdb_ | Hi, I have a dial plan that has the following: exten => _XXXXXXXXXX,1,Dial(DAHDI/g1/${EXTEN}) now I want to add the ability to Dial(DAHDI/g2/${EXTEN}) if the dahdi g1 goup is congested. How would I go about that? |
| 13:22.08 | SolmyrBhaal | will give it a try... |
| 13:22.23 | *** join/#asterisk angryuser_ (n=angryuse@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 13:22.47 | guyvdb_ | would it be priority 101? |
| 13:23.00 | ManxPower-work | guyvdb_: see macro-stdexten in the extensions.conf.sample. Basically check the value of DIALSTATUS after the first Dial to determine if the 2nd Dial is needed. |
| 13:23.19 | ManxPower-work | guyvdb_: any docs that mention n+101 are many years out of date. |
| 13:24.08 | [TK]D-Fender | guyvdb_: Just dial them back to back |
| 13:24.43 | [TK]D-Fender | guyvdb_: Or if the dial line would look identical except for the group, jsut make a 2nd group |
| 13:24.44 | guyvdb_ | thx |
| 13:25.09 | *** join/#asterisk voipmonk (n=shido6@dsl-67-204-37-228.acanac.net) |
| 13:25.15 | [TK]D-Fender | guyvdb_: I suggest the lastter |
| 13:25.19 | [TK]D-Fender | latter* |
| 13:25.21 | [TK]D-Fender | gah... |
| 13:25.24 | guyvdb_ | I have 2 groups already.... but I want to dial in g1 first and only if congested do i want to dial in g2 |
| 13:25.33 | voipmonk | laster gives u the same idea |
| 13:25.35 | voipmonk | :) |
| 13:25.42 | voipmonk | chuckles |
| 13:26.04 | voipmonk | hides before getting kicked in the head :_ |
| 13:26.40 | *** join/#asterisk jkroon (n=jkroon@dsl-244-41-117.telkomadsl.co.za) |
| 13:26.45 | voipmonk | u could use dialstatus or just make dialing out g2 the next priority :) |
| 13:27.07 | ManxPower-work | guyvdb_: you have your answer |
| 13:27.08 | *** join/#asterisk garymc (n=garymc@81.138.225.161) |
| 13:27.26 | voipmonk | but... you would need to know what dialstatus is and priorities which means you would need to .... oh my god... READ!!! |
| 13:27.40 | voipmonk | im so sorry |
| 13:29.01 | guyvdb_ | ManxPower-work found the example... thx |
| 13:29.39 | *** join/#asterisk darkskiez_ (n=dz@62-50-207-67.client.stsn.net) |
| 13:30.26 | *** join/#asterisk jkroon (n=jkroon@uriel.interexcel.co.za) |
| 13:31.12 | jkroon | ok guys, how do i go about trouble-shooting call cuttoffs? |
| 13:31.18 | jkroon | on asterisk 1.6.1.12 now |
| 13:31.35 | jkroon | this didn't happen with 1.6.1.6, however, we need other fixes since then. |
| 13:32.03 | ManxPower-work | jkroon: make sure you have the latest DAHDI |
| 13:32.07 | _cgc | would anyone know why one side of a phone call over a sip trunk would not work, you can hear the person talking from one side but not from the other, I have forwarded all relevant ports 5060, 10000-20000 |
| 13:32.18 | *** join/#asterisk catojo (n=catojo@189.24.114.76) |
| 13:32.19 | jkroon | it's SIP channels cutting off, but I'll do the dahdi upgrade thing anyway. |
| 13:32.24 | ManxPower-work | _cgc: incorrect localnet or externip= |
| 13:32.32 | *** join/#asterisk viq (n=viq@unaffiliated/viq) |
| 13:32.33 | jkroon | SIP <-> SIP. |
| 13:32.43 | ManxPower-work | jkroon: Next time give more details so people are not forced to assume. |
| 13:32.47 | _cgc | ManxPower-work: nah that is correct |
| 13:33.07 | jkroon | ManxPower-work, thanks :). i'm not even sure what's all relavent in this case. |
| 13:33.09 | _cgc | ManxPower-work: http://pastebin.ca/1746758 <------ call with sip debug logs |
| 13:33.10 | ManxPower-work | _cgc: then the other other thing could be the far end NAT'd phone. Make sure the PHONE has no NAT stuff enabled. |
| 13:33.31 | jkroon | i actually think it's something to do with app_queue ... only client we have this problem, also the only client where app_queue is in use. |
| 13:33.52 | jkroon | but now they said the same applies on outbound calls, which does not go via app_queue. |
| 13:34.26 | ManxPower-work | _cgc: good thing you spend the time removing all the important information from that pastebin |
| 13:34.48 | _cgc | ManxPower-work: lol, made that mistake before :/ |
| 13:35.24 | _cgc | its easy with vim and %s/ |
| 13:35.48 | ManxPower-work | _cgc: Do you have canreinvite=no in sip.conf? |
| 13:36.06 | ManxPower-work | jkroon: I wish you the BEST of luck. |
| 13:36.18 | _cgc | ManxPower-work: yes |
| 13:36.39 | jkroon | thanks ManxPower-work - any ideas where i can start looking perhaps? |
| 13:37.10 | ManxPower-work | jkroon: no idea where you would learn what is important to mention and what is not important |
| 13:38.59 | jkroon | in this case I really don't know what's important and what not, what I know is that the call comes in from a CISCO PRI->SIP GW, goes into app_queue, gets picked up by a SIP/ member, and gets cut off 30 seconds later. |
| 13:40.00 | ManxPower-work | set canreinvite=no in sip.conf |
| 13:40.20 | ManxPower-work | jkroon: every statement you make includes some ciritical piece of info you left out. |
| 13:40.49 | jkroon | would be useful for me for future if you can point out for me what you consider critical then I can focus around that. |
| 13:40.54 | _cgc | ManxPower-work: canreinvite=no is already set in sip.conf |
| 13:41.29 | jkroon | i've also got canreinvite=no in my sip.conf - I do this by default for other reasons (more accurate CDRs and recording purposes) |
| 13:43.55 | NET||abuse | I swear, arrg, i can't get the demo-congrats to not be the result when you call our phone number? have a sip trunk from our voip provider, we get a local phone number to our area, so the asterisk box picks up incoming calls, problem is i can't get incoming calls to ring our ringgroup? i just made a "main" ring group and stuck the 3 extensions for our softphones in. incoming call rule pattern _X. to our call group,, apply,, nothing happening. |
| 13:44.02 | *** join/#asterisk hohum (i=dcorbe@apollo.corbe.net) |
| 13:44.03 | NET||abuse | i just get "congratulations on installing.... blah blah" |
| 13:44.24 | ManxPower-work | NET||abuse: Don't forget to remind everyone you are using AsteriskNOW |
| 13:44.45 | ManxPower-work | And the fact you are too stubborn to ask on the correct channel. |
| 13:45.37 | NET||abuse | I always ask on the correct channel, no-one ever responds. |
| 13:45.57 | ManxPower-work | NET||abuse: Maybe you should rethink using a product that is not supported. |
| 13:46.02 | ManxPower-work | Asking here does no good either. |
| 13:46.07 | [TK]D-Fender | NET||abuse: that doesn't become our problem here... |
| 13:46.10 | NET||abuse | ManxPower-work, fair nuff :( |
| 13:46.19 | NET||abuse | I'm chancing my arm and I know it :) |
| 13:46.35 | NET||abuse | just hard when i seem to be getting no-where for days on end. |
| 13:46.35 | [TK]D-Fender | jkroon: Where is the complete call with SIP debug enabled for us to look at? |
| 13:46.37 | ManxPower-work | [TK]D-Fender: I guess it's too much to hope that NET||abuse gets banned? |
| 13:46.40 | NET||abuse | a bit frustrating :P |
| 13:46.54 | voipmonk | back again , NET||abuse |
| 13:46.59 | voipmonk | ? |
| 13:47.04 | NET||abuse | voipmonk, yeh, just nothing wants to work for me |
| 13:47.14 | jkroon | [TK]D-Fender, trying to get one but I'm not even seeing any indication in asterisk CLI that calls are being dropped. |
| 13:47.17 | voipmonk | well thats sort of not how it works anyway :) |
| 13:47.22 | NET||abuse | i can't get phone calls out, i can get the ring group working, the only thing that works is calls between extensions. |
| 13:47.54 | ManxPower-work | So you come here and waste our time? |
| 13:48.06 | NET||abuse | just hitting a wall ever time i go at it. and it's frustrating me :( |
| 13:48.13 | NET||abuse | ManxPower-work, it's your choice to respond or not. |
| 13:48.20 | ManxPower-work | NET||abuse: How, exactly, is that our problem? |
| 13:48.30 | [TK]D-Fender | NET||abuse: sonds like your install is fine and you simply ahve no clue on USING FreePBX... go take it up in there then |
| 13:48.32 | [TK]D-Fender | ~freepbx |
| 13:48.33 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 13:48.48 | ManxPower-work | [TK]D-Fender: he's been told many, many times. He refuses to accept it. |
| 13:49.07 | voipmonk | well thats a start - there are these things online.... they're called "tutorials", holy cow man if you saw one, I think you would just die... its like reading a book... but not... it has pictures... and oh my buddha... if you saw the ones with motion picture... on youboob.. I know you wouldnt be able to handle it :) |
| 13:49.22 | [TK]D-Fender | ManxPower-work: Last referral was to #asterisknow . THAT is of no use. #freepbx actually has people who help |
| 13:49.35 | ManxPower-work | [TK]D-Fender: AsteriskNOW uses FreePBX? |
| 13:49.38 | *** join/#asterisk coppice (n=chatzill@202.64.176.25) |
| 13:50.03 | [TK]D-Fender | ManxPower-work: ..... what rock have you been under? Did it hurt when it landed? :p |
| 13:50.12 | ManxPower-work | I thought it used whatever Digium GUI was. Maybe I was thinking of AsteriskGUI |
| 13:50.15 | [TK]D-Fender | ManxPower-work: I've even told you directly myself long ago |
| 13:50.32 | ManxPower-work | ~asterisknow |
| 13:50.33 | infobot | rumour has it, asterisknow is based on Asterisk, but is difficult to support in #asterisk for a number of reasons. Please seek support in #asterisknow instead. |
| 13:50.43 | ManxPower-work | maybe someone should update that channel name? |
| 13:50.59 | [TK]D-Fender | ManxPower-work: Whats wrong with the channel name? |
| 13:51.14 | ManxPower-work | [TK]D-Fender: you just said he should be asking on #FreePBX. |
| 13:51.45 | [TK]D-Fender | ManxPower-work: He isn't having a DISTRO PROBLEM, he's having a "OMG I'm incompetant with the GUI it comes BUNDLED with" problem |
| 13:52.14 | ManxPower-work | [TK]D-Fender: His problem is he's an idiot. |
| 13:52.15 | [TK]D-Fender | ManxPower-work: So let the GUI people help him |
| 13:52.26 | ManxPower-work | [TK]D-Fender: Been trying. He refuses to leave. |
| 13:52.37 | *** join/#asterisk muiro (n=muiro@unaffiliated/muiro) |
| 13:55.23 | NET||abuse | ManxPower-work, Well apologies for my previous actions, I will cease asking asterisk-gui stuff in here. |
| 13:55.46 | [TK]D-Fender | NET||abuse: It isn't Asterisk GUI stuff. Its ***FREEPBX*** stuff |
| 13:56.02 | [TK]D-Fender | NET||abuse: So go to #freepbx , not #asteriskgui or #asterisknow |
| 13:56.04 | gr0mit | NET||abuse, you are much better off using raw asterisk |
| 13:59.49 | _cgc | ManxPower-work: no idea's then? or do you need more info? |
| 14:00.01 | *** join/#asterisk shamelessn00b (n=chatzill@58-65-172-114.nayatel.pk) |
| 14:00.34 | jkroon | does anybody know what codec 126 is about? seems x-lite sends it for MOH or something? |
| 14:02.13 | *** join/#asterisk etfonhomey (n=etfonhom@ip-64-32-192-35.iad.megapath.net) |
| 14:04.02 | *** join/#asterisk af_ (n=getsmart@88-149-241-228.dynamic.ngi.it) |
| 14:04.22 | ManxPower-work | jkroon: "core show codecs" |
| 14:04.52 | ManxPower-work | _cgc: ask [TK]D-Fender |
| 14:04.57 | *** join/#asterisk mnt_real (n=sinan@bas12-montrealak-1167974851.dsl.bell.ca) |
| 14:05.45 | jkroon | ManxPower-work, that's weird. it's basically all the codecs that x-lite support, but I only receive it when a call goes on hold. |
| 14:06.15 | *** join/#asterisk italorossi (n=kvirc@189.23.15.3) |
| 14:06.43 | jkroon | could this be the result of a call cut? utils.c:1126 ast_carefulwrite: write() returned error: Broken pipe |
| 14:06.57 | jkroon | and if this is the case, can I guess (wager) network error? |
| 14:07.11 | tzafrir | jkroon, 126 = 127 - 1 = 2 + 4 +8 + 16 + 32 + 64 |
| 14:07.36 | _cgc | [TK]D-Fender: When I call out over a sip trunk, 1 side of the call has no sound, http://pastebin.ca/1746758 <--- call with sip debug on |
| 14:07.38 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:08.05 | ManxPower-work | jkroon: no, that shows what codecs ASTERISK supports |
| 14:08.17 | _cgc | [TK]D-Fender: any idea's? |
| 14:08.20 | Kobaz | aaaaxxeterisk |
| 14:08.21 | jkroon | i know, but i'm trying to locate the warning I receive. |
| 14:09.09 | voipmonk | love noisy feedback |
| 14:09.22 | jkroon | rtp.c:1796 ast_rtp_read: Unknown RTP codec 126 received from '192.9.200.179'. |
| 14:09.29 | voipmonk | and a bunch of nat retries |
| 14:09.29 | _cgc | [TK]D-Fender: If its me calling out over the trunk, I cannot hear them, but they can hear me fine |
| 14:09.33 | jkroon | thinks he should go find a good SIP guide and learn the protocol. |
| 14:09.48 | voipmonk | call goes out |
| 14:10.17 | _cgc | voipmonk: are you talking to me? |
| 14:10.19 | voipmonk | then someone dialed 6101 which doesnt exist as an extension |
| 14:10.43 | [TK]D-Fender | _cgc: Retransmitting #6 (no NAT) to 217.14.132.183:5060: <- clearly not good. Next you are masking IP's, and I am not seeing your CONFIGS. |
| 14:11.04 | [TK]D-Fender | jkroon: https://issues.asterisk.org/view.php?id=15157 |
| 14:12.25 | *** join/#asterisk stmaher (n=stephen@80.68.89.200) |
| 14:12.27 | stmaher | Hi guys.. |
| 14:12.31 | Kobaz | hi |
| 14:12.44 | Kobaz | thank you, come again |
| 14:12.55 | stmaher | Could someone please point me to an example code of a menu where if the call loops through it twice it autodisconnects the call |
| 14:13.07 | stmaher | its to stop calls taking up lines |
| 14:14.24 | jkroon | [TK]D-Fender, thanks. that clears things up. Especially the whole NAT keep-alive thing and it makes sense. |
| 14:14.29 | Kobaz | in extensions.conf or in ael |
| 14:14.38 | stmaher | extensions.conf |
| 14:14.41 | Kobaz | aw |
| 14:14.42 | jkroon | it shouldn't (cannot, I think) cause what I'm seeing in terms of the call cut-offs. |
| 14:15.18 | [TK]D-Fender | stmaher: Set a counter to 0 at the start of your menu. On t/i, increment the counter and then check if its too big. If not jump up and repeat the menu |
| 14:15.32 | ManxPower-work | [TK]D-Fender: Today is Mondat^noob |
| 14:15.38 | ManxPower-work | and Monday^noob |
| 14:15.47 | Kobaz | heh |
| 14:15.55 | stmaher | [TK]D-Fender thanks.. looking for some syntax for example |
| 14:16.14 | [TK]D-Fender | stmaher: core show application gotoif |
| 14:16.20 | [TK]D-Fender | stmaher: core show application set |
| 14:16.27 | stmaher | thanks |
| 14:16.34 | Kobaz | 1,set(attempts=2) n,gotoif($[${attempts} <= 0],done) n,Set(attempts=$[$attempts - 1]) n,dosomething done,hangup() |
| 14:16.37 | [TK]D-Fender | stmaher: And go read the CHANNELVARIABLES doc |
| 14:16.38 | Kobaz | something like that |
| 14:16.50 | Kobaz | except the gotoif syntax might be wrong |
| 14:17.00 | Kobaz | i haven't done raw extensions.conf in a while |
| 14:17.25 | stmaher | Thank you all |
| 14:19.01 | Kobaz | in ael, it's much more straightforward |
| 14:19.05 | Kobaz | attempts=2; while (${attempts} > 0) { attempts = ${attempts} - 1; dostuff; } hangup(); |
| 14:19.10 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 14:19.40 | [TK]D-Fender | Kobaz: virtually the same |
| 14:20.17 | stmaher | ael does look very c(ish) |
| 14:20.24 | _cgc | [TK]D-Fender: http://pastebin.ca/1746904 <--- some of my sip.conf and extensions.conf, and I am only masking my IP, username, password and mobile number as I am not posting them on the internet |
| 14:20.28 | stmaher | might take that approach when we upgrade :-) |
| 14:20.28 | stmaher | thanks |
| 14:20.36 | [TK]D-Fender | stmaher: Wouldn't advise |
| 14:20.45 | stmaher | whys that? |
| 14:21.16 | Kobaz | [TK]D-Fender: virtually, but much easier to maintain, no line numbers |
| 14:21.33 | Kobaz | stmaher: tk fender is against the march of progress :P |
| 14:21.49 | stmaher | LOL |
| 14:21.53 | stmaher | I dont want to start a channel war.. |
| 14:22.01 | stmaher | but it does look handy |
| 14:22.12 | Kobaz | stmaher: ael used to have all kinds of translation problems, so, previously it was pretty much an experimental extension |
| 14:22.15 | stmaher | it took me a while to get used to the dialplan of extensions.conf |
| 14:22.18 | [TK]D-Fender | Kobaz: AEL isn't progress. Its a trek SIDEWAYS which can only lead to more problems and limitations |
| 14:22.42 | Kobaz | me personally... i haven't run into any translation bugs in like, more than a year |
| 14:22.45 | _cgc | [TK]D-Fender: all the macro trunkdial does is play a sound and dial the arg1 |
| 14:22.52 | ManxPower-work | AEL is a poor way to learn Asterisk. |
| 14:22.54 | Kobaz | and i've been able to do anything that i have wanted to do in ael without problem |
| 14:22.55 | [TK]D-Fender | _cgc: What do you have forwarded to *? |
| 14:23.09 | Kobaz | ManxPower-work: i think it greatly speeds up learning |
| 14:23.30 | ManxPower-work | Kobaz: I disagree. There is so little docs and examples of AEL it makes it hard for noobs. |
| 14:23.32 | Kobaz | ManxPower-work: my coworker picked up dialplan coding in a day with ael... he was like wtf? when i showed him extensions.conf |
| 14:23.35 | ManxPower-work | I love AEL, BTW. |
| 14:23.37 | drmessano | So does users.conf :/ |
| 14:23.58 | jkroon | hmm, this is messed up. these messages (which I previously thought was funny): format_wav.c: Unable to set write file size co-incide EXACTLY with the end-time of the cut call. |
| 14:24.10 | Kobaz | ManxPower-work: there's examples for all the major constructs of ael/ael2 on the voip wiki |
| 14:24.16 | ManxPower-work | The other issue is that if you don't understand extensions.conf you won't understand WHY some times are the way they are in AEL. |
| 14:24.24 | [TK]D-Fender | [09:23]<Kobaz>ManxPower-work: i think it greatly speeds up learning <- considering iffy docs, and a marginal user base.... I tend to differ |
| 14:24.55 | *** join/#asterisk Akiraa (n=Akiraaaa@79.112.11.200) |
| 14:25.04 | Kobaz | extensions.conf == BASIC... which, I don't know about you, but people moved away from BASIC style programming a long time ago (except for legacy apps) |
| 14:25.23 | *** join/#asterisk ChkDigit (n=mike@static24-72-71-175.regina.accesscomm.ca) |
| 14:25.33 | _cgc | [TK]D-Fender: i'm dialing out from * over the sip trunk to a sip provider |
| 14:25.43 | Kobaz | structured programming has been around for ever, which is why I think ael speeds up learning, because it's generally the same as all other modern languages in principal |
| 14:26.01 | ManxPower-work | Kobaz: you are learning AEL, not Asterisk. |
| 14:26.03 | [TK]D-Fender | Kobaz: And AEL is a BASIC COMPILER. Congratulations on being more limited to something you haven't bothered learning yet.. and hope you don't run into problems |
| 14:26.09 | drmessano | There's like 5 people using AEL.. 2 of them are here, the other 3 are trying to figure out how to make that thing happen again that happened when they dropped those two big rocks together that they were carrying |
| 14:26.36 | stmaher | Im sorry i brought this up :-) |
| 14:26.46 | Kobaz | ManxPower-work: i've never run into problems where something worked the way it shouldn't have because it was in ael, or because asterisk translated it wrong |
| 14:26.56 | Kobaz | ManxPower-work: not saying that someone wouldn't, but from my experience it hasn't happened |
| 14:27.10 | ManxPower-work | Kobaz: testvar=*; |
| 14:27.22 | Kobaz | use Set() instead |
| 14:27.38 | Kobaz | you should only use a=b for straight numeric assignments |
| 14:27.46 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 14:27.51 | ManxPower-work | Kobaz: Of course, but if you don't understand how extensions.conf works you would not understand why you would need Set |
| 14:27.59 | Kobaz | true |
| 14:28.07 | jkroon | hmm yes, the cut offs seems to co-incide with those unable to set write file size messages. |
| 14:28.33 | ManxPower-work | You also need to understand extensions.conf in order to debug your AEL script, since the AEL script is translated INTO extensions.conf format at load time. |
| 14:29.05 | Kobaz | not necessarily, it's pretty easy to throw in noop's and see where code is, without needing line number mental mapping |
| 14:29.25 | Kobaz | okay, i agree it's useful to also know extensions.conf |
| 14:29.31 | Kobaz | i never said it wasn't |
| 14:29.32 | ManxPower-work | Personally, I use AEL for virtually everything I write. I love AEL. I am not under the wishful thinking cloud thinking it's easier for n00bs. |
| 14:29.39 | Kobaz | heh |
| 14:29.41 | Kobaz | k |
| 14:30.07 | Kobaz | anyways, i prefer AGI anyway :) |
| 14:30.27 | drmessano | That's redundant and now you're just trying to be elitist |
| 14:30.39 | Kobaz | redundant? |
| 14:30.49 | ManxPower-work | I usually use a mix of extensions.conf, extensions.ael and AGIs. |
| 14:30.50 | drmessano | anyways, I prefer AGI anyway |
| 14:31.07 | Kobaz | oh |
| 14:31.08 | Kobaz | my phrase |
| 14:31.13 | Kobaz | i thought the context was redundant |
| 14:31.32 | drmessano | Maybe you should try english before AEL.. Just sayin |
| 14:31.35 | Kobaz | heh |
| 14:31.36 | *** join/#asterisk KavanS (n=KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 14:31.42 | jkroon | i wonder ... btw, if anybody asks, CONNTRACK_SIP in the kernel is broken. |
| 14:32.00 | drmessano | CONNTRACK_SIP is tthe wrong way to go |
| 14:32.42 | voipmonk | oh wow |
| 14:33.20 | voipmonk | i must have been under a rock - there is a such thing? |
| 14:33.21 | [TK]D-Fender | jkroon: Would you like a new shovel? Something wider perhaps? |
| 14:33.45 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 14:34.10 | jkroon | [TK]D-Fender, actually yes :) but no, the CONNTRACK issue is something else I picked up whilst trying to troble-shoot something else, just noticed it's active on this box. |
| 14:34.19 | drmessano | I actually avoid AEL and extensions.conf altogether.. I have some AGIs that run in .NET and I basically run Asterisk from a Vista box via Gig-E crossover like a brick on the gas pedal and rope on the steering wheel |
| 14:34.22 | jkroon | wondering whether it might be what's causing my issue now. |
| 14:34.58 | Kobaz | wow |
| 14:34.58 | Kobaz | aughh |
| 14:35.01 | Kobaz | this is really bad |
| 14:35.20 | drmessano | This works rather well except needing to take down the system every 30 days for Windows updates |
| 14:35.45 | Kobaz | i had asterisk running on a box with a t1. it would accept the call, and play tracks... everything looks perfect on the console and logs... yet no audio was passing through |
| 14:35.52 | Kobaz | i restarted asterisk and now it works fine |
| 14:36.01 | drmessano | Kobaz: Sounds like an AEL problem |
| 14:36.04 | Kobaz | haha |
| 14:36.24 | Kobaz | <PROTECTED> |
| 14:36.24 | Kobaz | <PROTECTED> |
| 14:36.28 | Kobaz | this is agi |
| 14:36.41 | Kobaz | no audio whatsoever when it was playing that |
| 14:37.14 | Kobaz | no idea... i can't even debug it since there's no other information |
| 14:37.27 | drmessano | Maybe you need to upgrade your PPPROSS stack |
| 14:38.13 | Kobaz | the what what? |
| 14:38.17 | drmessano | PHP, Python, Perl, Ruby Or Some Shit <-- The complete "How to make life complicated as hell" suite |
| 14:38.41 | drmessano | Thats the answer to anything.. |
| 14:39.01 | [TK]D-Fender | Kobaz: Nothing to debug there |
| 14:39.04 | drmessano | "This site is running like crap, wonder what it is written in..." |
| 14:39.07 | Kobaz | [TK]D-Fender: i know |
| 14:39.14 | drmessano | "PPPROSS" |
| 14:39.23 | drmessano | "Ah.." |
| 14:40.16 | *** join/#asterisk _zen_ (n=_zen_@cpe-74-66-140-78.nyc.res.rr.com) |
| 14:40.52 | _cgc | [TK]D-Fender: if it helps, when i phone over the trunk, I cannot even hear the phone ring |
| 14:41.06 | Kobaz | _cgc: huh? |
| 14:41.28 | *** join/#asterisk |Rain| (i=rain@ev.il.net) |
| 14:41.33 | jkroon | ok, issue is unrelated, the warning comes due to update_header() in format_wav.c, not causing it. |
| 14:41.55 | ManxPower-work | Back in MY day we leaned Asterisk for several months before having the hubris to try installing it on a production system. Too bad the young whippersnappers today don't do that. |
| 14:42.09 | [TK]D-Fender | _cgc: Could be they don't send progress, and I don't see an answer to my previous question. |
| 14:42.13 | Kobaz | ManxPower-work: well that's always a good idea |
| 14:42.29 | |Rain| | anyone have any debugging tips for asterisk 1.4.x spewing 'channel.c: Exceptionally long voice queue length queuing to IAX2/<blah>' for pretty much all active IAX2 calls? |
| 14:42.31 | Kobaz | ManxPower-work: really it takes about a year or more to really get into asterisk |
| 14:42.50 | Kobaz | |Rain|: deadlocks would cause that |
| 14:43.03 | Kobaz | i'm sure there's other possible causes too |
| 14:43.06 | _cgc | [TK]D-Fender: what to you mean forwarded to *?, I'm dial out |
| 14:43.28 | |Rain| | I suspect it's a deadlock, but it's difficult to reproduce and hard to debug a live system |
| 14:44.07 | Kobaz | |Rain|: welcome to the club |
| 14:44.11 | |Rain| | I have segfaults relating to frame corruption that I've been trying to debug, too :/ |
| 14:44.28 | |Rain| | getting close to downgrading |
| 14:44.34 | Kobaz | what version? |
| 14:44.35 | ManxPower-work | |Rain|: make sure you are running the latest 1.4.x |
| 14:44.45 | |Rain| | I'm using 1.4.28 |
| 14:44.53 | Kobaz | ManxPower-work: that's not always the best idea |
| 14:44.54 | ManxPower-work | |Rain|: is that the latest? |
| 14:45.14 | drmessano | Avoiding 1.4 is a good idea :P |
| 14:45.15 | |Rain| | it is |
| 14:45.15 | ManxPower-work | Kobaz: I agree with regards to 1.6.1/1.6.2 and maybe even 1.6.0, but not 1.4 |
| 14:45.33 | Kobaz | ManxPower-work: i've had to downgrade numerous times in the 1.4 tree because stuff just kept crashing |
| 14:45.37 | Kobaz | in 1.6 too |
| 14:45.37 | |Rain| | drmessano: yeah, I have a non-production box testing 1.6.2.0 |
| 14:45.49 | *** join/#asterisk jtexter3 (n=jtexter3@72.242.229.213) |
| 14:46.00 | *** join/#asterisk Naikrovek (n=jjohnson@unaffiliated/naikrovek) |
| 14:46.25 | ManxPower-work | Kobaz: That's just typical Asterisk releases. |
| 14:46.26 | jtexter3 | Is it possible to whisper using app_meetme or app_confbridge? |
| 14:46.50 | ManxPower-work | jtexter3: does "core show application meetme" or "core show application confbridge" tell you that you can? |
| 14:47.06 | ManxPower-work | is app_confbridge new to 1.6.x? |
| 14:47.10 | Kobaz | yes |
| 14:47.41 | Kobaz | it uses native bridging instead of dahdi/zapotel |
| 14:47.56 | ManxPower-work | But what does it use for audio mixing? |
| 14:48.03 | Kobaz | that i don't know |
| 14:48.16 | drmessano | ManxPower-work: It handles that |
| 14:48.21 | drmessano | that was the idea |
| 14:49.15 | ManxPower-work | drmessano: I asked because app_meetme does not use Zaptel/DAHDI to bridge the channels. |
| 14:49.25 | ManxPower-work | It uses it for audio mixing at least. |
| 14:49.34 | *** join/#asterisk lordmortis (n=lordmort@203-206-67-161.dyn.iinet.net.au) |
| 14:49.34 | drmessano | Correct |
| 14:49.37 | drmessano | Thats all it uses it for |
| 14:49.47 | drmessano | app_confbridge is all your call logic and your mixing |
| 14:49.52 | ManxPower-work | So I was trying to get Kobaz to actually make a statement that is true. |
| 14:50.11 | drmessano | heh |
| 14:50.21 | Kobaz | ? |
| 14:50.35 | ManxPower-work | (9:47:41 AM) Kobaz: it uses native bridging instead of dahdi/zapotel <--- WRONG WRONG WRONG |
| 14:51.07 | Kobaz | oh, well... that's what i remembered reading |
| 14:51.25 | Kobaz | of all the things i've lost |
| 14:51.27 | drmessano | I never seen that stated |
| 14:51.30 | Kobaz | i miss my mind the most |
| 14:51.59 | Kobaz | drmessano: no, i remembered wrong |
| 14:52.00 | drmessano | Kobaz: Yes, and we can't debug because you're using AEL |
| 14:52.08 | drmessano | Kobaz: +1 |
| 14:52.16 | Kobaz | agi |
| 14:52.34 | drmessano | PPPROSS, whatever |
| 14:52.52 | Kobaz | what are you trying to debug? |
| 14:52.59 | drmessano | Your statement |
| 14:53.04 | Kobaz | which one |
| 14:53.15 | drmessano | You've forgotten already? |
| 14:53.25 | Kobaz | my t1 problem? |
| 14:53.26 | drmessano | The one where you said you like eating bananas naked in the park |
| 14:53.35 | drmessano | You dont remember that? |
| 14:53.38 | Kobaz | heh |
| 14:53.54 | *** join/#asterisk Gugge (n=gugge@vlan2.dlxhosting.dk) |
| 14:54.14 | *** join/#asterisk jo8330 (i=d04149c9@gateway/web/freenode/x-nuungykwjtkldyus) |
| 14:55.06 | jo8330 | Morning all. I have a 1-800 number setup but I'm finding that Asterisk does not get caller ID information. Is that usual that telco doesn't forward caller ID information for calls received through a 1-800 number? |
| 14:55.12 | jo8330 | just wondering what other people's experiences have been |
| 14:55.13 | drmessano | So who is gonna write the voicemail notification extension for google chrome? |
| 14:55.23 | Kobaz | jo8330: depends... ask your provider |
| 14:57.02 | [TK]D-Fender | jo8330: Where do we see that they're not delivering CID? Where are you even telling us what your calls are being deliverd OVER? |
| 14:57.32 | *** join/#asterisk lordmortis (n=lordmort@203-206-67-161.dyn.iinet.net.au) |
| 14:59.14 | jo8330 | bell T1 PRI |
| 15:00.28 | *** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 15:00.50 | [TK]D-Fender | jo8330: and where is the call to debug? |
| 15:05.09 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 15:05.09 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:06.10 | *** join/#asterisk rossand (n=aross@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 15:06.23 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 15:09.35 | _cgc | [TK]D-Fender: http://pastebin.ca/1746968 <--- more debug info, hope this helps a bit more |
| 15:11.22 | *** join/#asterisk Defraz (n=Defraz@corp.fuzecore.com) |
| 15:11.30 | *** part/#asterisk ManxPower-work (n=EWieling@216.186.151.147) |
| 15:11.30 | *** join/#asterisk neurosys (n=neurosys@c-71-196-20-208.hsd1.fl.comcast.net) |
| 15:11.37 | *** join/#asterisk catojo (n=catojo@189.24.114.76) |
| 15:12.42 | jo8330 | [TK]D-Fender: i don't see much to debug, telco doesn't give me callerid digits. i'll have to wait to hear back |
| 15:12.47 | _cgc | would line 448 be my problem saying 'Our prefcodec: 0x0 (nothing)'? |
| 15:13.14 | [TK]D-Fender | jo8330: Where do I see you enabling any debug? Or showing me any of a call at all? |
| 15:13.42 | jo8330 | ok ill show you, but i can tell there's nothing useful. |
| 15:13.48 | *** join/#asterisk smooth_penguin (n=smoove@59.95.33.64) |
| 15:14.21 | _cgc | jo8330: if your asking for their help shouldn't you be letting them decide whats useful? |
| 15:14.37 | [TK]D-Fender | jo8330: And you HAVE to be able to get CID on an 800#. You're PAYING for the call, they can't be masking it. |
| 15:14.49 | jo8330 | no i was just asking what people's experience has been. |
| 15:15.29 | [TK]D-Fender | jo8330: My experience has been that people come in here not knowing what they are doing and having preconceived ideas that there isn't more that can be seem all the while not showing us whats actually happening. |
| 15:15.48 | [TK]D-Fender | jo8330: So go show me a call with complete debug |
| 15:15.53 | jo8330 | http://pastebin.ca/1746979 |
| 15:17.07 | [TK]D-Fender | jo8330: that is nothing... where the hell is the PRI DEBUG? |
| 15:17.26 | [TK]D-Fender | jo8330: And you're masking #'s, and I don't see your dialplan or configs. |
| 15:17.31 | *** part/#asterisk shafu (n=giany@83.169.0.238) |
| 15:17.53 | jo8330 | ok gimme a min |
| 15:22.13 | voipmonk | line 11 looks good |
| 15:22.26 | voipmonk | line 31 is what you're talking about, yes? |
| 15:24.13 | Kobaz | ack |
| 15:24.19 | Kobaz | sangoma's website is deaded |
| 15:24.38 | jo8330 | http://pastebin.ca/1746979 |
| 15:24.57 | jo8330 | dialplan is simple, just AGI call to a python script that plays a welcome audio file |
| 15:25.30 | [TK]D-Fender | jo8330: Where is the NEW pastebin? |
| 15:25.46 | jo8330 | that's it, i replaced the old one |
| 15:25.57 | [TK]D-Fender | jo8330: NO |
| 15:26.02 | [TK]D-Fender | jo8330: it gets a new # |
| 15:26.08 | coppice | Kobaz: quick. start some rumours, and profit from their stock price |
| 15:26.13 | [TK]D-Fender | facepalms |
| 15:26.46 | Kobaz | coppice: heh |
| 15:26.55 | Kobaz | is sangoma public? |
| 15:26.55 | jo8330 | lol. ok http://pastebin.ca/1747005 |
| 15:27.20 | Kobaz | it seems so |
| 15:27.30 | *** join/#asterisk kannan (n=kannan@58.68.68.26) |
| 15:27.42 | Kobaz | http://www.google.com/finance?q=CVE%3ASTC |
| 15:27.56 | *** join/#asterisk e4 (n=e4@rrcs-76-79-59-194.west.biz.rr.com) |
| 15:28.29 | *** join/#asterisk fenrus_ (i=fenrus@oklart.com) |
| 15:28.44 | *** join/#asterisk Skeeter- (i=Skeeter@c216.218.2-65.clta.globetrotter.net) |
| 15:29.11 | Skeeter- | anyone got some good doc. for mysql along with asterisk?? |
| 15:29.16 | [TK]D-Fender | jo8330: is this a call where you dialed the 800# yourself? |
| 15:29.28 | [TK]D-Fender | Skeeter-: in the tarball |
| 15:29.32 | [TK]D-Fender | Skeeter-: and the BOOK |
| 15:29.33 | jo8330 | yes |
| 15:29.34 | [TK]D-Fender | ~book |
| 15:29.35 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 15:29.47 | Skeeter- | thanks |
| 15:30.12 | jo8330 | [TK]D-Fender: it's a seperate phone, not a line on that T1. If I make the same call to the equivalent number, then I get CID info as expected |
| 15:30.14 | *** join/#asterisk moy (n=moy@bas1-unionville55-1177733883.dsl.bell.ca) |
| 15:30.58 | [TK]D-Fender | jo8330: Is your 800# initially landing at your telco, or via 3rd party? |
| 15:31.05 | jo8330 | telco |
| 15:31.18 | [TK]D-Fender | jo8330: PB your configs |
| 15:33.26 | *** join/#asterisk kaldemar (n=kaldemar@unaffiliated/kaldemar) |
| 15:34.06 | jo8330 | ok, it's pretty much defaults other than the T1 config lines. sec |
| 15:34.09 | *** join/#asterisk benngard (n=benngard@90-230-92-67-no148.tbcn.telia.com) |
| 15:35.35 | *** join/#asterisk bpgoldsb (n=bpgoldsb@ip24-250-198-162.ga.at.cox.net) |
| 15:35.51 | [TK]D-Fender | jo8330: Less talk, more show... |
| 15:36.54 | bpgoldsb | `asterisk -rvv` fails with 'Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?)'. It does exist, and the asterisk user can read/write to it. Any ideas why it won't connect? |
| 15:37.04 | jo8330 | lol |
| 15:37.15 | jo8330 | i'd hug you if i could, you're funny. |
| 15:37.17 | jo8330 | http://pastebin.ca/1747017 |
| 15:37.43 | jo8330 | everything else is default OOB asterisk 1.6.2.0 |
| 15:37.45 | Kobaz | bpgoldsb: ps ax | grep asterisk (is asterisk running? |
| 15:37.46 | jo8330 | digium TE122 |
| 15:38.24 | beek | This is lovely... one way audio on a PRI connection. |
| 15:38.28 | bpgoldsb | Kobaz: Yep I'm in the process of checking out the arguements the Debian init script is passing to make sure that isn't the issue |
| 15:38.46 | beek | telco <-PRI-> Asterisk <-PRI-> Iwatsu Legacy system. |
| 15:38.50 | Kobaz | bpgoldsb: is it running?.. paste ps output |
| 15:39.10 | bpgoldsb | Kobaz: root 28483 0.0 0.1 6616 1872 ? D 19:59 0:00 /usr/sbin/asterisk -f -F -g -vvv -p -U asterisk -vvvg -c |
| 15:39.21 | Kobaz | okay so it's running as user asterisk |
| 15:39.30 | Kobaz | ls -al /var/run/asterisk/asterisk.ctl |
| 15:39.34 | beek | or telco <-PRI-> Asterisk <-PRI-> telco <-> Answering service. They hear us, we don't hear them. |
| 15:39.42 | bpgoldsb | Kobaz: srwxr-xr-x 1 asterisk asterisk 0 2010-01-11 10:36 /var/run/asterisk/asterisk.ctl |
| 15:39.54 | Kobaz | okay, and what user are you running asterisk -rx as? |
| 15:39.59 | bpgoldsb | root. |
| 15:40.06 | Kobaz | interesting |
| 15:40.10 | bpgoldsb | I agree :) |
| 15:41.16 | Kobaz | why are you using -f and -F |
| 15:41.27 | Kobaz | they are opposites of each other |
| 15:42.41 | bpgoldsb | I actually turned that off, I was wondering the same thing |
| 15:43.09 | bpgoldsb | But oddly enough, after turning that off, Asterisk failed to launch from safe_asterisk |
| 15:43.23 | bpgoldsb | I'll muck around some more on my own for now. It's probably something on my end. |
| 15:43.24 | soman | soman |
| 15:44.09 | Kobaz | bpgoldsb: oh, and don't use -p either.. it's a recepie for problems |
| 15:44.57 | Kobaz | bpgoldsb: safe_asterisk needs no-forking... since it runs it as a foreground process and will restart it if it dies |
| 15:45.16 | Kobaz | bpgoldsb: so use either -f or -F, but not both... safe_asterisk is gonna need -f |
| 15:45.20 | bpgoldsb | Kobaz: I was suspicious of them. I'm using the debian default arguements. Whereas, in the past, I've always built from source and used the Digium defaults |
| 15:45.34 | Kobaz | yeah, i always build from source |
| 15:46.35 | *** join/#asterisk |Cybex| (n=John@atwork-21.r-212.178.82.atwork.nl) |
| 15:47.55 | Kobaz | bpgoldsb: -p has a tendency to make asterisk suck up 100% cpu randomly |
| 15:48.15 | bpgoldsb | So where does safe_asterisk build it's list of arguements from? |
| 15:48.23 | [TK]D-Fender | jo8330: Ok, I'd try setting "immediate=no" explicitly, and "callerid=asreceived" (though the latter should be implicit), and definitely call up Bell on this |
| 15:48.28 | Kobaz | probably /etc/init.d/asterisk |
| 15:49.12 | tzafrir | Kobaz, also depending on your kernel |
| 15:49.21 | tzafrir | IIRC it's mostly safe as of 2.6.25 |
| 15:49.49 | tzafrir | you'd have to try there very hard to get your system hang |
| 15:50.38 | Kobaz | tzafrir: it's killed me in >= 2.6.27 |
| 15:51.22 | Kobaz | i asked around in -dev, and the concensus was "don't use it" |
| 15:52.39 | kannan | hello, in cases on "got SIP response 400 Bad request", is it invariable the asterisk server config at fault? |
| 15:52.47 | *** join/#asterisk The_Boy_Wonder (n=vossel@asterisk/batman-developer/dvossel) |
| 15:52.47 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 15:55.21 | [TK]D-Fender | kannan: No. Perhaps you should SHOW US the complete communication. |
| 15:55.27 | bpgoldsb | Hmm, well with that fixed, now I have to figure out why Asterisk can't find all it's XML doc info |
| 15:55.47 | p3nguin | "its" |
| 15:55.51 | *** join/#asterisk sgimeno (n=santiago@226.Red-80-33-64.staticIP.rima-tde.net) |
| 15:55.53 | jo8330 | [TK]D-Fender: thanks. Appreciate your investigation. i'll try that out and see what happens, but indeed I'll definitely call Bell |
| 15:57.10 | *** join/#asterisk lost_sou1 (n=noymfb@cpe-74-71-234-100.twcny.res.rr.com) |
| 15:57.52 | p3nguin | Why do people have such a hard time with possessive pronouns? They aren't new; they were taught in school. |
| 15:58.18 | bpgoldsb | p3nguin: I failed English. It's why I went to my safety-subject of Engineering. |
| 15:58.44 | p3nguin | interesting concept |
| 15:59.20 | *** part/#asterisk voipmonk (n=shido6@dsl-67-204-37-228.acanac.net) |
| 15:59.23 | *** join/#asterisk voipmonk (n=shido6@dsl-67-204-37-228.acanac.net) |
| 15:59.29 | fenrus_ | it's not that easy if you dont speak english every day :) |
| 15:59.31 | bpgoldsb | I have a feeling Asterisk developers were in a similiar situation with error messages like: 'xmldoc.c: Counldn't find function CHANNEL in XML documentat' |
| 15:59.55 | p3nguin | Those are just horrible typos. |
| 15:59.58 | shamelessn00b | hurr |
| 16:00.13 | shamelessn00b | finally managed to modify the applicaion mp3player |
| 16:00.26 | p3nguin | I'm sure the person didn't really think the word was "Counldn't" |
| 16:00.39 | shamelessn00b | now whenever the user presses a key instead of simply exiting and moving to the next line in dialplan |
| 16:00.47 | shamelessn00b | it jumps to the specified context |
| 16:01.02 | p3nguin | That didn't happen before? |
| 16:01.10 | shamelessn00b | no |
| 16:01.21 | shamelessn00b | I added a few lines of code in the .c file |
| 16:01.24 | voipmonk | shamelessn00b! |
| 16:01.33 | shamelessn00b | hey voipmonk sup |
| 16:01.37 | soman | Hi, I am using the asterisk 1.6 with TE121 card installed... I am getting the error WARNING[15106]: chan_dahdi.c:11826 pri_dchannel: PRI Error on span 0: We think we're the CPE, but they think they're the CPE too. what could be the problem |
| 16:01.59 | [TK]D-Fender | soman: Both ends are trying to act like CPE. So STOP :p |
| 16:02.00 | voipmonk | the sky the ceilin' how ya feelin'? |
| 16:02.06 | coppice | soman: how much clearer so you want it stated? |
| 16:02.08 | shamelessn00b | both ends of the PRI are configured as master |
| 16:02.26 | shamelessn00b | both ends are providing the clock |
| 16:02.49 | coppice | that has nothing to do with who provides the clock |
| 16:02.53 | voipmonk | there can be only one master... the one with the clock |
| 16:03.00 | voipmonk | -l |
| 16:03.13 | soman | <[TK]D-Fender>: how to do that |
| 16:03.15 | voipmonk | thats not always true |
| 16:03.24 | voipmonk | just cuz u have a big clock doesnt mean anything |
| 16:04.00 | shamelessn00b | voipmonk I modified the source of the application mp3player ^^ |
| 16:04.09 | voipmonk | i noticed... |
| 16:04.35 | *** join/#asterisk ManxPower-work (n=EWieling@216.186.151.147) |
| 16:04.38 | shamelessn00b | and only cuz I was having issues with the same old problem |
| 16:04.46 | shamelessn00b | musiconhold and meetme |
| 16:05.01 | shamelessn00b | the streams kept on stopping at randiom |
| 16:06.23 | [TK]D-Fender | soman: vi chan_dahdi.conf |
| 16:07.04 | soman | <[TK]D-Fender>: yes, signalling is set to pri_cpe |
| 16:07.34 | shamelessn00b | anyone used sox to downsample 44 k wav files to 8 k mp3 |
| 16:08.05 | [TK]D-Fender | soman: And what are you connecting it to? |
| 16:08.16 | Naikrovek | 8k mp3? no why would you do that |
| 16:08.24 | [TK]D-Fender | shamelessn00b: WTF are you transcoding TO MP3 for? |
| 16:08.39 | _cgc | shamelessn00b: isn't it illegal to use mp3 without buying a license? |
| 16:08.40 | *** part/#asterisk stmaher (n=stephen@80.68.89.200) |
| 16:08.49 | soman | <[TK]D-Fender>: to a PRI cable |
| 16:08.50 | shamelessn00b | songs |
| 16:09.01 | shamelessn00b | [TK]D-Fender: songs |
| 16:09.08 | [TK]D-Fender | shamelessn00b: Thats the interpretation of the sound. I'm asking about the FORMAT |
| 16:09.19 | Naikrovek | shamelessn00b: encode them to 8k wav. mp3 sounds like ass at that sample rate |
| 16:09.19 | ManxPower-work | shamelessn00b: transcode them to .wav |
| 16:09.58 | voipmonk | Naikrovek: why do I think about Area51 when I see your name |
| 16:10.04 | voipmonk | or Aliens |
| 16:10.07 | Naikrovek | or .ogg if you're hell bent on using some compressed audio. i recommend .wav though |
| 16:10.08 | shamelessn00b | actually I have to stream them and cant find a streamer for wav files |
| 16:10.36 | Naikrovek | voipmonk: got me. if you figure it out, let me know. i'm dying to fell cool(er) |
| 16:10.37 | ManxPower-work | shamelessn00b: then don't stream them in Asterisk |
| 16:10.49 | shamelessn00b | they sound pretty decent actually |
| 16:10.58 | shamelessn00b | even better than wav |
| 16:10.58 | voipmonk | alienware? |
| 16:11.11 | Naikrovek | mpg123 can stream them and convert them to proper format and samplerate at the same tiem |
| 16:11.16 | ManxPower-work | shamelessn00b: asterisk will just transcode the audio anyway |
| 16:11.19 | shamelessn00b | I calculated the MOS values |
| 16:11.23 | bmoraca | shamelessn00b, that comment right there shows that you don't know much about voice codecs |
| 16:11.44 | *** part/#asterisk ManxPower-work (n=EWieling@216.186.151.147) |
| 16:11.49 | Naikrovek | "even better than wav" haha |
| 16:12.01 | Naikrovek | bmoraca: yes, that speaks volumes |
| 16:12.02 | shamelessn00b | :P |
| 16:12.19 | coppice | calculating MOS values sounds very marketing dept :-) |
| 16:12.21 | Naikrovek | shamelessn00b: wav is uncompressed, unmolested audio. raw audio |
| 16:13.02 | Naikrovek | you can change the sample rate, and sample size in a wav file, and lose data, but it's still uncompressed |
| 16:13.04 | bmoraca | Naikrovek, not quite unmolested...PCM isn't exact if you sample it at 8000hz using 8 bits, as telephones do |
| 16:13.12 | shamelessn00b | Naikrovek: but how do I stream uncompressed 44k 16 bit wav files, any idea? |
| 16:13.40 | shamelessn00b | when I already know all that bandwidth is going to be wasted anyways |
| 16:13.52 | bmoraca | shamelessn00b, how are you trying to stream them? there are a hundred ways...windows media services is one example |
| 16:14.07 | shamelessn00b | why should I stream in 44k |
| 16:14.17 | shamelessn00b | when its already going to get downsampled at 8k |
| 16:14.22 | [TK]D-Fender | shamelessn00b: Where are they being streamed from? |
| 16:14.35 | shamelessn00b | from a streaming server |
| 16:14.39 | shamelessn00b | Im using icecast |
| 16:14.42 | drmessano | lol |
| 16:14.52 | bmoraca | shamelessn00b, why are you asking this in an asterisk channel? |
| 16:14.53 | [TK]D-Fender | shamelessn00b: WHERE THE FUCKING HELL IS IT RELATIVE TO YOUR DAMN SERVER? |
| 16:14.57 | bmoraca | check #icecast |
| 16:15.05 | shamelessn00b | what |
| 16:15.16 | shamelessn00b | Im not asking how to fkin stream audio into asterisk |
| 16:15.26 | [TK]D-Fender | sense more people with serious comprehension issues. |
| 16:15.26 | shamelessn00b | I've already done that part |
| 16:15.44 | drmessano | Oh god... now we're compressing in icecast, and transcoding from the icecast stream to Asterisk |
| 16:15.59 | shamelessn00b | facepalms |
| 16:16.04 | [TK]D-Fender | drmessano: Don't forget the extra #2 coffee filter... |
| 16:16.20 | shamelessn00b | I'm using sox to downsample and convert wav into mp3 |
| 16:16.23 | shamelessn00b | then streaming |
| 16:16.29 | drmessano | WHY NOT JUST STREAM THE SOUND OF A FRIDGE CYCLING ON AND OFF???? Erm, sorry |
| 16:16.30 | shamelessn00b | and asterisk is playing the stream as is |
| 16:16.46 | drmessano | Asterisk isnt playing it "as-is" |
| 16:16.58 | shamelessn00b | mpg123 spits the stream on stdout |
| 16:16.59 | thehar | russellb: i have had to block your travel updates on friendface.. they piss me off. lol *jealous* |
| 16:17.09 | shamelessn00b | of that channel |
| 16:17.16 | drmessano | i dont know of ANY phone that supports MP3 |
| 16:17.27 | shamelessn00b | as raw |
| 16:17.34 | drmessano | So therefore, you're transcoding |
| 16:17.39 | bmoraca | shamelessn00b, http://www.lmgtfy.com/?q=sox convert 44khz to 8khz first link |
| 16:18.08 | drmessano | WAV ---> MP3 ---> Phone's native format |
| 16:18.27 | drmessano | I guess "codec" would be more appropriate |
| 16:18.47 | russellb | thehar: you blocked me? :'-( |
| 16:19.02 | thehar | russellb: no no.. i blocked that app you use for your travels on my newsfeed |
| 16:19.05 | coppice | drmessano: or dec, since your signal only goes one way |
| 16:19.16 | thehar | i could never block teh russellb |
| 16:19.17 | thehar | woof |
| 16:19.19 | drmessano | True |
| 16:19.41 | russellb | thehar: <3 ... tripit is pretty nice for organizing travel plans, though. |
| 16:19.57 | thehar | russellb: hehe yes yes however it reminds me of where i'm not going |
| 16:20.20 | drmessano | Why not get a nice ESATA drive, load that bad boy up with all your Miley Cyrus tunes, hook it up to the PBX, and let it roll |
| 16:21.01 | [TK]D-Fender | drmessano: Royalties ;) |
| 16:21.14 | [TK]D-Fender | litigates |
| 16:21.34 | coppice | ESATA sounds perfect for Miley Cyrus. So easy to unplug and dispose of |
| 16:21.41 | drmessano | [TK]D-Fender: Unless that Icecast server is on Sealand, he's got that problem either way |
| 16:22.58 | [TK]D-Fender | drmessano: So far he never said WHAT music he was streaming. AFAIK, Miley Cryrus hasn't released anything into the PD yet.... so he could be off the hook for other mystery stuff |
| 16:23.26 | *** join/#asterisk sebbl (n=Momofu@HSI-KBW-078-043-193-153.hsi4.kabel-badenwuerttemberg.de) |
| 16:23.34 | coppice | people should be hung for playing Miley Cyrus, even after copyright has expired |
| 16:24.02 | drmessano | [TK]D-Fender: For the sake of argument, I will assume he's got an icecast box loaded up with royalty free tunes he jams to. |
| 16:24.11 | drmessano | ROLLS EYES SO FAR BACK INTO HEAD BRAIN HURTS |
| 16:24.12 | drmessano | Erm, sorry |
| 16:24.55 | drmessano | goes off to smoke some banana peels |
| 16:25.04 | casix | is possible to monitor a parked call? or monitor the call after someone get this call out of parking? |
| 16:26.02 | p3nguin | shamelessn00b: If you're wanting to use it for MoH, I use "mpg123 -q -b 128 --preload 32 -r 8000 -f 2048 -m -s $streamurl" and it works perfectly. |
| 16:26.34 | [TK]D-Fender | casix: Sure... call Monitor before picking up the call. |
| 16:32.25 | _cgc | msg voipmonk have you heard of the sound for 1 side of a outbound sip call not working before? |
| 16:33.31 | p3nguin | hmm |
| 16:34.49 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 16:36.13 | *** join/#asterisk murraytm (n=murraytm@wsip-68-224-219-238.br.no.cox.net) |
| 16:37.07 | shamelessn00b | p3nguin: no I've had enough of MoH |
| 16:37.09 | shamelessn00b | lol |
| 16:37.22 | shamelessn00b | MoH + MeetMe = epic fail |
| 16:37.28 | p3nguin | What other reason would a person stream mp3s onto Asterisk? |
| 16:37.54 | murraytm | is there a catch to getting custom date formats to work with say.conf mode=new ? SayUnixTime(,CST,ABd 'digits/at' IMp) doesn't play anything at all if mode=new but works as expected if mode=old. |
| 16:37.58 | shamelessn00b | cuz apparently wav doesnt sound as good as mp3s do on cell phones |
| 16:38.07 | shamelessn00b | for some odd reason that I fail to understand |
| 16:38.32 | p3nguin | What is the usage of putting the music onto a cell phone? |
| 16:38.35 | *** join/#asterisk darkskiez (n=dz@62-50-207-156.client.stsn.net) |
| 16:38.44 | shamelessn00b | Im playing the same songs using the playback app in wav format in a seperate context |
| 16:38.47 | casix | [TK]D-Fender: when I try it the monitor stops just after picking up the call. This is a piece of the code: http://pastebin.com/m282cf556 |
| 16:39.01 | shamelessn00b | p3nguin: it makes $$$ |
| 16:39.31 | *** join/#asterisk moy (n=moy@189.162.161.141) |
| 16:39.37 | p3nguin | I still don't get it. You just have a music extension? People call your number and listen to music? That's all? |
| 16:39.39 | shamelessn00b | dunno, we have many users dialing into our boxes that wanna call and listen to music on cell phones |
| 16:39.41 | shamelessn00b | lol |
| 16:40.04 | p3nguin | On a cell phone, nevertheless. |
| 16:40.04 | shamelessn00b | they can get lyrics |
| 16:40.09 | [TK]D-Fender | casix: Then dial a non-rebridging local channel to do the pickup then. |
| 16:40.24 | shamelessn00b | or set the songs as thier ringtones or ringbacktones |
| 16:40.26 | shamelessn00b | etc etc |
| 16:40.41 | shamelessn00b | then we are offering song dedications |
| 16:40.44 | shamelessn00b | and kareoke |
| 16:40.49 | shamelessn00b | stuff like that |
| 16:41.46 | *** join/#asterisk ManxPower-work (n=EWieling@216.186.151.147) |
| 16:43.25 | *** join/#asterisk jtodd (i=n6qlzbwd@asterisk/community-director-and-tie-dye-shirt-lover/jtodd) |
| 16:43.26 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 16:44.30 | casix | [TK]D-Fender: ok, i'll try. Do you know how to play a dynamic announcement to the called before bridge the call? I'm try to do it with the G option of dial and then play the announcement and join the calls parking one and getting it from the other. Any best solution? oh and monitor the final conversation |
| 16:44.54 | *** join/#asterisk imcdona (n=t@c-24-19-203-112.hsd1.wa.comcast.net) |
| 16:45.33 | [TK]D-Fender | casix: Go try stuff and show us a failure |
| 16:46.55 | casix | [TK]D-Fender: ok thx :) |
| 16:47.07 | *** join/#asterisk moy (n=moy@189.162.161.141) |
| 16:48.02 | *** join/#asterisk ruben23 (n=AGENT@122.55.48.243) |
| 16:48.59 | *** join/#asterisk linuxviewer (n=example@ip72-222-249-220.ph.ph.cox.net) |
| 16:50.29 | *** join/#asterisk _cgc (n=_cgc@94-193-99-128.zone7.bethere.co.uk) |
| 16:53.18 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 16:53.31 | *** join/#asterisk mnicholson_ (n=mnichols@nat/digium/x-pxmeyohcxqmjzjqv) |
| 16:53.53 | *** join/#asterisk The_Boy_Wonder (n=vossel@asterisk/batman-developer/dvossel) |
| 16:54.03 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 16:54.03 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 16:55.01 | *** join/#asterisk fish-bulb (n=cstewart@nat/digium/x-pyxgrnprprnikskh) |
| 16:55.15 | *** join/#asterisk Chesther (n=cam2@cam2-mac.cit.cornell.edu) |
| 16:56.46 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 16:56.47 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 17:10.33 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 17:11.49 | *** join/#asterisk Badrobot- (n=badrobot@cpe-76-173-229-89.socal.res.rr.com) |
| 17:16.09 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 17:16.09 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 17:18.00 | *** join/#asterisk Geminizer (n=whoami@cpe-76-180-27-4.buffalo.res.rr.com) |
| 17:18.15 | Geminizer | hello... can a T1 support 50 concurrent calls ? |
| 17:18.23 | [TK]D-Fender | Geminizer: No |
| 17:18.25 | Naikrovek | voice t1: no |
| 17:18.42 | Naikrovek | data t1: yes, if G729 and an IAX2 trunk is used |
| 17:18.44 | [TK]D-Fender | Geminizer: Non-data, that is |
| 17:18.47 | *** join/#asterisk joako (n=ston3d@opensuse/member/joak0) |
| 17:18.50 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 17:19.00 | Geminizer | what is the upper bound for T1, and what would need to be in place for supporting 50 calls? |
| 17:19.14 | Naikrovek | you can get like 48 G729 calls over a SIP "trunk" |
| 17:19.14 | [TK]D-Fender | Geminizer: Perhaps you should mention what you putting OVER it. |
| 17:19.47 | [TK]D-Fender | 1544/33 |
| 17:20.09 | [TK]D-Fender | Assuming best case scenario... which never happens |
| 17:20.10 | Naikrovek | i thought 32 |
| 17:20.51 | Naikrovek | data T1 + G729 + IAX2 trunk = (about) 140 simultaneous calls |
| 17:21.03 | Naikrovek | unfortunately IAX2 trunks to providers aren't common |
| 17:21.10 | Geminizer | ok... so even if a SIP trunk is being used, no change in carrier alone can increase the capacity? |
| 17:21.25 | Naikrovek | you're limited to the amount of data a T1 can carry |
| 17:21.36 | [TK]D-Fender | Naikrovek: Int heory... have you seen IAX2 trunking survivability stats? ;) |
| 17:21.47 | Naikrovek | [TK]D-Fender: haven't |
| 17:22.10 | [TK]D-Fender | Naikrovek: the Titanic had a high passenger capacity too ;) |
| 17:22.31 | Geminizer | So, for ulaw and a SIP trunk, what would be a "comfortable" capacity? |
| 17:22.40 | Naikrovek | 17 calls |
| 17:22.52 | coppice | [TK]D-Fender: icebergs pass well above the fibres |
| 17:23.04 | Geminizer | wow... ok, thanks |
| 17:23.57 | Naikrovek | a T1 is not a lot of data |
| 17:24.59 | [TK]D-Fender | As compared to other mediums |
| 17:25.04 | Naikrovek | correct |
| 17:25.10 | Naikrovek | wish i could get some other mediums in here |
| 17:25.21 | Naikrovek | in my place of employment, i mean |
| 17:25.43 | Naikrovek | going to have to get something else, and leave phone on the T1 |
| 17:25.58 | Naikrovek | stupid provider has apparently not billed us for that T1 for 18 months, i'm told |
| 17:26.08 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.157) |
| 17:26.14 | *** join/#asterisk raden_work (n=jon@69-179-99-17.stat.centurytel.net) |
| 17:32.38 | Naikrovek | so rather than upgrade that connection i wanna get another and put everythign but the phone on it |
| 17:33.51 | p3nguin | Where did that 33 (or 32) value come from to calculate the number of calls? I mean, what does that value represent? |
| 17:36.24 | [TK]D-Fender | p3nguin: Number of simultaneous calls. What have you missed? |
| 17:36.36 | p3nguin | Uh, no. |
| 17:37.03 | p3nguin | If 1544/33 = number of simul.calls, then 33 does not equal the number of simul. calls. |
| 17:37.16 | sbrath | Is the only way to get SRTP in Asterisk to use the SVN asterisk-srtp trunk ? |
| 17:37.26 | p3nguin | So what does the 33 value represent? |
| 17:37.59 | _cgc | [TK]D-Fender: i fixed my problem with the audio not working 1 way, it was that the audio was coming from a different IP address and was being blocked coming in on ports 10000-20000 :) |
| 17:38.19 | [TK]D-Fender | _cgc: SMRT |
| 17:38.28 | _cgc | thanks for your help :) |
| 17:38.36 | [TK]D-Fender | p3nguin: kbps/call... seriously... |
| 17:41.24 | *** join/#asterisk maximo (n=maximo@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 17:41.53 | p3nguin | If the required bandwidth of a G.729 call is 31.2 Kbps, how did you arrive at 33? Is this some arbitrary number that includes overhead or what? You're not being complete, and you're acting like I'm supposed to be psychic and know what you mean. |
| 17:43.27 | [TK]D-Fender | p3nguin: I rounded up a tad |
| 17:43.47 | p3nguin | For the purpose of overhead, or just for the heck of it? |
| 17:45.04 | p3nguin | How do we factor in SIP/RTP and/or IAX2 (with and without trunking) when trying to calculate the number of simultaneous calls? |
| 17:45.20 | p3nguin | Maybe someone made a chart. |
| 17:46.12 | Tim_Toady | there are some charts, astricon 2009 "IAX to carrirers" presentation |
| 17:46.19 | p3nguin | Unless the bandwidth chart I use is all-inclusive... which I don't think it is. |
| 17:46.22 | Tim_Toady | comparing sip and iax peers |
| 17:47.53 | *** join/#asterisk raden_work (n=jon@69-179-99-17.stat.centurytel.net) |
| 17:48.28 | raden_work | there any disadvantage to the unlocked PAP2T's ? |
| 17:48.33 | *** join/#asterisk Katty (n=User@adsl-70-253-169-127.dsl.stlsmo.swbell.net) |
| 17:48.57 | Tim_Toady | apart from being free to use it? |
| 17:49.29 | Katty | hi. |
| 17:49.33 | Katty | my asterisk does not work /at all/ |
| 17:49.33 | [TK]D-Fender | raden_work: if they were previously unlocked they might try to contact the provisioner again and relock |
| 17:49.34 | Katty | how to fix pls. |
| 17:49.51 | [TK]D-Fender | Katty: rm -r....awww fukkit |
| 17:49.54 | raden_work | [TK]D-Fender, thank |
| 17:49.56 | raden_work | you |
| 17:50.04 | Katty | rm -r / |
| 17:50.31 | *** join/#asterisk |AnToS| (n=31749@93-44-17-172.ip95.fastwebnet.it) |
| 17:50.35 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 17:50.36 | Geminizer | nah... rm -rf / |
| 17:50.53 | coppice | [TK]D-Fender: I guess the locking scheme is the only bit they ever bothered to properly debug |
| 17:51.04 | *** join/#asterisk DarkFibre (n=dmelouk@127.159.119.70.cfl.res.rr.com) |
| 17:52.04 | Katty | how to use rm????' |
| 17:52.09 | Katty | i do not know linux |
| 17:52.32 | Zhad | [TK]D-Fender: IIRC, you can fix it so that it doesn't. |
| 17:52.43 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 17:52.50 | Katty | hi tony |
| 17:52.58 | Zhad | There is a Yahoo Groups group with lots of info about it. |
| 17:53.20 | Zhad | took his PAP2s out of service a long time ago. and they were never locked in the first place. |
| 17:54.04 | carrar | Y*A*W*N |
| 17:55.44 | Naikrovek | offtopic: anyone have a canon imagerunner 210 |
| 17:56.05 | Katty | i don't |
| 17:56.05 | p3nguin | I have a canon, but it doesn't deal with images. :) |
| 17:56.15 | Naikrovek | mine doesn't either - it's a photocopier |
| 17:56.19 | Naikrovek | well |
| 17:56.27 | Naikrovek | maybe that's the image in the "imagerunner" moniker |
| 17:56.36 | Katty | i am familier with copiers |
| 17:56.43 | Katty | i'm actually certified on 1 kyocera model |
| 17:56.47 | Katty | but not canons. |
| 17:56.48 | Naikrovek | neat |
| 17:56.51 | raden_work | [TK]D-Fender, if you were running a ITP what would you use for POTS Adapaters ? |
| 17:57.19 | [TK]D-Fender | Katty: Using Kyrocera photocopiers certainly qualifies you as "certifiable" :p |
| 17:57.22 | Naikrovek | i have an imagerunner and it's only got 67k page count, and it's giving me an error |
| 17:57.53 | Naikrovek | on phone with service folks |
| 17:58.09 | Katty | [TK]D-Fender: why don't you say something nice to me. |
| 17:58.15 | Katty | [TK]D-Fender: like, Katty you're such a lovely person! |
| 17:58.20 | Katty | [TK]D-Fender: or Katty! let's hug! |
| 17:58.27 | ZenBSDi | Katty, you a girl? |
| 17:58.42 | raden_work | ZenBSDi, something like that |
| 17:58.42 | Katty | no i'm a 50 year old male in a basement. |
| 17:58.54 | p3nguin | HEY! |
| 17:58.59 | ZenBSDi | probably a gurl :p cross dressers :p |
| 17:59.11 | Katty | hmm. |
| 17:59.17 | Katty | i did wear one of ryan's hoodies once |
| 17:59.27 | Katty | it's warm ^_^ |
| 17:59.41 | Katty | hi p3nguin |
| 17:59.48 | p3nguin | hello, katty |
| 18:00.07 | Katty | do you have sunny skies up north? |
| 18:00.10 | *** join/#asterisk nny (n=scott@64.203.239.83) |
| 18:00.17 | p3nguin | yeah |
| 18:00.27 | p3nguin | Up to 30 degrees, already. |
| 18:00.28 | Katty | :>>> |
| 18:00.38 | Katty | 31F here, 22 with windchill |
| 18:00.44 | Katty | pressure is falling tho :< never a good sign. |
| 18:01.17 | p3nguin | I don't know if we'll keep the sun long enough to melt the snow, but it has been shining when the clouds aren't in the way. |
| 18:01.45 | nny | so someone asked me if I could write a dialplan that could allow "seamless" transfers from desk phone to cell phone. Not looking for answers so much as opinions on if it can be done or not. By seamless I mean "press *X, cell phone rings, caller still attached to deskphone" press something else on cell or desk and call is now on cell |
| 18:02.11 | nny | i know features.conf would allow me to inject the *x part |
| 18:02.16 | p3nguin | Of course it is possible. |
| 18:02.35 | nny | well then I shall start trying to craft it |
| 18:02.41 | Katty | it's your diaplan. you can do whatever you like. |
| 18:02.50 | nny | sweet! rick rolls for everyone! |
| 18:02.52 | Katty | you can feed the cats if you want. |
| 18:03.16 | nny | heh I joke that it can make toast with a proper X11 or serial port setup |
| 18:03.41 | Chesther | "Press 1 for toast. Press 2 for a bagel with cream cheese." |
| 18:03.42 | nny | x10* |
| 18:03.42 | ZenBSDi | my $agi = new Asterisk::AGI; $Katty = $agi->get_data("is_it_a_boy_or_a_gurl",10000,1); return $Katty; |
| 18:03.53 | Katty | you can make toast |
| 18:03.58 | Katty | why not just run the eject command |
| 18:04.05 | Katty | and rig something up to the cdrom drive gears |
| 18:04.11 | nny | hehe |
| 18:04.24 | nny | i read a story about a pc that did that in a server room somewhere |
| 18:04.44 | nny | http://thedailywtf.com/Articles/ITAPPMONROBOT.aspx |
| 18:04.44 | Katty | ZenBSDi: why does it matter? |
| 18:05.12 | Katty | ZenBSDi: are you gender biased? |
| 18:05.17 | ZenBSDi | if($server =~ /PROBLEMS/) { die "$server has issues\n"; } |
| 18:05.17 | Katty | ZenBSDi: ARE YOU SEXIST |
| 18:05.27 | ZenBSDi | O.o |
| 18:05.32 | p3nguin | He wants to cyber you... maybe? |
| 18:05.45 | nny | i put on my lineman's set and hat |
| 18:05.52 | ZenBSDi | my $wood = "needs pleasing"; bless($wood); |
| 18:06.02 | hardwire | nny: the one with two beer holders and giant crazy straws? |
| 18:06.06 | *** join/#asterisk Ad-Hoc (n=nimbus@62.1.130.79.dsl.dyn.forthnet.gr) |
| 18:06.10 | Katty | ZenBSDi: that is highly inappropriate. |
| 18:06.11 | bpgoldsb | Hmm, was there a change between 1.6.1 and 1.6.2 that de-randomized SIP channel names? |
| 18:06.27 | ZenBSDi | Sorry, I've been coding perl + asterisk::agi all weekend :p |
| 18:06.31 | hardwire | bpgoldsb: those are hashes, I thought |
| 18:06.40 | hardwire | in all versions |
| 18:07.04 | p3nguin | ${UNIQUEID} is no longer unique? |
| 18:07.04 | nny | hardwire: oh god no, beer can holders are so inappropriate |
| 18:07.07 | bpgoldsb | Well, my 1.6.2 install ation has SIP/116-0000000(1-5) |
| 18:07.14 | nny | hardwire: this one has enough room for two handles of jack |
| 18:07.18 | ZenBSDi | I luv t3h * w /usr/bin/perl |
| 18:07.34 | bpgoldsb | On the first few calls I made, they were sequential. Hash, I'd expect to be non-sequential |
| 18:08.03 | hardwire | bpgoldsb: yeh.. I thought the hash of the index number + the SIP name = the channel name |
| 18:08.19 | ZenBSDi | bpgoldsb, I've noticed that a few times too.. depending on how far apart the calls are the hash could be or won't be that sequential |
| 18:08.36 | hardwire | I haven't noticed that btw.. I am using 1.6.2 and getting hashes |
| 18:08.57 | ZenBSDi | I'm still on 1.4 :p |
| 18:09.00 | bpgoldsb | Perhaps my terms are messed up. SIP/112-00000000... 00000000 is the channel or the index #? |
| 18:09.16 | bpgoldsb | I'm still on 1.2.2x :) |
| 18:09.19 | ZenBSDi | I need to setup 1.6 and start doing some kewl perls for 1.6 and ramps its up with memcached |
| 18:09.22 | bpgoldsb | Thus why I'm deploying 1.6.2 |
| 18:09.46 | hardwire | bpgoldsb: SIP/112-ABCDEFG would be the channel reference that is unique enough to query. |
| 18:09.51 | Kobaz | in 1.6.x the channel suffix was changed to be incremental instead of randomly generated |
| 18:10.01 | hardwire | at least unique in the midst of all other calls at that time |
| 18:10.12 | Kobaz | yeah, channel names can get reused |
| 18:10.30 | Kobaz | it'll take a while to get a reused channel name with the new method though |
| 18:10.36 | hardwire | Kobaz: I haven't noticed sequential names.. I'm using 1.6.2 |
| 18:10.37 | hardwire | weird |
| 18:10.52 | Kobaz | hardwire: oh hmm, it does that on mine |
| 18:10.59 | Kobaz | with sip anyway... iax is still random |
| 18:11.09 | hardwire | Kobaz: weird indeed.. maybe you guys are missing the crypto modules :) |
| 18:12.00 | Kobaz | answered SIP/3030-000001e6 |
| 18:12.03 | Kobaz | answered SIP/3007-000001e7 |
| 18:12.27 | hardwire | you have a lot of calls going |
| 18:12.42 | Kobaz | it's always incrementing by one |
| 18:13.08 | Kobaz | not really a lot of calls |
| 18:13.12 | Kobaz | one call an hour? |
| 18:13.14 | Kobaz | it's a small office |
| 18:13.16 | Katty | they're all from me |
| 18:13.26 | hardwire | err. |
| 18:13.27 | Kobaz | yeah, i know how much you love me :) |
| 18:13.44 | hardwire | SIP/ancwas-b5501f98 SIP/ancwas-086d42f0 |
| 18:13.44 | Katty | yep. |
| 18:13.49 | hardwire | 1.6.2 |
| 18:13.55 | Kobaz | this is on 1.6.0.20 |
| 18:14.28 | hardwire | 1.6.2.0~rc2-0.7616 |
| 18:14.29 | hardwire | heh |
| 18:14.53 | Kobaz | rc2 |
| 18:14.53 | Kobaz | heh |
| 18:15.00 | Kobaz | i had a lot of problems wiht rc2 |
| 18:15.04 | Kobaz | odbc kept crashing |
| 18:16.13 | Kobaz | all i want for christmas is a sta-ble asterisk... a sta-ble asterisk |
| 18:16.20 | Kobaz | do de do de do |
| 18:16.24 | fenrus_ | :) |
| 18:16.31 | hardwire | 1.2 is pretty stable |
| 18:16.35 | Kobaz | yeah |
| 18:16.36 | *** join/#asterisk Bioh (n=biohazar@216.113.117.197) |
| 18:16.44 | Kobaz | the problem is, i need quite a number of the features in 1.6 |
| 18:16.44 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 18:16.48 | hardwire | it lacks a fancy |
| 18:17.07 | Kobaz | it sucks that 1.6.0 is going to be eol in a few months |
| 18:17.08 | hardwire | ah.. my other machines are 1.6.2.0-0.7901 |
| 18:17.10 | Katty | it lacks fancy? |
| 18:17.14 | Katty | put some racing stripes ont he side. |
| 18:17.23 | hardwire | Katty: get real, it's software. |
| 18:17.27 | hardwire | you can't paint on a software |
| 18:17.36 | hardwire | is all about limits today |
| 18:17.36 | Kobaz | there needs to be a long-standing maintenance branch that's rock super damn solid |
| 18:17.48 | ManxPower-work | Apparently you don't know Katty very well. |
| 18:17.51 | hardwire | Kobaz: that would mean you'd pay for it. |
| 18:18.07 | Kobaz | probably |
| 18:18.16 | fenrus_ | i'd paint my asterisk box if it made it rock solid. |
| 18:18.27 | Katty | gets real |
| 18:18.34 | Kobaz | some of the stuff in 1.6.2 is really awesome too |
| 18:18.35 | hardwire | Katty: nono.. don't get real |
| 18:18.35 | Katty | I R REAL KATTY NOW |
| 18:18.43 | Kobaz | like the dialplan Originiate(), and the mwi control |
| 18:18.51 | carrar | For realz? |
| 18:19.02 | Kobaz | for mad reels yo |
| 18:19.03 | hardwire | Kobaz: I'm a dahdi lover. |
| 18:19.04 | hardwire | sigh |
| 18:19.32 | Kobaz | they should have picked a better name for dahdi |
| 18:19.32 | Kobaz | heh |
| 18:19.56 | Chesther | Kobaz: 1.8 will be the next long-term-solid release. |
| 18:19.59 | Katty | for realzzzzzzz yo |
| 18:20.04 | Kobaz | Chesther: heh |
| 18:20.27 | Kobaz | .0 is hardly ever all that great |
| 18:20.38 | Chesther | True. |
| 18:20.40 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 18:20.44 | [TK]D-Fender | fenrus_: Fill it with paint and it will be solid |
| 18:20.45 | Katty | .0 is a great number. |
| 18:20.49 | Katty | .0 spinach in dinner. |
| 18:21.20 | Chesther | They talked about this at Astricon. After getting some, um, spirited feedback on the numbering scheme for 1.6.x, they changed it again. |
| 18:21.25 | [TK]D-Fender | [13:19]<hardwire>Kobaz: I'm a dahdi lover. <- that's odd... someone once called me a mother-f.... something or other... |
| 18:21.37 | hardwire | a mahtheri fucker. |
| 18:21.38 | hardwire | err.. |
| 18:21.40 | Chesther | There will be "standard" releases and "long term support releases". The standard ones will live for a year or two. |
| 18:21.40 | hardwire | I said it.. sorry |
| 18:21.48 | Chesther | The LTS ones, more like 4-5 years. |
| 18:21.50 | hardwire | bans self from channel for the day |
| 18:21.51 | Katty | applauds [TK]D-Fender |
| 18:21.56 | Kobaz | Chesther: hmm, where's that written up? |
| 18:22.12 | fenrus_ | http://www.asterisk.org/asterisk-versions |
| 18:22.20 | leifmadsen | Kobaz: http://blogs.asterisk.org |
| 18:22.32 | Kobaz | ooooh |
| 18:22.43 | leifmadsen | yes, we do document this stuff :) |
| 18:22.49 | Kobaz | well |
| 18:22.50 | Kobaz | yeah i know |
| 18:22.56 | Kobaz | but i saw the release chart a few weeks ago |
| 18:22.59 | Kobaz | and it didn't look like that |
| 18:23.10 | Kobaz | well, i saw a different one |
| 18:23.14 | leifmadsen | it's gone through some revisions as people have provided feedback (yes we listen to feedback!) |
| 18:23.31 | Kobaz | yeah |
| 18:23.34 | Kobaz | yay |
| 18:23.38 | Katty | p3nguin: ohohoh |
| 18:23.40 | leifmadsen | ~asteriskversioning |
| 18:23.41 | infobot | hmm... asteriskversioning is Information about the new Asterisk versioning method with the 1.6.x series is available here: http://www.asterisk.org/node/48602 |
| 18:23.42 | Katty | p3nguin: ohooh, it's 33F now :> |
| 18:23.58 | leifmadsen | infobot: no, asteriskversioning is http://www.asterisk.org/asterisk-versions |
| 18:23.58 | infobot | okay, leifmadsen |
| 18:24.06 | leifmadsen | ~asterisk-versions |
| 18:24.15 | leifmadsen | infobot: asterisk-versions is http://www.asterisk.org/asterisk-versions |
| 18:24.16 | infobot | okay, leifmadsen |
| 18:25.15 | leifmadsen | infobot asterisk-1.6-versioning is http://blogs.asterisk.org/2009/06/24/about-the-new-asterisk-versioning-method/ |
| 18:25.16 | infobot | okay, leifmadsen |
| 18:25.27 | Kobaz | i like the new versioning |
| 18:25.28 | *** join/#asterisk djMax (n=chatzill@66.92.91.132) |
| 18:25.31 | Kobaz | that's quite good |
| 18:26.02 | leifmadsen | we just give timeframes now |
| 18:26.19 | leifmadsen | we tried something, didn't work as expected, changed it to work better |
| 18:26.28 | Kobaz | well, the last time i looked at that chart, it didn't have standard/lts |
| 18:26.38 | Katty | puppy is napping on the couch so cutely :> |
| 18:26.40 | Katty | takes photo |
| 18:26.52 | fenrus_ | giev picture! |
| 18:27.05 | Kobaz | http://www.uniquedaily.com/wp-content/uploads/2009/10/fendsoff-bullying-birds.jpg |
| 18:27.23 | hardwire | damnit. |
| 18:27.26 | hardwire | one more blog to read |
| 18:27.30 | hardwire | punches leifmadsen |
| 18:27.36 | hardwire | sends leifmadsen a bill |
| 18:27.44 | leifmadsen | watches hardwire break his hand on steel |
| 18:27.49 | hardwire | orly? |
| 18:27.55 | leifmadsen | o'reilly. |
| 18:28.02 | hardwire | ya'reilly? |
| 18:28.13 | leifmadsen | o'rally |
| 18:28.24 | hardwire | walks off after being outwitted. |
| 18:28.29 | hardwire | walks it oooooffff.. |
| 18:28.30 | hardwire | walks it oooooffff.. |
| 18:28.55 | hardwire | is doing evil dahdi things today. |
| 18:29.01 | Kobaz | kicks firefox |
| 18:29.03 | Kobaz | stop being slow |
| 18:29.11 | hardwire | err.. evil DUNDi things today |
| 18:29.11 | hardwire | dangit. |
| 18:29.28 | nny | if adblock for chrome wasn't causing me issues I'd say use that ha |
| 18:29.43 | Katty | http://hphotos-snc3.fbcdn.net/hs131.snc3/17877_645591008657_37617946_36451246_3072433_n.jpg <- 90lbs in a ball. |
| 18:30.48 | Kobaz | i need to hurry up and port my new features to trunk so they make it into 1.8 |
| 18:31.21 | nny | can't you set a feature with DYNAMIC_FEATURES=buttscratcher via globals? |
| 18:31.41 | Katty | why don't you try it and find out |
| 18:32.15 | *** join/#asterisk xpot (n=xpot@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net) |
| 18:32.28 | nny | Katty: working on it, my business partner had DYNAMIC_FEATURES=> in the dialplan for something he was testing and wasn't sure if the => was a typo |
| 18:32.33 | *** join/#asterisk cusco (n=trilili@2001:0:53aa:64c:24a0:422f:a077:acde) |
| 18:32.35 | cusco | hello! |
| 18:32.40 | Katty | hai cusco |
| 18:32.57 | cusco | we just got a new pri service... |
| 18:33.05 | cusco | i just set up a test box |
| 18:33.14 | cusco | span 1 is active |
| 18:33.23 | Katty | how about them cookies. |
| 18:33.29 | cusco | bats ate them |
| 18:33.36 | Katty | :< |
| 18:33.49 | Katty | DARN YOU BATMAN |
| 18:34.07 | cusco | batmans is history, we got the "Edward" from twilight |
| 18:34.14 | cusco | anyways |
| 18:34.23 | Katty | mister sparkles. |
| 18:34.31 | cusco | yes thats the one |
| 18:34.42 | cusco | is reading trough the forth - breaking dawn |
| 18:34.44 | Katty | why does mister sparkles like high school girls? |
| 18:35.00 | cusco | he can't read her mind, thats why :p |
| 18:35.03 | Katty | does mister sparkles have an investment portfiolio? |
| 18:35.12 | jo8330 | [TK]D-Fender: Telco fixed CID on 800 numbers, now it works perfectly |
| 18:35.42 | cusco | should do! after 800 years living trough ordinary society, I bet he knows how to get round |
| 18:35.57 | cusco | anyway... DDI is 210066500 |
| 18:36.05 | cusco | when I call that number asterisk says: |
| 18:36.14 | [TK]D-Fender | joVery rare kind of issue... they should knwo better... but this is Bell Quebec we're talking about... |
| 18:36.18 | cusco | -- Extension '6500' in context 'incoming-pri' from '210332300' does not exist. Rejecting call on channel 0/1, span 1 |
| 18:36.29 | cusco | whats with the extension 6500 ? |
| 18:36.37 | Katty | why doesn't mister sparkles wear a matte finish foundation? |
| 18:36.41 | [TK]D-Fender | cusco: because your telco is only sneding 4 digits as the DID |
| 18:36.49 | cusco | ah! |
| 18:36.50 | [TK]D-Fender | cusco: Go tell them to send 10 |
| 18:37.04 | cusco | I see.... |
| 18:37.40 | p3nguin | Is that normal for a telco to only send 4? |
| 18:37.54 | cusco | not with our other telco |
| 18:37.57 | carrar | if you tell them too |
| 18:38.17 | p3nguin | I would expect by default, you'd get 10 or 11 digits. |
| 18:38.47 | nny | damn |
| 18:38.47 | carrar | 11 digits? |
| 18:38.58 | [TK]D-Fender | 10 |
| 18:39.08 | leifmadsen | I have seen many telcos only send the last 4 digits |
| 18:39.08 | p3nguin | 1 plus the 10-digit number |
| 18:39.10 | jo8330 | [TK]D-Fender: hehe. yeah it's not the first issue for this T1. hopefully it's the last. thanks again for your help earlier. |
| 18:39.14 | carrar | Please send me 25 digits every time!!! |
| 18:39.16 | Katty | we get sent 7 digits. |
| 18:39.18 | Katty | it's irritating |
| 18:39.28 | leifmadsen | I've worked with 2-3 customers whose telcos only send 4 digits |
| 18:39.31 | leifmadsen | (on a T1) |
| 18:39.34 | leifmadsen | (PRI) |
| 18:39.47 | Katty | what happens when your area code/dids overlap? :< |
| 18:39.51 | jo8330 | i like all my digitz |
| 18:39.52 | Katty | I KNOW WHAT HAPPENS |
| 18:39.52 | nny | hmm |
| 18:39.53 | leifmadsen | they don't :) |
| 18:39.53 | Katty | DOOM HAPPENS |
| 18:39.59 | leifmadsen | (not in that area anyways) |
| 18:40.08 | nny | any way to see what a command from applicationmap is doing if it's failing? |
| 18:40.08 | carrar | Use the BFG9000 |
| 18:40.11 | Katty | DOOM ON YOU |
| 18:40.12 | Katty | DOOM ON YOU |
| 18:40.25 | Katty | what is that from? |
| 18:40.31 | carrar | DOOM |
| 18:40.53 | Katty | ice age? |
| 18:41.08 | Naikrovek | futurama. Bender: we're all doomed! DOOOOOO [commercial break] OOOOMED |
| 18:41.11 | nny | i tried a simple cellswitch => *9,callee,Transfer,SIP/SOMENUMBER and it just dies, no console output |
| 18:41.35 | Katty | yeah it was the scene from the melons |
| 18:41.36 | Katty | and the birds |
| 18:41.37 | nny | er crao |
| 18:41.39 | nny | crap* |
| 18:41.40 | *** join/#asterisk mbthv9467 (n=mbt@fw.stratfor.com) |
| 18:41.42 | nny | i know why it dies |
| 18:41.49 | nny | but still wondering why the console says nada |
| 18:41.53 | Katty | "So you got three melons?" "if you weren't smart enough to plan ahead, then doom on you! doom on you! doom on you!" |
| 18:42.24 | Katty | mine? :> |
| 18:42.26 | Katty | mine! |
| 18:42.30 | p3nguin | nny: Did you try basic DTMF transfers and skip the feature code? |
| 18:42.35 | Katty | mine! mine! mine! mine! mine! mine! mine! |
| 18:42.52 | nny | p3nguin: hmm no will try that first |
| 18:43.13 | Katty | infobot: mine? |
| 18:43.14 | infobot | [mine] ircII |
| 18:43.22 | Katty | :< |
| 18:43.24 | Katty | bummer. |
| 18:43.27 | p3nguin | You might have to tweak the dialplan to allow dialing outbound from your current context. |
| 18:43.52 | nny | p3nguin: that seems to be the case |
| 18:45.23 | boch | does asterisk rotate its own logs ? |
| 18:45.44 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 18:46.48 | Katty | well it wouldn't want the fire to go out. |
| 18:48.18 | boch | im having this notice once per minute: Rotated Logs Per SIGXFSZ (Exceeded file size limit) file sizes are 40byte |
| 18:48.43 | carrar | 40byte log files are huge!! :) |
| 18:49.01 | *** join/#asterisk puzzled (n=foobar@puzzled.xs4all.nl) |
| 18:49.15 | p3nguin | Maybe logrotate is having a bad hair day. |
| 18:49.18 | carrar | fix your log rotate script |
| 18:49.25 | nny | p3nguin: hmm got it to barf out an error, but now nothing again heh |
| 18:49.42 | nny | p3nguin: declined, call miserably fails. heh |
| 18:49.46 | boch | asterisk is rotating the logs, it is getting the SIGXFSZ signal, but how can i know the sender of this signal |
| 18:50.26 | p3nguin | /etc/logrotate.d/asterisk causes mine to rotate, so I dunno. |
| 18:50.32 | Katty | marians. |
| 18:51.23 | leifmadsen | Asterisk 1.4.29-rc1, 1.6.0.21-rc1, 1.6.1.13-rc1, and 1.6.2.1-rc1 are now available for testing! More information can be found in the release announcement: http://www.asterisk.org/node/49884 |
| 18:51.35 | Katty | THE MARSIES ARE COMING!! PUT ON YOUR FOIL HATS |
| 18:51.44 | ChannelZ | make sure the limits in your /etc/asterisk/logger.conf aren't crazy |
| 18:52.06 | Kobaz | caaaaraaaazzzy |
| 18:52.09 | Katty | aww why can't it be the marsies sending the signal? )= |
| 18:52.11 | ChannelZ | heh |
| 18:52.40 | Katty | that'd be so much more fun. |
| 18:52.47 | Katty | always ruining my fun. |
| 18:52.48 | Katty | sulks. |
| 18:55.24 | mbthv9467 | hello all. Anyone ever run into multiple CLI 'originate' commands running synchronously? |
| 18:55.35 | ManxPower-work | boch: chances are you ran out of inodes for that directory. Make sure you don't have a bazillion log files in that directory |
| 18:56.27 | boch | ManxPower-work, i deleted the directory and created again, and once again started roting after few seconds, i checked the process limits and file size is unlimited |
| 18:57.42 | leifmadsen | OK... I think I'm officially nearly sick |
| 18:57.54 | Katty | :< |
| 18:58.00 | Katty | i'm sick too. stayed home today. |
| 18:58.05 | Katty | applies blanket to leifmadsen |
| 18:59.48 | leifmadsen | I stayed home too |
| 18:59.51 | leifmadsen | (I work from home) |
| 19:00.09 | leifmadsen | just took a big dose of Cold-FX and now having a glass of OJ to see if I can combat this sickness |
| 19:00.30 | Katty | what's in Cold-FX(tm) |
| 19:02.05 | *** join/#asterisk Skeeter- (i=Skeeter@c216.218.2-65.clta.globetrotter.net) |
| 19:02.48 | nny | so my idea is getting closer, but followme isn't quite qhat I need. anyone have a brief suggestion on how to transfer a call to another phone, yet still keep the caller on the phone while the second phone is being dialed? |
| 19:03.58 | Katty | i don't get the question |
| 19:03.59 | leifmadsen | Katty: 300mg of CVT-E002, a proprietary ChemBioPrint (CBP) product containing great than 80% poly-furanosyl-pyranosyl-saccharides extracted from Paneax quinqufolius (North American ginseng, root) |
| 19:04.09 | Katty | the caller is going to stay on the channel until one of them hangs up |
| 19:04.23 | Katty | leifmadsen: so it's ginseng? |
| 19:04.27 | nny | Katty: ok say I am have a call to my deskphone, and I want to move it to my cell. However I don't want the caller to hear the ringing or the transfer process |
| 19:04.31 | leifmadsen | Katty: or some extract from it... I guess |
| 19:04.47 | Katty | nny: what about music |
| 19:04.50 | nny | Katty: so essentially if I hit *something, it dials my cellphone and waits for the pickup before moving the call |
| 19:04.54 | leifmadsen | nny: sounds like an Originate() and Bridge() |
| 19:04.56 | carrar | leifmadsen, juice carrots, apples and catalope |
| 19:05.07 | nny | leifmadsen: shall research |
| 19:05.08 | leifmadsen | carrar: I don't have a juicer |
| 19:05.14 | carrar | (don't mix catalop in that juic emix) :) |
| 19:05.20 | carrar | GET ONE!! |
| 19:05.20 | Katty | catalope?! |
| 19:05.24 | carrar | Thats why you're sick |
| 19:05.37 | carrar | Just catalope juiced by it's self is awesome |
| 19:05.39 | Katty | he's sick cause someone breathed on him! |
| 19:05.47 | leifmadsen | I'm sick because I had a party on Saturday and likely someone else was sick -- I haven't been sick in several years |
| 19:05.47 | Katty | catalope?! |
| 19:05.50 | carrar | with skin |
| 19:06.06 | carrar | it's like a thick shake full of vitamins |
| 19:06.08 | nny | leifmadsen: looks like it's time I learn to use the AGI heh |
| 19:06.14 | Katty | you mean cantaloupe? :P |
| 19:06.28 | carrar | sure :) |
| 19:06.59 | carrar | http://www.salmonellablog.com/12_cantaloupe.jpg |
| 19:07.06 | carrar | pop that whole slice in there |
| 19:07.08 | nny | leifmadsen: unless there is a way to use Originate from the dialplan |
| 19:07.09 | carrar | mmmmm |
| 19:07.14 | carrar | and supah healthy |
| 19:07.30 | leifmadsen | nny: there is -- use Originate() (available in 1.6.2.0 and later) |
| 19:07.53 | carrar | You'll have titanium resistance to sickness! :) |
| 19:08.01 | nny | leifmadsen: still on 1.4, mainly due to some needs to rewrite macros into gotos, hmm i'll have to consider both options |
| 19:08.15 | nny | leifmadsen: slack i know |
| 19:08.19 | leifmadsen | nny: Macro() still exists in 1.6.2.0 |
| 19:08.27 | leifmadsen | I have absolutely not idea why people think it isn't |
| 19:08.33 | leifmadsen | s/not/no |
| 19:08.47 | leifmadsen | it's not recommended, but it still exists |
| 19:09.13 | nny | leifmadsen: gotcha, dunno figured it was more principle of using it than.. yeah. Anything I should be weary of before compiling 1.6 over 1.4? |
| 19:10.07 | leifmadsen | nny: never do that? |
| 19:10.15 | leifmadsen | nny: test servers are a must |
| 19:10.25 | nny | leifmadsen: yeah this is kind of a test server :D |
| 19:10.36 | leifmadsen | I would never recommend someone just upgrading a production server |
| 19:10.41 | carrar | It's Asterisk!! Pop that new release right into production! :) |
| 19:10.43 | nny | leifmadsen: yeah no i hear you |
| 19:10.54 | leifmadsen | nny: then go nuts :) |
| 19:10.57 | Katty | It's Asterisk!!! wrap it another layer of foil and ROLL IT DOWN A HILL! |
| 19:11.29 | leifmadsen | It's Log! It's Log! It's better than bad... it's good! |
| 19:11.44 | carrar | Please send PICS of Asterisk wrapped in foil |
| 19:12.20 | ManxPower-work | nny: UPGRADE*.txt |
| 19:12.24 | ManxPower-work | 8-) |
| 19:12.27 | nny | ManxPower-work: ty |
| 19:12.47 | ManxPower-work | nny: (In theory) it contains all major changes |
| 19:13.19 | Katty | carrar: http://www.owensworld.com/funnyimages/files/foil_big.jpg |
| 19:13.32 | carrar | hahah |
| 19:15.04 | carrar | Think of how much less lighting you would need |
| 19:15.33 | nny | carrar: http://i.imgur.com/bmbhhl.jpg |
| 19:16.25 | *** join/#asterisk rocksfrow (n=kyle@74-92-146-97-WashingtonDC.hfc.comcastbusiness.net) |
| 19:17.12 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:17.23 | rocksfrow | how would i go about sending a specific inbound phone number from a block directly to a specifc extension |
| 19:17.45 | rocksfrow | i currently have two trunks, one with 14 channels (everything but fax), and one with 1 channel (fax) |
| 19:17.45 | ManxPower-work | exten => 2125551212,1,Goto(context,extension,priority) |
| 19:17.57 | ManxPower-work | Now go read the Asterisk book. |
| 19:17.58 | ManxPower-work | ~book |
| 19:17.59 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 19:18.03 | nny | ~movie |
| 19:18.04 | infobot | from memory, movie is http://www.wakeworld.com/VideoGuide/getvideo.asp?ProductID=100128 |
| 19:18.08 | nny | hahaha |
| 19:18.20 | nny | wow was being a smart ass |
| 19:18.21 | nny | :\ |
| 19:18.25 | rocksfrow | lol |
| 19:20.36 | *** join/#asterisk TSM2 (n=the_soft@87-194-32-212.bethere.co.uk) |
| 19:23.25 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 19:25.28 | nny | score! segfault :D |
| 19:25.51 | chuckf | http://i.imgur.com/bmbhhl.jpg |
| 19:26.17 | nny | chuckf: you like my 13337 gimp skillz? |
| 19:26.27 | chuckf | they are lovely |
| 19:26.55 | *** join/#asterisk titter (n=titter@c-76-101-240-142.hsd1.fl.comcast.net) |
| 19:29.39 | *** join/#asterisk Victor_Yure_ (n=victor@unaffiliated/victoryure/x-837844) |
| 19:29.44 | *** join/#asterisk vgster (n=vgster@94-194-190-189.zone8.bethere.co.uk) |
| 19:30.40 | Katty | :< the ups guy scared away all the critters :<<< |
| 19:30.54 | Katty | i will sue him for DISTURBING THE PEACE |
| 19:30.55 | nny | segfault due to stupidity, someone needs to file a bug report on that |
| 19:31.10 | Katty | sue it |
| 19:31.12 | Katty | SUE SEGFAULT |
| 19:31.34 | nny | [bug: 420247 - User must compile addons when upgrading, L2read] |
| 19:32.29 | titter | Anyone know if app_rxfax works successfully with 1.6.x and DAHDI? |
| 19:35.58 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 19:36.00 | nny | so far so good with 1.6, been meaning to test the various apps (openfire, fop) with 1.6 anyways |
| 19:38.00 | titter | Do I need to compile agx-ast-addons still? I saw there is a DAHDI dir in the svn now |
| 19:38.14 | jameswf | so whats the latest koolaid... |
| 19:38.37 | ManxPower-work | titter: 1.6 comes with a fax app |
| 19:41.10 | mbthv9467 | blah |
| 19:42.09 | titter | Does it support DAHDI? |
| 19:42.52 | rocksfrow | why would the DID be coming through blank? |
| 19:43.05 | rocksfrow | Entering from-zaptel with DID == " |
| 19:43.28 | ManxPower-work | rocksfrow: because you are using FreePBX, not Asterisk. |
| 19:43.40 | ManxPower-work | Try asking on the ~freepbx |
| 19:43.44 | rocksfrow | ... |
| 19:43.46 | rocksfrow | not asterisk? |
| 19:43.50 | ManxPower-work | ~freepbx |
| 19:43.51 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 19:43.53 | rocksfrow | freepbx is the frontend FOR asterisk, no? |
| 19:43.59 | Kobaz | no |
| 19:44.05 | Kobaz | it's *a* front end |
| 19:44.13 | rocksfrow | well, sorry *a* |
| 19:44.19 | rocksfrow | he said you're using freepbx, not asterisk |
| 19:44.22 | rocksfrow | that's not the case? |
| 19:44.25 | rocksfrow | i'm using both, yes? |
| 19:44.35 | rocksfrow | no problem, will go to #freepbx just want to be sure i'm not confused here. |
| 19:44.42 | rocksfrow | but i am indeed using asterisk, lol. |
| 19:44.45 | Kobaz | freepbx is a wrapper around a bunch of asterisk stuff |
| 19:44.52 | ManxPower-work | rocksfrow: Technically you are using an Asterisk that was "reprogrammed" via config files for FreePBX |
| 19:45.14 | Kobaz | it has a whole bunch of custom dialplan that makes supporting people who say they use asterisk, very difficult for us |
| 19:45.16 | rocksfrow | ManxPower-work, so the asteriskNOW distro, isn't asterisk? |
| 19:45.19 | ManxPower-work | So pretty much nothing we tell you to solve your problem and most any config file and setting we might tell you will be WRONG for your setup. |
| 19:45.21 | Kobaz | since hardly anyone uses freepbx here |
| 19:45.32 | rocksfrow | ah, okay. |
| 19:45.33 | rocksfrow | thx. |
| 19:46.09 | Kobaz | there's a whole bunch of frontends for asterisk, freepbx is one of them |
| 19:46.17 | Kobaz | personally i recommend just going with straight asterisk |
| 19:46.38 | rocksfrow | with no front end/ |
| 19:46.45 | Kobaz | nope |
| 19:47.10 | Kobaz | frontends are only good for doing what the frontend is programmed to allow you to do |
| 19:47.12 | rocksfrow | how do you expect anybody who isn't a pbx expert to manage it? |
| 19:47.13 | titter | no need for a front end, the conf files are pretty simple once you gain understanding of them ... plus you have more control. |
| 19:47.54 | Kobaz | rocksfrow: you don't have to be a pbx expert, or an asterisk expert, you just need to know what you need to build what you want |
| 19:48.21 | titter | ManxPower-work: do you know if it supports DAHDI now? |
| 19:48.25 | Kobaz | er. know what you need to know... |
| 19:52.14 | Kobaz | rocksfrow: if all you want to do is set up a half dozen phones, with some voicemail and outside/inside dialing, then by all means use a frontend |
| 19:52.46 | Kobaz | rocksfrow: if you want to build custom apps or add functionality other than that, you'll need to get to asterisk itself |
| 19:53.41 | rocksfrow | kobaz: im very interested in what you're saying because i do plan on implementing custom code to incorporate our internal database within the customer service system |
| 19:53.50 | rocksfrow | but, i cans till dot hat with a freepbx system, no? |
| 19:53.56 | Kobaz | you can |
| 19:53.58 | rocksfrow | having freepbx, doesn't stop me from doing separate coding? |
| 19:54.01 | rocksfrow | right.. |
| 19:54.12 | [TK]D-Fender | rocksfrow: Good luck with that... LOTS of work, and you'll not only have to learn *, but also FreePBX. |
| 19:54.13 | Kobaz | you'll need to learn freepbx's ways of doing things, in order to maintain use of the frontend |
| 19:54.20 | Kobaz | exactly |
| 19:54.20 | rocksfrow | #freepbx is not very busy :-/ |
| 19:54.58 | rocksfrow | Set("Zap/11-1", "DID=s") in new stack |
| 19:55.09 | rocksfrow | i just don't understand what is and why setting the DID to s |
| 19:55.10 | rocksfrow | lol |
| 19:55.13 | *** join/#asterisk JAMMAN2110 (n=james@unaffiliated/jamman2110) |
| 19:55.15 | Kobaz | ~book |
| 19:55.16 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 19:55.18 | [TK]D-Fender | rocksfrow: And you've continuously failed to specify what its coming IN on. |
| 19:55.20 | rocksfrow | :-p |
| 19:55.21 | rocksfrow | yes book.. |
| 19:55.38 | Kobaz | Zap is a tdm/analog subsystem |
| 19:55.39 | rocksfrow | it's coming IN on a zaptel trunk |
| 19:55.47 | [TK]D-Fender | rocksfrow: .what KIND? |
| 19:55.55 | Kobaz | s is the 'default' extension, in asterisk |
| 19:56.08 | [TK]D-Fender | not quite |
| 19:56.15 | rocksfrow | default extension? |
| 19:56.20 | Kobaz | well, not default, but |
| 19:56.21 | rocksfrow | i'm calling from an outside number |
| 19:56.22 | [TK]D-Fender | rocksfrow: No. Now answer the questio |
| 19:56.24 | [TK]D-Fender | n |
| 19:56.30 | nny | leifmadsen: trying out Originate now, is there any examples of how to use it with Bridge for a call transfer like I mentioned? |
| 19:56.34 | rocksfrow | i would expect DID to be set to the dialed # |
| 19:56.34 | Kobaz | which is why i quoted it |
| 19:56.40 | Kobaz | rocksfrow: depends on the setup |
| 19:56.49 | Kobaz | rocksfrow: and, we don't know your set, and we're not familiar with freepbx |
| 19:56.53 | rocksfrow | [TK]D-Fender, i'm trying...it's a t1 |
| 19:56.59 | [TK]D-Fender | rocksfrow: What signalling... |
| 19:57.20 | leifmadsen | nny: likely not -- this is an edge case feature request that you'll have to be clever in solving |
| 19:57.28 | rocksfrow | man i need to call back the phone company, i just called them trying to get information about the connection |
| 19:57.36 | Kobaz | nny: you can use Bridge() |
| 19:57.38 | rocksfrow | and all he would say is its a voip..and he said its a t1 but with more lines? |
| 19:57.46 | [TK]D-Fender | rocksfrow: You've already configured your system for it and you don;'t even know what you have? |
| 19:57.52 | rocksfrow | he wouldnt give me any technical details, obviously he didnt know anymore than i do :-p |
| 19:57.52 | Kobaz | hehe |
| 19:58.05 | [TK]D-Fender | rocksfrow: Do you randomly fill up rental cars with deisel and hope they all work too? |
| 19:58.07 | Kobaz | rocksfrow: i think you need to hire some support |
| 19:58.11 | rocksfrow | [TK]D-Fender, this system was set up by a previous employee. |
| 19:58.19 | [TK]D-Fender | MY FAVOURITE STORY |
| 19:58.19 | rocksfrow | [TK]D-Fender, are you serious? lol |
| 19:58.22 | Kobaz | ~book |
| 19:58.23 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 19:58.29 | Kobaz | it's *free* |
| 19:58.34 | Kobaz | go read it, and then come back :) |
| 19:58.34 | rocksfrow | dude, i ordered the book man |
| 19:58.42 | [TK]D-Fender | rocksfrow: don't ask for help when you don't even know what you're fixing <- |
| 19:58.51 | rocksfrow | .... |
| 19:58.51 | nny | leifmadsen: hear ya, in a nut shell would it be Originate - and use the viariable "Â ${ORIGINATE_STATUS}" to perform either a Bridge() or do nothing? |
| 19:59.03 | Kobaz | rocksfrow: get the pdf online, while you wait for your hard copy to arrive in the mail |
| 19:59.07 | Kobaz | rocksfrow: it will answer a lot of questions |
| 19:59.27 | [TK]D-Fender | rocksfrow: [14:58]<rocksfrow>dude, i ordered the book man |
| 19:59.30 | rocksfrow | [TK]D-Fender, the system is currently working |
| 19:59.32 | Kobaz | nny: no, you'll need to Originate you call to the Bridge() application |
| 19:59.36 | leifmadsen | nny: you'll have to play around -- it may not even end up being those applications, but that's what I would have started with in my initial thinking -- I've never implemented that, and don't have time to lab anything up right now |
| 19:59.39 | [TK]D-Fender | rocksfrow: http://www.angelfire.com/crazy2/hear_sheepdog/ |
| 19:59.46 | nny | Kobaz: gotcha |
| 19:59.51 | rocksfrow | i'm just tyring to add a STATIC route to an extension for a specific number in the block of numbers allocated |
| 19:59.55 | [TK]D-Fender | rocksfrow: pastebin your zapata.conf and EVERYTHING it links to |
| 20:00.02 | Kobaz | nny: at least, i think that's what you want to to... but you didn't really explain the situation |
| 20:00.23 | nny | Kobaz: bascially I am looking to transfer a call, but wait until the other line picks up |
| 20:00.25 | Kobaz | rocksfrow: do you want freepbx help or asterisk help |
| 20:00.36 | Kobaz | rocksfrow: because 'static route' isnt not an asterisk term |
| 20:00.47 | Kobaz | rocksfrow: we can help you with asterisk... but you gotta pick your fancy |
| 20:00.51 | rocksfrow | neither is a freepbx term (??), i'm just saying |
| 20:00.52 | Kobaz | it's one or the other |
| 20:00.55 | Kobaz | k |
| 20:00.57 | rocksfrow | it's a static route, thats english language, lol. |
| 20:00.58 | rocksfrow | no terms. |
| 20:01.13 | titter | hes waiting to route a call based on the incoming did I assume |
| 20:01.18 | titter | wants* |
| 20:01.26 | rocksfrow | yes quite common i would assume |
| 20:01.30 | rocksfrow | i have 14 numbers you know |
| 20:01.31 | Kobaz | which is pretty easy |
| 20:01.36 | titter | In * it's easy |
| 20:01.36 | rocksfrow | xx10 - xx24 |
| 20:01.37 | Kobaz | but you need to give more details |
| 20:01.38 | [TK]D-Fender | rocksfrow: "static route" is a networking term. Go plug your T1 into your DSL modems RJ-45 and it'll get "static" |
| 20:01.46 | rocksfrow | i want xx20 to go to extension 5000, and bypass the IVR |
| 20:01.55 | rocksfrow | [TK]D-Fender, lol, smartass. |
| 20:01.57 | Kobaz | heh |
| 20:01.59 | [TK]D-Fender | rocksfrow: And I'm still waiting for my pastebin... |
| 20:02.01 | [TK]D-Fender | ~pb |
| 20:02.02 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
| 20:02.03 | [TK]D-Fender | ^^^^^^^^^^^^^^^ |
| 20:02.04 | Kobaz | details! |
| 20:02.08 | Kobaz | he wants his details |
| 20:02.10 | ManxPower-work | rocksfrow: then use exten => _XX20,1,Goto(context,5000,1) |
| 20:02.15 | Kobaz | fender needs his fix |
| 20:02.16 | [TK]D-Fender | ManxPower-work: LOL |
| 20:02.37 | ManxPower-work | [TK]D-Fender: he's asking on #Asterisk, I will provide Asterisk answers. |
| 20:02.40 | rocksfrow | ManxPower-work, i figured out how to add the actual rule, but when watching debug during the call, the did is getting set to 's', not xxxx |
| 20:02.49 | Kobaz | paaassstteeeeee |
| 20:02.51 | ManxPower-work | rocksfrow: that's FreePBX doing that, not Asterisk |
| 20:02.52 | rocksfrow | yes asterisk answers are fine |
| 20:02.56 | rocksfrow | working on it working on it |
| 20:02.57 | [TK]D-Fender | rocksfrow: And you're still not providing the information requested |
| 20:03.06 | rocksfrow | ManxPower-work, asterisk -r |
| 20:03.11 | rocksfrow | is where i see it |
| 20:03.16 | rocksfrow | [TK]D-Fender, working on it` |
| 20:03.19 | rocksfrow | doing it now! |
| 20:03.20 | ManxPower-work | rocksfrow: correct. FreePBX wrote that config file. |
| 20:03.31 | ManxPower-work | rocksfrow: Did you ever use Windows 95 or Windows 98? |
| 20:03.38 | rocksfrow | unfortunately |
| 20:03.42 | [TK]D-Fender | rocksfrow: then stop talking until you've got it |
| 20:03.54 | [TK]D-Fender | rocksfrow: CONCENTRATE |
| 20:03.59 | Kobaz | use the force |
| 20:03.59 | ManxPower-work | rocksfrow: so everything you know about DOS should apply to Windows98, right? It runs on top of DOS (..er..Asterisk) afterall |
| 20:04.13 | rocksfrow | ManxPower-work, you makek no sense what so ever. |
| 20:04.21 | Kobaz | i think he makes sense |
| 20:04.25 | ManxPower-work | rocksfrow: You'll understand soon enough. |
| 20:04.27 | rocksfrow | i understand the concept of a front end and back end |
| 20:04.30 | titter | ManxPower-work: the fax app in 1.6 supports dahdi now? just wondering before I test this |
| 20:04.30 | Kobaz | he's comparing freepbx to asterisk, like dos to windows |
| 20:04.31 | rocksfrow | damnit let me paste |
| 20:04.39 | Kobaz | windows 3.1 is a gui for dos |
| 20:04.58 | rocksfrow | i meant the all-around point |
| 20:04.59 | rocksfrow | nvm. |
| 20:05.01 | rocksfrow | geesh |
| 20:05.02 | Kobaz | heh |
| 20:05.06 | ManxPower-work | titter: your question makes NO sense. fax isn't going to work over SIP. What else would it be running on other than DAHDI/Zaptel? |
| 20:05.41 | Kobaz | ManxPower-work: i've never worked with fax really, but i've sent a fax over vonage... isn't that sip? |
| 20:06.16 | titter | ManxPower-work: it does make sense considering there are people running 1.4 with dahdi which wasn't supported with agx-ast-addons in 1.4 |
| 20:06.34 | Kobaz | goes back to breaking stuff |
| 20:06.54 | titter | http://sourceforge.net/projects/agx-ast-addons/ |
| 20:07.12 | rocksfrow | [TK]D-Fender, http://pastebin.com/m74e8482 |
| 20:07.44 | *** join/#asterisk tzafrir (n=tzafrir@212.179.75.202) |
| 20:07.52 | Kobaz | watches [TK]D-Fender's eyes go wild from the details |
| 20:07.53 | titter | I am going to order a PRI, and setup a dedicated fax to e-mail server ... before I sign this PRI contract, and setup tons of 8xx numbers, want to make sure this works ... so I am going to test it on my PRI now with a free DID and I will let you know. I just didn't have 1.6 server in front of me. |
| 20:07.57 | nny | Kobaz: you mind if I pb you my logic for Originate once I think I figured it out? |
| 20:08.04 | Kobaz | nny: sure |
| 20:08.07 | [TK]D-Fender | rocksfrow: signalling=fxs_ks <- doesn't support DID's |
| 20:08.11 | nny | Kobaz: thanks |
| 20:08.13 | [TK]D-Fender | rocksfrow: signalling=fxs_ks <-- should be NO |
| 20:08.27 | rocksfrow | should be NO? |
| 20:08.33 | [TK]D-Fender | rocksfrow: Guess you should stop using CAS signalling on digital links |
| 20:08.38 | Kobaz | heh |
| 20:08.39 | rocksfrow | thanks for the help man, really. |
| 20:08.48 | [TK]D-Fender | rocksfrow: immediate=yes ; alt = no |
| 20:09.03 | [TK]D-Fender | rocksfrow: ^^ NO |
| 20:09.11 | Kobaz | rocksfrow: he knows his stuff... just be patient |
| 20:09.15 | [TK]D-Fender | rocksfrow: Fix this last one, restart *, retest |
| 20:09.24 | rocksfrow | so ony the immediate? |
| 20:09.25 | rocksfrow | update yes to no? |
| 20:09.28 | Kobaz | yes |
| 20:09.28 | *** join/#asterisk _cgc (n=_cgc@94-194-207-211.zone8.bethere.co.uk) |
| 20:09.29 | rocksfrow | what did you saya boutt he first? |
| 20:09.31 | Kobaz | go go go |
| 20:09.34 | rocksfrow | don't touch fxs_ks? |
| 20:09.36 | [TK]D-Fender | rocksfrow: What card do you have? |
| 20:09.46 | ManxPower-work | Talk slow, he's a GUI user. |
| 20:09.50 | Kobaz | hahah |
| 20:09.53 | rocksfrow | lol |
| 20:09.57 | [TK]D-Fender | rocksfrow: fxs signalling does not typically support transmitting a DID |
| 20:10.00 | rocksfrow | ManxPower-work, i'm a gui user with shit i dont know about, sure. |
| 20:10.11 | Kobaz | dahdi_hardware |
| 20:10.16 | rocksfrow | interesting.. |
| 20:10.22 | [TK]D-Fender | ManxPower-work: Lets jsut stop at "non-telco informed" shall we... |
| 20:10.35 | rocksfrow | i'll take that, sure. lol |
| 20:10.37 | ManxPower-work | rocksfrow: just remember virtually none of the asterisk documentation will apply to your setup. |
| 20:10.37 | Kobaz | rocksfrow: you really want PRI |
| 20:10.41 | [TK]D-Fender | rocksfrow: what card? |
| 20:10.48 | Kobaz | rocksfrow: type dahdi_hardware |
| 20:11.05 | rocksfrow | natta |
| 20:11.10 | Kobaz | what? |
| 20:11.10 | [TK]D-Fender | ; changed to kewl start from loop start, since Cavtel wasn't providing disconnect supervision <-- sure as shit looks lok dumb analog commentary to me |
| 20:11.19 | [TK]D-Fender | rocksfrow: what card? |
| 20:11.33 | rocksfrow | [TK]D-Fender, trying to find out now.. |
| 20:11.39 | Kobaz | rocksfrow: quit out of the asterisk shell, and type |
| 20:11.39 | Kobaz | oh |
| 20:11.41 | Kobaz | it's zaptel |
| 20:11.49 | [TK]D-Fender | "this PCB is best viewed with... YOUR EYES" |
| 20:12.06 | Kobaz | is it zaptel_hardware for zap drivers? |
| 20:12.10 | Kobaz | i forget, it's beena while |
| 20:12.40 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 20:12.48 | Kobaz | nope, zaptel doesn't have that |
| 20:12.56 | Kobaz | type lspci, and look for your telephony card |
| 20:13.29 | tzafrir | Kobaz, actually latest versions of zaptel have it. But even with 1.4.12.1 it may still be unreliable |
| 20:13.38 | rocksfrow | yay |
| 20:13.39 | rocksfrow | 02:01.0 Communication controller: Digium, Inc. Wildcard TE205P (rev 02) |
| 20:13.41 | Kobaz | good |
| 20:13.57 | rocksfrow | [TK]D-Fender, <rocksfrow> 02:01.0 Communication controller: Digium, Inc. Wildcard TE205P (rev 02) |
| 20:14.06 | [TK]D-Fender | rocksfrow: Ask your telco to cut you over to PRI signalling |
| 20:14.13 | Kobaz | so, you are using analog signalling on a digital link |
| 20:14.23 | [TK]D-Fender | Kobaz: Channel banks do taht, and CAS |
| 20:14.35 | [TK]D-Fender | Kobaz: Stupid normally |
| 20:14.42 | Kobaz | yeah, but it'll be nicer for him to use pri, i agree |
| 20:14.48 | rocksfrow | the guy from telco told me i had a digital link |
| 20:14.52 | Kobaz | you do |
| 20:14.53 | Kobaz | it's t1 |
| 20:15.29 | rocksfrow | what sort of changes would be required after switching to pri signalling? |
| 20:15.31 | Kobaz | rocksfrow: and, before you go and break everything |
| 20:15.37 | [TK]D-Fender | rocksfrow: Yippy-kai-yay. No ask about having them switch the signalling to PRI |
| 20:15.46 | [TK]D-Fender | now* |
| 20:16.07 | Kobaz | rocksfrow: install asterisk on a new box, and experiement with that, so you still have a working phone system while you play |
| 20:16.08 | rocksfrow | i'm wondering why we're not already |
| 20:16.12 | rocksfrow | are there any drawbacks? |
| 20:16.21 | rocksfrow | kobaz: totally the plan |
| 20:16.33 | [TK]D-Fender | rocksfrow: Yes, you'd have missed out on all of this fun we've had because of it |
| 20:16.38 | rocksfrow | Kobaz, fortunately, the current system is not _that_ important |
| 20:16.40 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:16.45 | Kobaz | that's good |
| 20:16.50 | [TK]D-Fender | rocksfrow: why not? |
| 20:16.58 | rocksfrow | but we are going to be setting up a customer service system, so thats why i'm trying to get involved with this. |
| 20:17.07 | Kobaz | rocksfrow: no, no drawbacks, pri is generally what people use these days for t1 phone service |
| 20:17.14 | drmessano | [TK]D-Fender: They're a tech startup.. they may not need it in a week |
| 20:17.20 | rocksfrow | [TK]D-Fender, I meant as far as needing 100% uptime. its currently only internal office phones.. |
| 20:17.33 | [TK]D-Fender | Waiter: CHECK PLEASE! |
| 20:17.37 | Kobaz | hehe |
| 20:17.40 | rocksfrow | but i must say |
| 20:17.43 | drmessano | lol |
| 20:17.47 | rocksfrow | asterisk has ran for 2 yrs |
| 20:17.51 | rocksfrow | without me touching it basically |
| 20:17.54 | Kobaz | yeap |
| 20:17.56 | Kobaz | that's what it should do |
| 20:17.57 | rocksfrow | since the last guy left |
| 20:17.57 | rocksfrow | lol |
| 20:18.09 | rocksfrow | drmessano: what? |
| 20:18.11 | rocksfrow | a tech startup? |
| 20:18.11 | Kobaz | unless you have one of those new fangled crashy versions |
| 20:18.21 | Kobaz | but yeah, asterisk should be able to run unattended |
| 20:18.46 | rocksfrow | drmessano: much more than tech startup- sir. |
| 20:18.52 | rocksfrow | very judgemental, damn..lol |
| 20:18.57 | Kobaz | rocksfrow: go peruse the online asterisk book pdf |
| 20:18.59 | drmessano | socksfrown: It was a JOKE |
| 20:19.05 | nny | lol |
| 20:19.05 | Kobaz | rocksfrow: it really will answer a lot of questions |
| 20:19.32 | rocksfrow | Kobaz, yes will do. |
| 20:19.38 | rocksfrow | thx guys for your help |
| 20:19.43 | Kobaz | thank you, come again |
| 20:20.09 | rocksfrow | Kobaz, so switching to pri doesnt require a huge amount of reconfiguration? |
| 20:20.21 | Kobaz | rocksfrow: one line, in the config file you just pasted |
| 20:20.25 | Kobaz | no, not a whole lot |
| 20:20.27 | rocksfrow | you think i should get a completely separate, second t1 for the new setup? |
| 20:20.52 | drmessano | I would also get a copy of "IRC for Dummies", "The Notebook", and "The National Geographic Field Guide to North American Squirrel Nuts" if you plan to come back |
| 20:20.56 | Kobaz | rocksfrow: if your phone system is is unimportant enough where it can be down for a week or two, while you putz with it |
| 20:21.16 | Kobaz | then you dont need another t1 |
| 20:21.48 | rocksfrow | drmessano, cute |
| 20:21.50 | Kobaz | otherwise, get yourself one of these: http://www.vconsole.com/client/ |
| 20:22.10 | jaytee | and "The National Geographic Field Guide to North American Squirrel Nuts" PMSL |
| 20:22.21 | Kobaz | er, one of these: http://www.vconsole.com/2-Port-T1/E1/PRI-%2B-4-Port-FXS-PSTN-Simulator-p-26.html |
| 20:22.49 | rocksfrow | oh sweet |
| 20:23.06 | rocksfrow | one last q |
| 20:23.13 | Kobaz | if you plan on doing lots of asterisk, for clients |
| 20:23.17 | Kobaz | definitly get one of those |
| 20:24.02 | drmessano | Does someone make an Asterisk technician simulator.. That would be far more useful |
| 20:24.11 | jaytee | :-) |
| 20:25.26 | rocksfrow | yeah that would be sweet too |
| 20:32.53 | nny | Kobaz: hmm heh. am I on the right path? http://pastebin.com/mefa1678 |
| 20:33.30 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 20:33.46 | Kobaz | well |
| 20:33.51 | Kobaz | first you should explain what you're trying to do |
| 20:34.05 | *** join/#asterisk Defraz (n=Defraz@corp.fuzecore.com) |
| 20:34.12 | Kobaz | but, that looks like it'll do what i think you want to do |
| 20:34.23 | Kobaz | but we need to make sure what i think you need, and what you actually need... match up |
| 20:34.38 | Kobaz | you're missing some arguments |
| 20:35.15 | nny | Kobaz: for the test I am just testing a transfer to the number 8432980648 (I am calling another number and hitting the macro via *9 from features.conf) |
| 20:35.32 | drmessano | But what if he thinks that you think that he thinks what you're thinking is not really what he thinks you think he is thinking? |
| 20:37.18 | Kobaz | if you think that's thinking, you better think again |
| 20:37.26 | nny | i think |
| 20:37.58 | Kobaz | er |
| 20:38.02 | nny | Kobaz: yeah I need to define the channel |
| 20:38.11 | nny | Kobaz: that's probably going to be a variable |
| 20:38.13 | Kobaz | you want to transfer an existing call to somewhere else? |
| 20:38.18 | nny | well kind of |
| 20:38.27 | nny | without the caller hearing the xfer |
| 20:38.27 | Kobaz | by hitting a * code while the call is in progress? |
| 20:38.34 | nny | yes |
| 20:38.39 | Kobaz | okay so |
| 20:38.47 | Kobaz | you should have said that the last time i asked you to explain what you're doing |
| 20:38.50 | nny | but want to connect the transfered number first |
| 20:38.51 | Kobaz | what you pasted won't work |
| 20:38.51 | nny | heh sorry |
| 20:38.57 | Kobaz | oh |
| 20:38.59 | Kobaz | okay |
| 20:39.02 | Kobaz | that might work then |
| 20:39.19 | Kobaz | so you have callers, A and B, and outside destination C |
| 20:39.26 | nny | correct |
| 20:39.28 | Kobaz | you want to call C, and then link it to B |
| 20:39.33 | nny | yes |
| 20:39.36 | Kobaz | and A will hang up |
| 20:39.39 | nny | yes |
| 20:39.49 | Kobaz | okay, the originate to bridge will work then |
| 20:40.14 | nny | hmm |
| 20:40.25 | nny | features.conf asks which side of the channel to originate the command |
| 20:40.27 | Kobaz | you need to pass bridge, the channel that you want to hijack |
| 20:40.32 | nny | yeah |
| 20:41.01 | Kobaz | the side config in features.conf, is which side will this command run on |
| 20:41.02 | nny | ${CHANNEL} ? |
| 20:41.06 | Kobaz | caller, and callee |
| 20:41.26 | nny | yeah but if I wanted the ability to do both depending on if the person dialed out or was dialed to it may prove tricky no? |
| 20:41.31 | Kobaz | in your case, it might not matter which side runs it, since originate doesn't pass any audio to the executor |
| 20:41.35 | nny | ahh ok |
| 20:41.59 | Kobaz | what you need to be able to control, is who has access to it |
| 20:42.07 | Kobaz | and that's with DYNAMIC_FEATURES |
| 20:42.18 | nny | yeah I have that set in my logic for the test |
| 20:42.28 | Kobaz | okay so... what's your current issue |
| 20:43.14 | nny | just trying to figure out the proper way to pass the channel variable to bridge, so that it always affetcs "B"'s channel |
| 20:43.25 | Kobaz | ah yes |
| 20:43.34 | Kobaz | so it will matter which side it runs on |
| 20:43.37 | nny | yeah |
| 20:43.41 | Kobaz | does A always call B |
| 20:43.51 | nny | for the test yes, in theory for everyday use no |
| 20:43.53 | Kobaz | ie... is it always an outgoing call... or always an incomming call |
| 20:44.03 | Kobaz | that gets pretty tricky |
| 20:44.06 | nny | yeah heh |
| 20:44.31 | Kobaz | so, in your dialplan |
| 20:44.41 | Kobaz | you'll need to enable a different feature, depending on the direction |
| 20:44.49 | Kobaz | you'll need one *9 code, for caller, and one *9 code for callee |
| 20:44.56 | nny | ahhh |
| 20:45.01 | nny | ok |
| 20:45.05 | Kobaz | and in dialplan, you need to figure out which one to enable, depending on the direction of the call |
| 20:45.12 | nny | so I can use *9 for both, and just Set the one I need for the direction |
| 20:45.15 | nny | nice |
| 20:45.21 | Kobaz | yes you can, but you need to give them different names |
| 20:45.33 | nny | i <3 asterisk heh |
| 20:45.34 | Kobaz | so call i, callee_starnine and caller_starnine |
| 20:45.35 | Kobaz | or whatever |
| 20:45.37 | nny | ok working on it now |
| 20:45.42 | Kobaz | hah yeah, asterisk is pretty damn flexible |
| 20:46.29 | Kobaz | sooo |
| 20:46.43 | Kobaz | if i remember right, you can't pass variables from features.conf |
| 20:46.52 | Kobaz | so you'll need a GoSub |
| 20:47.02 | nny | using a macro atm for the command |
| 20:47.09 | nny | but that's improper afaik |
| 20:47.31 | nny | dialout_cellswitch => *9,callee,Macro,cellswitch |
| 20:47.38 | Kobaz | callee_starnine => *9,callee,GoSub,foo,s,1(*9) |
| 20:47.43 | Kobaz | macro works too |
| 20:47.48 | Kobaz | gosub is the new method |
| 20:47.49 | nny | well |
| 20:47.52 | nny | I should not use macros |
| 20:47.55 | nny | so I will dot hat |
| 20:47.57 | nny | do that* |
| 20:48.02 | nny | need to retrain brain very soon |
| 20:48.14 | Kobaz | so now... it will run on the callee's channel |
| 20:48.35 | Kobaz | so, in that dialplan sub.. ${CHANNEL} will be the callee's channel, which you can pass to Bridge |
| 20:48.47 | Kobaz | well, which you can pass to originate, which will then pass to bridge |
| 20:49.59 | Kobaz | you can do all kinds of funky stuff with asterisk, you just need to kind of put the various pieces together |
| 20:50.08 | nny | yeah |
| 20:52.19 | Kobaz | post orfice time |
| 20:52.23 | Kobaz | good luck |
| 20:52.27 | nny | thanks |
| 20:54.31 | nny | well it works except for one small part, the Originate seems to mute the channels while it is happening, is this something that can be worked around? |
| 20:55.52 | nny | http://pastebin.com/m752ca7b is results right now |
| 20:56.55 | *** join/#asterisk Badrobot- (n=badrobot@cpe-76-173-229-89.socal.res.rr.com) |
| 21:01.15 | *** join/#asterisk voipmonk (n=shido6@dsl-67-204-37-228.acanac.net) |
| 21:02.59 | nny | I assume Originate isn't "muting" channels so much as taking over the channel long enought to perform the operation, so this may not work either for what I am trying to do. any imput appreciated |
| 21:04.27 | voipmonk | its a girl |
| 21:04.42 | nny | ? |
| 21:04.50 | nny | more voipmonks? |
| 21:05.15 | voipmonk | :) |
| 21:05.32 | voipmonk | shopping for 4d imaging places |
| 21:05.43 | voipmonk | got the sonogram pics and movie |
| 21:05.49 | nny | nice gratz! |
| 21:06.00 | voipmonk | =) |
| 21:06.55 | nny | is trying to figure out if Originate is supposed to supress the channel as it works |
| 21:07.07 | voipmonk | supress? |
| 21:07.12 | voipmonk | its in use |
| 21:07.14 | nny | http://pastebin.com/m752ca7b |
| 21:07.18 | nny | doing: |
| 21:07.33 | nny | exten => s,n,Originate(SIP/8432980648@vitel-outbound,app,Bridge,${CHANNEL}) |
| 21:07.45 | nny | goal: seamless transfer to a cell phone |
| 21:08.49 | *** join/#asterisk af_ (n=getsmart@88-149-241-228.dynamic.ngi.it) |
| 21:09.24 | nny | was trying to do it in such a way that it didn't bridge the two channels until the 3rd was answered, but once I iniate it the first two channels can't communicate |
| 21:11.14 | *** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd) |
| 21:11.14 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 21:11.37 | *** join/#asterisk lynxsys (n=thelynx@82-71-19-61.dsl.in-addr.zen.co.uk) |
| 21:11.45 | *** join/#asterisk Rawplayer (n=kevin@wickedleaks.nl) |
| 21:11.58 | Rawplayer | hello, i'm looking for someone with experiance from the SIA protocol |
| 21:13.03 | ChannelZ | I like Sia but not as a protocol |
| 21:13.17 | fenrus_ | sia ice-cream? :) |
| 21:13.33 | ChannelZ | Sia the singer |
| 21:14.02 | Rawplayer | haters :< |
| 21:14.12 | Rawplayer | it's for alarm signalling |
| 21:14.16 | Rawplayer | burglar alarms |
| 21:14.30 | ChannelZ | Clearly you are lost |
| 21:14.32 | [TK]D-Fender | ChannelZ: http://www.bandweblogs.com/sia.jpg |
| 21:14.46 | Rawplayer | lost? |
| 21:14.48 | Rawplayer | from? |
| 21:14.48 | ChannelZ | TK: That's the one |
| 21:14.54 | nny | too bad she's into other girls |
| 21:14.59 | ChannelZ | yeah |
| 21:15.01 | [TK]D-Fender | ChannelZ: Ex had that album |
| 21:15.10 | Rawplayer | lol |
| 21:15.21 | ChannelZ | Lost as in alarms != asterisk |
| 21:15.36 | p3nguin | [tk]d-fender: I read that as your ex had that problem. |
| 21:15.44 | [TK]D-Fender | [16:14]<nny>too bad she's into other girls <- only too bad if its exclusive ;) |
| 21:15.59 | nny | [TK]D-Fender: http://www.volenet.cz/files/lesbians.jpg |
| 21:15.59 | [TK]D-Fender | p3nguin: lysdexics of the world untie! |
| 21:16.08 | p3nguin | exactly |
| 21:16.34 | ChannelZ | nny: haha nice |
| 21:16.42 | nny | :D |
| 21:19.05 | nny | so anyone familiar with Originate, Is there a way to keep the channels open while it attempts the call?? Sorry to be redundant, can't find any other info, the googles, they do nothing |
| 21:19.29 | [TK]D-Fender | nny: HUH? |
| 21:19.38 | [TK]D-Fender | nny: keep WHAT channels "open"? |
| 21:20.34 | nny | [TK]D-Fender: sorry recap: Trying to have A call B and transfer to C, but not until C answers. Using "s,n,Originate(SIP/8432980648@vitel-outbound,app,Bridge,${CHANNEL})" results are http://pastebin.com/m752ca7b |
| 21:20.51 | nny | [TK]D-Fender: the issue is when I perform that, A and B can no longer talk while C is dialed |
| 21:21.15 | nny | reading book over to find anymore info about originate and how it works |
| 21:21.49 | [TK]D-Fender | nny: Try doing a System + call file to test |
| 21:22.26 | rocksfrow | anybody have any links to some recommended hardware for asterisk servers? ..or any personal advice? |
| 21:22.33 | rocksfrow | i'm shopping for two servers, one to be used as a backup |
| 21:22.48 | nny | [TK]D-Fender: you mean just test with an arbitrary shell command instead of Bridge? |
| 21:22.55 | rocksfrow | its only for an system with ~40 phones |
| 21:23.05 | [TK]D-Fender | nny: Correct |
| 21:23.10 | nny | [TK]D-Fender: k |
| 21:23.23 | [TK]D-Fender | rocksfrow: For your needs any P4 should do |
| 21:23.35 | |Rain| | I really hope 1.6.2 is less deadlock-prone than 1.4 |
| 21:24.15 | rocksfrow | [TK]D-Fender, what about RAM? |
| 21:25.03 | [TK]D-Fender | rocksfrow: 256 minimum, 1G recommended |
| 21:25.29 | rocksfrow | [TK]D-Fender, any tips concerning redundancy? |
| 21:26.39 | [TK]D-Fender | rocksfrow: Buy another box. Sync often |
| 21:26.40 | drmessano | Dont buy a backup.. buy 1 good server.. |
| 21:26.58 | rocksfrow | drmessano, that's what i'm debating.. |
| 21:27.07 | drmessano | Mirrored drives, redundant power, 4 hour gold support, etc |
| 21:27.20 | *** join/#asterisk evil_gordita (n=evilgord@ip70-188-50-186.rn.hr.cox.net) |
| 21:27.20 | drmessano | Nothing fancy.. |
| 21:27.22 | rocksfrow | drmessano, right. |
| 21:28.04 | rocksfrow | when you guys are doing setups, do you mostly always have the voip servers on-site, or do you ever go datacentre route? |
| 21:28.33 | [TK]D-Fender | rocksfrow: on-site if you know what's good for you |
| 21:28.54 | rocksfrow | cool cool |
| 21:29.15 | [TK]D-Fender | checkout time, later all |
| 21:29.39 | rocksfrow | later, thanks again tk |
| 21:29.46 | drmessano | I still havent worked out the mentality of "My former PBX was a black meaningless box shoved in the janitors closet, but this asterisk box needs its own rack, in a datacenter, with $1500 a month worth of redundant connectivity |
| 21:30.03 | rocksfrow | drmessano, care for an explanation? |
| 21:30.26 | rocksfrow | it's quite simple, and relatively common situation. |
| 21:30.46 | rocksfrow | since the companies birth, they've been outsourcing all customer service / fullfillment. |
| 21:30.55 | drmessano | People tend to way overbuy for Asterisk |
| 21:31.08 | drmessano | or they go cheap as shit |
| 21:31.18 | rocksfrow | over the past 5 years the companys revenue has really grown for us to need customer service in house |
| 21:31.20 | drmessano | $10000 box or a $300 box.. |
| 21:31.28 | rocksfrow | heh |
| 21:31.36 | drmessano | Im serious |
| 21:31.37 | rocksfrow | we've been running our current system on an old dell desktop |
| 21:31.45 | nny | ugh |
| 21:31.50 | rocksfrow | lol |
| 21:32.10 | rocksfrow | drmessano, i'm thinking just a nice beefy desktop server, in the relay rack i already have |
| 21:32.17 | nny | ok so trying what D-fender suggested about using System in stead of bridge with Originate and the two channels sitll lose connectivity |
| 21:32.54 | rocksfrow | drmessano, do you recommend any specific hardware? |
| 21:33.18 | nny | er rather, the two channels can't communicate while originate is qualifying the number. is this "as designed" or is there a way to get run Originate in the background?? |
| 21:33.25 | drmessano | You need at minimum a server class box.. tower form factor is fine.. Mirrored drives, redundant power, and a way to hang it on the wall without violating the warranty |
| 21:34.29 | *** join/#asterisk catojo (n=catojo@189.24.66.25) |
| 21:35.02 | rocksfrow | thx |
| 21:36.43 | nny | Kobaz: if you pop back in nd feel like helping more lemme know. |
| 21:40.35 | *** join/#asterisk mrbnet (n=mrbnet@74-95-100-233-Minnesota.hfc.comcastbusiness.net) |
| 21:41.06 | Kobaz | nny: originate will end once the first party answers |
| 21:41.28 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 21:41.28 | *** join/#asterisk hfb (n=hfb@pool-98-119-147-30.lsanca.dsl-w.verizon.net) |
| 21:41.31 | Kobaz | nny: you'll need to first send the call to a local channel, which then answer()'s, and then dials the outside number |
| 21:42.29 | Kobaz | and that will free up the flow to allow audio to pass while the other side is ringing |
| 21:42.45 | nny | Kobaz: interesting, any advice on how to do that? |
| 21:43.08 | nny | "send the call" you mean the originate call? |
| 21:43.27 | Kobaz | yeah |
| 21:43.33 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 21:44.00 | nny | so I want Originate to run against a local channel, and when it answers bridge to the $CHANNEL (Callee or Caller) |
| 21:44.08 | Kobaz | exten => s,n,Dial(Local/8432980648@outbound... |
| 21:44.34 | Kobaz | and then context outbound { _X! => { Answer(); Dial(SIP/foo... |
| 21:44.45 | nny | Kobaz: so Originate isn't needed here? |
| 21:44.49 | Kobaz | er |
| 21:44.51 | Kobaz | i mean |
| 21:45.06 | Kobaz | wait, i'm all backwards |
| 21:45.11 | nny | hehe np |
| 21:45.12 | nny | so am I |
| 21:45.15 | Kobaz | he |
| 21:45.16 | Kobaz | heh |
| 21:45.44 | Kobaz | exten => s,n,Originate(Local/number@outbound,app,Bridge,... |
| 21:45.52 | Kobaz | and then |
| 21:45.57 | Kobaz | <PROTECTED> |
| 21:46.16 | Kobaz | so once it answers, Originate is no longer in the foreground, and audio will continue |
| 21:46.40 | bpgoldsb | Is it possible to wildcard Sip hints? i.e. if I want 'hint(SIP/200) 200 => Dial(SIP/200);', but for extensions 201-250 also |
| 21:47.50 | nny | Kobaz: something linke http://pastebin.com/m4234a1cd ? |
| 21:48.12 | nny | er |
| 21:48.13 | nny | lol |
| 21:48.14 | nny | wow |
| 21:48.20 | nny | fail at the second part hold on |
| 21:48.42 | nny | hahha wow that's not even close, sorry more coffee here |
| 21:48.47 | Kobaz | heh |
| 21:48.52 | Kobaz | you need dial(${EXTEN} |
| 21:49.20 | Kobaz | sip/${EXTEN} |
| 21:49.35 | titter | stupid question ... if I run make menuselect, and add an addon, will I lose my conf files (already backed them up) but is there a better way |
| 21:49.53 | seanbright | your conf files in /etc/asterisk? no |
| 21:50.14 | nny | http://pastebin.com/m693c79c1 ? |
| 21:51.00 | titter | im lazy lol |
| 21:51.11 | Kobaz | nny: that looks better |
| 21:51.16 | nny | gah |
| 21:51.17 | nny | s,1, |
| 21:51.18 | nny | heheh |
| 21:51.23 | nny | brewing now! |
| 21:51.33 | nny | er s//1 |
| 21:51.38 | nny | not X!,s |
| 21:51.39 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 21:51.43 | *** join/#asterisk ruben23 (n=AGENT@122.55.48.243) |
| 21:51.45 | Kobaz | oh yeah |
| 21:51.48 | Kobaz | see, that's why ael is nice |
| 21:51.55 | Kobaz | don't need to screw with line numbers |
| 21:52.00 | nny | have to add that to my list of things I need to learn |
| 21:52.19 | Kobaz | http://www.voip-info.org/wiki/view/Asterisk+AEL2 |
| 21:52.23 | [TK]D-Fender | nny: So totally DON'T :P |
| 21:52.44 | Kobaz | hah |
| 21:52.51 | Kobaz | :) |
| 21:52.57 | [TK]D-Fender | nny: Kobaz just wants not to feel alone ;) |
| 21:53.04 | nny | heh |
| 21:53.16 | seanbright | he's not alone |
| 21:53.26 | Kobaz | whew |
| 21:53.31 | *** join/#asterisk aidinb (n=Aidin@24-182-32-138.static.lnbh.ca.charter.com) |
| 21:53.41 | nny | Kobaz: callee heard ring, and nothing between transfer, couldn't continue conversation during process |
| 21:53.57 | nny | http://pastebin.com/m687c4357 |
| 21:53.58 | Kobaz | did the answer() kick in |
| 21:54.08 | nny | it appears so |
| 21:54.12 | Kobaz | hmm |
| 21:54.47 | nny | [TK]D-Fender: the System with Originate also took over the channel while it qualified the call |
| 21:55.27 | Kobaz | originate should release the line back to bridging once the first party answers |
| 21:55.33 | Kobaz | you can try this real quick |
| 21:55.35 | nny | oh it does |
| 21:55.39 | Kobaz | oh okay |
| 21:55.53 | Kobaz | so what's the issue then |
| 21:55.55 | nny | but during it disconnects to the two calls |
| 21:56.11 | Kobaz | while originate is running, no audio will pass |
| 21:56.17 | nny | crap hehe |
| 21:56.20 | Kobaz | but |
| 21:56.32 | Kobaz | you can speed it up a little bit |
| 21:56.35 | Kobaz | try this real quick |
| 21:57.10 | nny | so the Local thing isn't needed, i'll revert back to V1 with just the macro |
| 21:57.12 | Kobaz | exten => s,n,TrySystem(asterisk -rx "originate Local/8432980648@celloutbound application Bridge ${CHANNEL}") |
| 21:57.26 | Kobaz | if the pre-answer with local doesn't help, that should |
| 21:57.30 | Kobaz | that will completely background it |
| 21:57.32 | nny | will try |
| 21:57.38 | Kobaz | wont wait for an answer |
| 21:57.45 | nny | ahh well |
| 21:57.47 | nny | nm |
| 21:57.48 | Kobaz | it would be nice if the Originate() had a nowait |
| 21:58.07 | nny | trying to get it so the party being bridged isn't aware of the process |
| 21:58.38 | nny | so they would hear the ring.. all of this is pretty much the same as a follow me or tranfser right now |
| 21:58.51 | Kobaz | what is the party hearing |
| 21:59.11 | Kobaz | you want it to immediatly start ringing? |
| 21:59.15 | Kobaz | or just like, poof, the call is hijacked |
| 21:59.25 | nny | with the Local/ they hear ringing for a sec, with Just Originate it's silence while it originates |
| 21:59.28 | nny | well |
| 21:59.41 | nny | the idea is A and B aren't d/c'd until C answers |
| 21:59.45 | *** join/#asterisk hfb (n=hfb@pool-98-112-226-53.lsanca.dsl-w.verizon.net) |
| 21:59.48 | Kobaz | yeap |
| 22:00.05 | nny | so they would continue talking, and when C answers, B would talk immediately to C, no ring or silence |
| 22:00.11 | nny | however not sure if that's possible |
| 22:00.25 | Kobaz | okay so |
| 22:00.33 | Kobaz | to get rid of the ring |
| 22:00.43 | nny | well even with the ring they can't hear A right? |
| 22:00.49 | nny | er without* |
| 22:01.15 | Kobaz | if you use a musicless musiconhold context, and set Dial to use moh |
| 22:01.19 | Kobaz | actually, but. that makes no sense |
| 22:01.27 | nny | exten => s,n,Originate(SIP/8432980648@vitel-oubound,app,Bridge,${CHANNEL}) |
| 22:01.31 | nny | ^^ plays no ring |
| 22:01.36 | nny | but also silences A and B |
| 22:01.38 | Kobaz | the originate is backgrounded, either party shouldn't hear ringing at all |
| 22:02.02 | Kobaz | did you try out the asterisk -rx |
| 22:02.03 | nny | so while Asterisk is originating the SIP/8432980648@vitel-oubound A and B lose audio |
| 22:02.08 | Kobaz | correct |
| 22:02.10 | nny | can try that now sb |
| 22:02.16 | Kobaz | okay, so you didn't try that |
| 22:02.20 | Kobaz | that should fix all your trouble |
| 22:02.44 | Kobaz | the problem is, any dialplan application running, during audio bridgeing, will stop bridging until it's done |
| 22:03.01 | Kobaz | and annoyingly, Originate() doesn't end until it gets an answer |
| 22:03.14 | nny | ok will test give me a sec |
| 22:03.16 | nny | thanks for the help btw |
| 22:03.19 | Kobaz | np |
| 22:03.22 | Kobaz | you owe me big |
| 22:03.28 | Kobaz | :P |
| 22:05.02 | nny | hehe if this works i'll buy ya dinner |
| 22:05.53 | Kobaz | mmm, food |
| 22:06.30 | Kobaz | there will still be a blip in the audio, while that command is running |
| 22:06.46 | Kobaz | if it's too long for you, there are other ways, but you'll need to write some more code |
| 22:07.15 | nny | hmm didn't work |
| 22:07.24 | Akiraa | Has anyone worked with Skype and asterisk, or built VoIP networks with Skype terminations? |
| 22:07.32 | nny | one sec pb output |
| 22:07.34 | Kobaz | what didn't work |
| 22:07.38 | Kobaz | you broke it |
| 22:07.41 | nny | SystemTry |
| 22:07.42 | nny | hehehe |
| 22:07.45 | Kobaz | that's unpossible |
| 22:07.48 | Kobaz | TrySystem |
| 22:07.52 | nny | er yeah that |
| 22:08.08 | nny | http://pastebin.com/m26e1e126 |
| 22:09.06 | Kobaz | well you don't need the local channel anymore |
| 22:09.09 | Akiraa | It may be cheap-ish to divert some outside calls through a Skype line in some scenarios |
| 22:09.10 | Kobaz | since you're backgrounding it |
| 22:09.11 | nny | ok ok |
| 22:09.46 | Kobaz | oh |
| 22:09.48 | Kobaz | umm |
| 22:09.52 | Kobaz | why are you running hangup |
| 22:10.01 | nny | that was after the Originate |
| 22:10.05 | Kobaz | exten => s,n,Hangup() <--- axe that out |
| 22:10.07 | nny | otherwise it just starts over |
| 22:10.09 | Kobaz | that will hang up the call |
| 22:10.11 | nny | and loops |
| 22:10.25 | nny | k |
| 22:10.47 | Kobaz | if it's set up right, it will only run once per feature code recieved |
| 22:14.15 | nny | hmm |
| 22:14.17 | nny | doesn't call the originate number with that methos |
| 22:14.18 | nny | method |
| 22:14.20 | nny | pb one sec |
| 22:14.45 | nny | http://pastebin.com/m7ce5cd5b |
| 22:15.06 | nny | i hit *9, it shows the command, but the existing call stays connected and the originate number doesn't ring |
| 22:15.37 | Kobaz | the asterisk -rx isn't running |
| 22:15.52 | nny | yeah odd it shows the UNI connection |
| 22:15.54 | nny | UNIX* |
| 22:16.03 | nny | capital O? |
| 22:16.12 | Kobaz | doesn't matter |
| 22:16.17 | nny | hmm |
| 22:16.22 | Kobaz | what asterisk version |
| 22:16.26 | nny | 1.6.2 |
| 22:16.40 | Kobaz | k, so you're not running trunk or something silly like that |
| 22:16.42 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 22:16.43 | nny | nah |
| 22:16.48 | Kobaz | okay so |
| 22:16.49 | Kobaz | make your call |
| 22:16.55 | Kobaz | and for sanity |
| 22:16.56 | nny | try that from console |
| 22:16.57 | nny | ? |
| 22:17.05 | Kobaz | do the originate commandline instead of *9 |
| 22:17.19 | nny | k |
| 22:18.06 | *** join/#asterisk mocker (n=mocker@206.55.118.85) |
| 22:18.14 | nny | No such command 'originate SIP/8432980648@vitel-outbound application Bridge ' |
| 22:18.36 | nny | have to preface it with something? |
| 22:18.38 | Kobaz | oooooh |
| 22:18.42 | Kobaz | yes |
| 22:18.48 | Kobaz | 1.6.2 the prefix for originate is core |
| 22:18.52 | nny | ahh yeah |
| 22:19.00 | Kobaz | i knew it was something stupid |
| 22:19.01 | nny | learning that as I try to type things :D |
| 22:19.25 | Kobaz | all else fails, break things into small pieces |
| 22:20.28 | nny | hmm not core either |
| 22:20.33 | titter | ive broken more mx518 mice than I can remember |
| 22:20.46 | titter | logitech sends free replacements |
| 22:20.55 | nny | hey that's what I have |
| 22:21.01 | Kobaz | for the 1.6.2 rc's it was core originate, i think |
| 22:21.02 | nny | the top buttons are starting to suck |
| 22:21.05 | Kobaz | i haven't played much with 1.6.2 |
| 22:21.06 | titter | ya |
| 22:21.10 | nny | Kobaz: yeah let me look one sec |
| 22:21.12 | Kobaz | maybe it's channel originate |
| 22:21.14 | titter | call logitech and tell them mouse1 is broken |
| 22:21.29 | nny | titter: will do thanks! |
| 22:21.38 | nny | Kobaz: that's it |
| 22:21.38 | titter | say you have tried it on a few systems, and their latest drivers to get around their scripts of questions |
| 22:21.53 | titter | they will send a new one, and not have you send back the old one .. just read them the serial on the bottom |
| 22:22.24 | nny | Kobaz: nice |
| 22:22.43 | nny | Kobaz: think it works, let me test with business partner cell phone once he gets off of 1800HOTGOATS |
| 22:22.52 | *** join/#asterisk ticoit (n=ticoit@201.191.151.155) |
| 22:23.00 | nny | could hear audio while it called my cell |
| 22:23.24 | Kobaz | i like goats |
| 22:23.46 | nny | so TrySystem doesn't connect the two until the command succeeds? |
| 22:23.52 | nny | not sure what the fix was, but highly intrigued |
| 22:23.54 | Kobaz | well |
| 22:24.10 | Kobaz | TrySystem is... run something in a shell |
| 22:24.13 | Kobaz | via system() |
| 22:24.21 | Kobaz | asterisk -rx, just runs console commands |
| 22:24.23 | nny | yeah except it has $SYSTEMSTATUS right? |
| 22:24.43 | nny | er nm |
| 22:24.45 | nny | so does System |
| 22:25.02 | nny | not sure why doing that way would keep audio open |
| 22:25.06 | Kobaz | well |
| 22:25.08 | Kobaz | it's much faster |
| 22:25.13 | Kobaz | and it doesn't wait for anything |
| 22:25.26 | Kobaz | well, it's faster because it doesn't wait for anything |
| 22:25.33 | Kobaz | Originate() waits for an answer() |
| 22:25.36 | nny | if I tried the same command in dialplan, the whole time the call was being dialed the audio was muted |
| 22:25.45 | Kobaz | from the commandline, originate does not wait |
| 22:25.48 | Kobaz | and will return immediatly |
| 22:25.49 | nny | ahhh |
| 22:26.10 | Kobaz | so your *9 finishes, very quickly |
| 22:26.16 | Kobaz | and originate is now fully in the background |
| 22:26.19 | nny | so Originate still qualifies the call, but doesn't wait in regards to holding the two channels |
| 22:26.41 | Kobaz | not sure what you mean by qualifies the call, but it does all the usual stuff you would do if you were making a call any other way |
| 22:26.42 | nny | so i can add some error trapping and variables, but just run the command from System |
| 22:26.57 | nny | er qualify as in wait for the call to answer before Bridge |
| 22:26.59 | Kobaz | you will not get any status from the asterisk -rx command |
| 22:27.09 | nny | that should be ok |
| 22:27.14 | Kobaz | that's originate's standard behavior |
| 22:27.28 | Kobaz | it will wait for A to pick up, which is the parameter on the left |
| 22:27.38 | Kobaz | once A picks up, it will run B... which is... the right parameter |
| 22:27.42 | nny | if the call fails for some reason, (other than answer) will the call just carry on as normal between A and B? |
| 22:27.48 | Kobaz | correct |
| 22:28.05 | Kobaz | originate will time out, and go byebye |
| 22:28.05 | *** join/#asterisk fofware (n=chatzill@host109.186-125-122.telecom.net.ar) |
| 22:28.17 | nny | the whole idea was to emulate this request (c/p from g voice) |
| 22:28.18 | nny | To switch phones in the middle of an incoming call, just press * while you're talking, and your other phones will ring. Then, for example, you can pick up the call from your mobile phone (if you're about to head out), or from your desk. There are no passcodes or PINs to enter and, best of all, your caller won't even hear the switch. |
| 22:28.35 | nny | i will have to post this on voip-info somehow, sure other people will want to use it |
| 22:28.53 | Kobaz | good luck getting something onto voip-info |
| 22:29.08 | nny | hehe why? have some stuff from other issues/ info |
| 22:29.20 | Kobaz | there's so little maintenance |
| 22:29.28 | nny | ahh yeah I agree |
| 22:29.28 | Kobaz | most stuff is really really old, noone updates anything |
| 22:29.38 | nny | i'll find a way to share it |
| 22:29.41 | nny | cookbook thing maybe |
| 22:29.44 | Kobaz | forge.asterisk.org |
| 22:29.48 | nny | pm me your email btw |
| 22:30.10 | nny | i need to go caulculate four cheeseburgers from the wnedy's 99c menu :D |
| 22:30.16 | Kobaz | heh |
| 22:30.17 | Kobaz | sounds good |
| 22:30.31 | nny | you use pp? |
| 22:30.36 | Kobaz | i do |
| 22:30.38 | nny | k |
| 22:30.48 | Kobaz | paypal is really annoying these days though |
| 22:30.57 | Kobaz | they take a chunk out of user to user transactions now |
| 22:30.58 | nny | yeah I agree, any alternatives? |
| 22:31.05 | Kobaz | pen and paper? |
| 22:31.11 | nny | heh yeah |
| 22:31.27 | nny | sent |
| 22:31.34 | Kobaz | i sent myself money from my business account to my personal |
| 22:31.37 | Kobaz | and they took like 5% |
| 22:31.39 | Kobaz | i'm like wtf |
| 22:31.41 | nny | ugh lol |
| 22:31.53 | nny | middle man tax, always sucks |
| 22:32.02 | Kobaz | yeap |
| 22:32.44 | Kobaz | oo, i'm rich |
| 22:32.49 | nny | hehe |
| 22:33.10 | nny | wait till PP takes their cut, it'll be 3 cheeseburgers |
| 22:33.15 | Kobaz | yeah |
| 22:33.16 | Kobaz | probably |
| 22:34.04 | nny | last Q would this work in 1.4? |
| 22:34.14 | nny | i was told originate only works as a command in 1.6.2 |
| 22:34.14 | Kobaz | it should, since it's just -rx |
| 22:34.20 | Kobaz | need to just use regular originatre |
| 22:34.27 | nny | gotcha |
| 22:34.30 | nny | will test |
| 22:35.13 | Kobaz | take out channel originate, and just do originate |
| 22:35.19 | Kobaz | speaking of paypal |
| 22:35.22 | Kobaz | pokes vally |
| 22:35.33 | Kobaz | heh |
| 22:35.39 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 22:35.52 | Kobaz | i don't do it for the money, but... he did say he was gonna give me something |
| 22:36.10 | Kobaz | but thanks... i wasn't expecting that |
| 22:37.08 | nny | anytime, always glad to |
| 22:37.12 | nny | thanks for the help! |
| 22:39.40 | Kobaz | np |
| 22:45.42 | *** part/#asterisk ruben23 (n=AGENT@122.55.48.243) |
| 22:49.40 | nny | hmm wonder which variable would be caller extension number, CHANNEL is callee |
| 22:49.49 | nny | i'll noop some, i can find it |
| 22:51.02 | nny | lol callerid(num) should do it |
| 22:51.16 | nny | before I mangle it to be the box's number, anyways |
| 22:53.41 | nny | that worked |
| 22:53.46 | nny | <3 asterisk |
| 22:55.16 | nny | hmm can I pass a user defined variable to TrySystem? |
| 22:57.51 | Kobaz | yeah you're already passing ${CHANNEL} |
| 22:58.26 | nny | hmm odd |
| 22:58.59 | nny | i'll have to pb it |
| 23:00.19 | nny | http://pastebin.com/m3a6f53f9 |
| 23:00.32 | nny | it changes my user variable, and the SystemTry ignores it all togther |
| 23:01.46 | nny | the general goal was to change which number is dialed based on the calling phone, (i have cell phone numbers defined as CELLEXT at the top_ |
| 23:05.00 | nny | -- Executing [s@macro-cellswitch:1] NoOp("SIP/vitel-outbound-0000003e", "") in new stack is confusing, since I never set $CELL as $CHANNEL or $CALLERID(num) after the initial set. So does the variable match whatever Set tells it to regardless of context, and change it when the CID changes? I'll have to look up Set |
| 23:05.24 | nny | and also figure out a way to pass the number as a variable to SystemTry |
| 23:05.27 | nny | er trySystem |
| 23:06.15 | nny | Note that Set() changes behaviour in Asterisk 1.6 which can be controlled via asterisk.conf:Â |
| 23:07.11 | dzup | am confuse here, this is the very first time working in asterisk, i need to rent a voip provider cheap and unlimited calling to usa/canada, can some one recommend me one? |
| 23:07.24 | dzup | i need 10 channels |
| 23:07.32 | nny | dzup: all the ones I use have per minute |
| 23:07.52 | dzup | nny: you find one cheap? |
| 23:07.53 | Akiraa | Is VoIP traffic liable to be "shaped" (read: intentionally fucked) by ISPs? |
| 23:08.08 | nny | dzup: .012 outbound |
| 23:08.29 | leifmadsen | Akiraa: not on any ISP I'd use -- I don't even have a home phone line anymore, just cell and VoIP for business |
| 23:08.38 | dzup | nny: that works, you have a URI handly? |
| 23:08.44 | leifmadsen | Per minute is usually a better deal |
| 23:09.02 | nny | http://vitelity.com/ or http://www.flowroute.com/ |
| 23:09.03 | leifmadsen | I've had good luck with bandwidth.com in the US, or Unlimitel.ca in Canada |
| 23:09.13 | nny | leifmadsen: i'll have to check bandwidth |
| 23:09.51 | nny | they have a flat rate per line, that's interesting |
| 23:09.55 | leifmadsen | i have a call centre in Florida that does 50 channels with bandwidth.com and I haven't had any issues with them for over a year |
| 23:10.05 | nny | they have a PRI rate? |
| 23:10.21 | leifmadsen | i.e. I'm the consultant and I just manage the system and haven't been called by the company running the call centre :) |
| 23:10.24 | nny | er rather, something comparable* |
| 23:10.31 | nny | that's always good |
| 23:10.38 | nny | i'll have to ping them tomorrow for more info |
| 23:10.52 | nny | always hard to show savings when it's per minute vs per line |
| 23:11.57 | nny | so should System be able to use variables defined by Set? |
| 23:12.00 | *** join/#asterisk ttl- (n=patrick@d5153A420.access.telenet.be) |
| 23:12.04 | dzup | leifmadsen nny thanks am checking :) |
| 23:12.05 | nny | or only asterisk variables |
| 23:12.23 | dzup | those work with my asterisk i suppoust? |
| 23:12.32 | nny | dzup: yeah they should |
| 23:12.42 | nny | dzup: bandwidth lists digium and I use vitelity daily |
| 23:12.49 | nny | flowroute was a suggestion from a cliejt |
| 23:12.51 | nny | client* |
| 23:13.30 | dzup | nny: what plan you choose from doanload.com for those 50 lines?, am planning to have 16 channels dho |
| 23:13.53 | nny | you mean bandwidth.com ? |
| 23:14.31 | Kobaz | nyy mm |
| 23:15.00 | Kobaz | ${CELL${CELL}} |
| 23:15.03 | Kobaz | what's that for? |
| 23:15.13 | nny | it should = CELL190 |
| 23:15.19 | nny | i can use less confusing variables |
| 23:15.20 | Kobaz | ah |
| 23:15.25 | nny | CELL190=8432980648 |
| 23:15.30 | Kobaz | yeah but |
| 23:15.44 | nny | but it looks like SET is matching CIDnum as it changes |
| 23:16.27 | Kobaz | think about what that's gonna evaluate to |
| 23:16.37 | Kobaz | the inner is evaluated first |
| 23:16.43 | Kobaz | so you'll have ${CELLCELL190} |
| 23:16.49 | nny | er |
| 23:16.52 | Kobaz | which will be nothing |
| 23:16.58 | Kobaz | since that's not a variable you're previously set |
| 23:17.06 | nny | er |
| 23:17.11 | nny | let me clean it up one sec |
| 23:17.14 | nny | i was being hasty |
| 23:17.16 | ManxPower-work | I recently used ${${variable}} |
| 23:17.17 | Kobaz | quit breaking stuff |
| 23:17.29 | Kobaz | i had it all nice for you |
| 23:17.32 | Kobaz | :) |
| 23:17.32 | nny | hehe |
| 23:17.59 | nny | well $CELL is Set("SIP/190-0000003d", "CELL=190") in new stack |
| 23:18.09 | Kobaz | ManxPower-work: yeah i've done that to fake arrays in astdb |
| 23:18.17 | nny | so wouldn't it be $CELL{190} |
| 23:18.24 | nny | or rather .. eh nm |
| 23:18.29 | nny | let me simplify it hahaha |
| 23:18.51 | Kobaz | okay well |
| 23:18.53 | Kobaz | you figure it out |
| 23:18.57 | Kobaz | i'll go back to breaking my own stuff |
| 23:20.27 | nny | lol working on it |
| 23:20.34 | nny | gah |
| 23:20.51 | nny | http://pastebin.com/m22106983 |
| 23:21.07 | nny | <PROTECTED> |
| 23:21.12 | nny | notice the NoOp before |
| 23:21.34 | nny | it's changing $CELL to SIP/vitel-outbound-00000004 |
| 23:21.45 | ManxPower-work | Kobaz: I use ${VARIABLE[${SUBSCRIPT}]} for arrays |
| 23:22.11 | Kobaz | ManxPower-work: but what about storing an array in astdb |
| 23:22.27 | nny | and then the TrySystem "should" try (based on that NoOp) SIP/vitel-outbound-00000004@vitel-outbound, which is still wrong, but it's not even doing that |
| 23:22.33 | ManxPower-work | Kobaz: that's what sqlite is for. 8-) |
| 23:22.50 | Kobaz | are you reading it wrong |
| 23:22.53 | Kobaz | Set("SIP/190-0000003d",... |
| 23:23.05 | nny | ? |
| 23:23.06 | Kobaz | the output ... the first item is the channel it's acting on |
| 23:23.16 | Kobaz | why would CELL be the channel |
| 23:23.28 | nny | it shouldnt be |
| 23:23.38 | nny | i am setting CELL in the earlier context |
| 23:23.43 | nny | <PROTECTED> |
| 23:23.50 | Kobaz | looks good |
| 23:24.00 | nny | but when I NoOp it in the macro it's NoOp("SIP/vitel-outbound-00000004", "") in new stack |
| 23:24.08 | nny | and when I call it from the command it's empty |
| 23:24.24 | nny | I could just hard code the number, but it would only work for one extension |
| 23:24.40 | Katty | hi |
| 23:25.35 | nny | I R confused |
| 23:26.09 | Katty | i stay confused. |
| 23:26.11 | nny | hehehe |
| 23:26.18 | nny | maybe I need to set GLOBAL |
| 23:26.36 | *** join/#asterisk ruben23 (n=AGENT@122.55.48.243) |
| 23:28.18 | Katty | i got new products while i was out :> |
| 23:28.49 | nny | hahaha |
| 23:28.52 | nny | ok sorry Kobaz |
| 23:28.56 | *** join/#asterisk dgilmore (n=dgilmore@fedora/dgilmore) |
| 23:29.19 | nny | the NoOp SIP/vitel-outbound-00000004 was me reading output wrong |
| 23:29.28 | nny | GLOBAL was missing, hence the variable was empty |
| 23:29.36 | nny | shoots self with clue shotgun |
| 23:30.31 | *** join/#asterisk Brady1408 (n=chatzill@qw6.atadvantage.com) |
| 23:30.50 | Kobaz | heh |
| 23:31.07 | dgilmore | has anyone seen issues with iax and nat? i cant connect to any of my providers. i happen to have access to one of my providers * boxes. and its seeing my gateways ip not mine |
| 23:31.21 | Kobaz | iax plays pretty nice with nat |
| 23:31.56 | dgilmore | i just changed my router out and it stopped working |
| 23:32.05 | titter | what did you change it to? |
| 23:32.06 | Brady1408 | I have a werid bug I can't seem to google an answer too. the asterisk box says that all of my sound files that I'm trying to use do not exist in any format. they all exist in /var/lib/asterisk/sounds and I've checked the permissions on that folder and it all looks good |
| 23:32.08 | dgilmore | when i ssh to the * box its using the correct ip |
| 23:32.26 | nny | Kobaz: now to get pyscho, I won't bug ye any further, but gonna attempt to allow the reverse to happen after :D |
| 23:32.26 | dgilmore | titter: i moved from openwrt to a regular linux box |
| 23:32.37 | nny | Kobaz: I should have enough info to do that |
| 23:32.40 | nny | and thanks again |
| 23:33.00 | Kobaz | k |
| 23:33.01 | Kobaz | np |
| 23:33.20 | nny | thanks for the forge link, I'll attempt to share it there after |
| 23:33.30 | nny | this is a feature in Mitel systems |
| 23:33.37 | nny | that requires an extra linux server lol |
| 23:33.38 | Kobaz | yrah |
| 23:33.39 | Kobaz | yeah |
| 23:34.25 | Kobaz | i've done a lot of "take this old pbx, and make asterisk replace it" |
| 23:34.49 | Kobaz | replaced a siemens system recently |
| 23:35.11 | nny | yeah me too |
| 23:35.14 | nny | the hardest is key systems |
| 23:35.19 | *** part/#asterisk CRCinAU (n=CRCinAU@irc.crc.id.au) |
| 23:35.26 | nny | I won't do SLA |
| 23:35.30 | *** join/#asterisk CRCinAU_ (n=CRCinAU@irc.crc.id.au) |
| 23:35.31 | nny | just a kludge |
| 23:35.40 | CRCinAU_ | man I wish someout would remove the ban on my nick :| |
| 23:35.57 | CRCinAU_ | and with that, under asterisk 1.6.2.x, do you need app_fax for T.38 passthru? |
| 23:37.17 | Kobaz | probably |
| 23:37.39 | nny | hmm |
| 23:38.12 | CRCinAU_ | I can't find it documented anywhere :( |
| 23:38.24 | CRCinAU_ | I can see app_fax is required for sending or receiving... but nothing about passthru |
| 23:39.09 | *** join/#asterisk voipmonk (n=shido6@dsl-67-204-37-228.acanac.net) |
| 23:39.46 | nny | can I redefine a global variable in a context, but not as a exten => ? |
| 23:40.07 | nny | er rather, set it locally for that context |
| 23:45.06 | [TK]D-Fender | nny: No such thing as contextual scope |
| 23:45.35 | nny | [TK]D-Fender: thanks I see that, for some dial outs I changes the CID, but usually have the user in their own context for it |
| 23:45.58 | *** join/#asterisk xpot-mobile (n=xpot@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net) |
| 23:46.13 | CRCinAU_ | I managed to find this: From version 1.4, Asterisk supports T.38 negotiation for SIP users, and the related passthrough of UDPTL T.38 data. This allows many T.38 nodes to communicate through an Asterisk box. |
| 23:46.30 | CRCinAU_ | however 1) it doesn't mention 1.6.x - nor do I see anything about it's requirements ;( |
| 23:46.53 | p3nguin | "its" |
| 23:46.56 | CRCinAU_ | and 2) no mention of config stuff required (if any) to enable t38 passthru |
| 23:47.03 | CRCinAU_ | that too. |
| 23:47.23 | CRCinAU_ | I also managed to find: Asterisk 1.6 support G.711 and T.38 FAX origination and termination. |
| 23:47.34 | CRCinAU_ | but again, nothing about passthru :( |
| 23:58.32 | nny | i am just a butterball of questions... I have a Cisco 504g that seems to mis handle sidetone, wondering if it's defective or a feature... my spa962 doesn't seem to have the issue.. if anyone is using them and has similar experience lemme know |
| 23:59.02 | *** join/#asterisk neurosys (n=neurosys@173-9-159-182-miami.txt.hfc.comcastbusiness.net) |
| 23:59.44 | nny | wondering if the tech enabled echo cancel on the handset |
