| 00:03.11 | hardwire | how often is the outgoing spool checked? |
| 00:03.19 | hardwire | and is there a concurrency limit? |
| 00:03.48 | ESCulapio__ | with that I can do, I can not with IP phones, but as I do with the analog |
| 00:07.53 | manxpower | hardwire: once per second, I think and no. |
| 00:08.38 | manxpower | hardwire: that's why you create the spool file in a different directory on the same partition, then mv it to the correct location. you can date the file in the future and it won't be processed until that time |
| 00:10.57 | bmoraca | hardwire: any reason you can't use AMI Originate? i've found that it works far better than call files |
| 00:14.26 | p3nguin | esculapio__: Instead of auto-dial, configure the dialplan (extensions.conf) to take ANY keypress and dial the other SIP extension. |
| 00:14.41 | p3nguin | esculapio__: It's not auto, but you would only have to press one button. |
| 00:16.48 | thehar | Will SIPPEER(.|status) give status of INUSE? |
| 00:16.55 | thehar | or just OK and Unavailable, etc. |
| 00:17.27 | *** join/#asterisk PanicMan (i=SS7@122.102.33.80) |
| 00:17.42 | PanicMan | anybody worked with IAX2+trunkMTU |
| 00:18.56 | PanicMan | hello, anybody alive ! |
| 00:19.37 | PanicMan | seems, everyone is busy :( |
| 00:22.26 | ESCulapio__ | p3nguin, but to do better put, a call button |
| 00:23.10 | *** join/#asterisk seanbright (n=sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 00:23.48 | p3nguin | esculapio__: SIP devices don't notify Asterisk of off-hook status, so that's why you have to dial at least one button. |
| 00:24.34 | [TK]D-Fender | dialing jsut 1 digit does nothing either.... until the phone passes something along which is when it thinks you're done. |
| 00:28.36 | manxpower | Just configure the phone to dial a predetermined extension when you ho off hook. It's not rocket science |
| 00:28.41 | ESCulapio__ | p3nguin, As I do with the analogic |
| 00:30.01 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 00:33.22 | hardwire | bmoraca: I probably should use AMI originate.. |
| 00:33.25 | PanicMan | iax iax iax |
| 00:33.33 | PanicMan | any helper |
| 00:33.52 | hardwire | manxpower: I usually move files in to the spool dir.. 1 scan per second is good enough |
| 00:34.49 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 00:35.59 | *** part/#asterisk PanicMan (i=SS7@122.102.33.80) |
| 00:38.22 | ESCulapio__ | mangala, how ? |
| 00:38.49 | ESCulapio__ | mangala, as? |
| 00:44.23 | *** join/#asterisk ethicx (n=geneticx@65.2.59.65) |
| 00:46.04 | *** join/#asterisk jtodd (i=o05i8mpk@ns.fox-den.com) |
| 00:46.04 | *** mode/#asterisk [+o jtodd] by ChanServ |
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| 02:35.11 | martianixor | hi |
| 02:36.33 | martianixor | guys I'm trying to register, but I get Wrong Password in /var/log/asterisk/full although I'm dead sure credentials are correct, I made sure and let other people make sure, It's simple pass anyway for testing so It's impossible that It's wrong |
| 02:36.48 | *** join/#asterisk phunyguy (n=phunyguy@h69-130-65-56.kgldga.dsl.dynamic.tds.net) |
| 02:52.27 | Naikrovek | just watched Sunshine Cleaning. awesome movie |
| 02:52.31 | Naikrovek | awesome++ |
| 02:55.49 | L2Logic | Hello.. does anyone know how to enable early media with SIP endpoints ? |
| 03:07.20 | AeroCloud | Question: When using an AGI script to $agi->exec('Dial','blahblah'); is there a way to run a function when a person answers the call? |
| 03:08.07 | AeroCloud | I dont want to run a macro, as thats not within the AGI script |
| 03:12.59 | russellb | you could run a macro that executes another AGI :-( |
| 03:13.35 | AeroCloud | can I pass about 10 variables into it? |
| 03:13.40 | russellb | yup |
| 03:13.51 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 03:13.59 | russellb | or you could have another thread in your AGI that connects to manager and watches for an event that the call was answered |
| 03:17.29 | AeroCloud | wouldnt that be processor intensive? |
| 03:18.08 | AeroCloud | so lets say I want to run the Macro with the M() inside the options of dial |
| 03:18.23 | AeroCloud | how would I pass a current variable into that macro |
| 03:18.31 | AeroCloud | from the agi |
| 03:20.25 | martianixor | and when I remove secret It registers normally ! |
| 03:20.57 | AeroCloud | basically I'm writing my own custom CDR because of an asterisk farm, need to have 1 central CDR with my specific fields, not the normal CDR fields |
| 03:21.16 | AeroCloud | right now it added in the ring time to the seconds |
| 03:26.57 | *** join/#asterisk miguel3239 (n=elguero@ns1.nashuacs.com) |
| 04:10.09 | *** join/#asterisk dmz (n=dmz@64.203.233.195.dyn-cm-pool35.hargray.net) |
| 04:14.08 | drmessano | ~book |
| 04:14.08 | 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 |
| 04:19.09 | *** join/#asterisk lowlevel (n=Stuart@lowlevel.ca) |
| 04:22.17 | *** join/#asterisk spck (n=spck@75-134-20-74.dhcp.mdsn.wi.charter.com) |
| 05:16.19 | asif | Hey is cdr_yada no longer used? |
| 05:17.08 | *** join/#asterisk brunner (n=chris@68.119.87.106) |
| 05:18.15 | brunner | Is it possible to use asterisk with VoicePulse unlimited subscriptions? |
| 05:22.37 | TJNII | brunner: (732) 339-5100 <- Call and ask. You would be a paying customer. |
| 05:24.09 | *** join/#asterisk maour (n=gnu@unaffiliated/maour) |
| 05:28.25 | *** join/#asterisk coppice (n=chatzill@68.166.17.210.dyn.pacific.net.hk) |
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| 05:58.10 | manxpower | most providers have a 3,000 - 6,000 min cap on their "unlimited" accounts |
| 05:58.42 | jql | makes a note to cap at between 3k and 6k /month |
| 06:01.21 | manxpower | jql: It's sometime in the fine brint |
| 06:01.28 | manxpower | and print too |
| 06:08.02 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
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| 06:21.55 | *** part/#asterisk TheOpenSourcerer (n=TheOpenS@81.178.65.1) |
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| 06:49.11 | dustybin | is there a way to pass STRINGS + VARIABLES from asterisk into the shell for scripts? |
| 06:49.40 | dustybin | like CALLER ID, etc |
| 06:58.53 | *** join/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 07:07.46 | *** join/#asterisk ccesario (n=ccesario@189-19-6-236.dsl.telesp.net.br) |
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| 07:16.27 | *** part/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
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| 07:21.55 | *** join/#asterisk Milad (n=milad@unaffiliated/slackark) |
| 07:23.51 | Milad | Anyone has a problem with using COMMAND in AMI ? when I run asterisk option " asterisk -f -U root -vvvvg -c" it did not response any text ! but when I run " asterisk -f -vvvvg -c" everything works |
| 07:27.20 | *** join/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
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| 08:10.41 | asif | hello all! |
| 08:11.24 | asif | anyone here used cdr_yada? |
| 08:12.40 | *** join/#asterisk gr0mit (n=tim@2001:8b0:3d9:0:d4cf:a866:245:3c3d) |
| 08:15.23 | iksik | hm, when some users uses ie. context 'foo' which includes context default. dialplan show 5000@default - should returns me my users? or dialplan show 5000@foo only? |
| 08:38.42 | *** join/#asterisk louben (n=lou@212-70-216-131.ath.static.tee.gr) |
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| 10:26.20 | *** join/#asterisk whirm (n=whirm@89.131.71.242) |
| 10:26.32 | whirm | hi all |
| 10:27.59 | garymc | Hi, I finaly got ssh working. Im at home and I can ring my office extensions via my house through zoiper but we cant hear each other. I need to pastebin my asterisk CLI from putty |
| 10:28.16 | garymc | but putty works oddly so im having trouble copying the text from the putty window |
| 10:28.38 | garymc | so i can pastebin it |
| 10:28.46 | garymc | Anyone help me out here? |
| 10:31.49 | iksik | asif what's that? |
| 10:32.01 | louben | can we see the logs? |
| 10:32.05 | garymc | never mind done it |
| 10:32.36 | whirm | I'm trying to compile chan_bluetooth using the precompiled asterisk in debian (1.4.21) is this possible? all the doc I could find tells me to unpack the chan_bluetooth tgz over the asterisk source and recompile everything... |
| 10:33.01 | garymc | here is my zoiper calling my office extension with no voice. Anyone know why this isnt working here? http://pastebin.ca/1562991 |
| 10:33.38 | garymc | it worked fine when i was in the office with my laptop and softphone, but when i got home to test it rings the extensions but we cant hear each other :( |
| 10:34.53 | box2 | ipkall is run by the CIA AAAAAAAAAAAAAHHHHHHHHHH |
| 10:35.32 | garymc | Is anyone on here good with diagnosing problems in asterisk? |
| 10:35.48 | garymc | I know TK D-Fender is but hes obviously asleep right now |
| 10:37.23 | whirm | I just found about chan_mobile |
| 10:37.38 | whirm | wich of those is better? |
| 10:46.37 | garymc | Not many people about this time of day midday in UK here |
| 10:47.41 | garymc | [TK]D-fender : I got my Asterisk CLI pastebin for my call from home to office extension using zioper. Could you take a look and advise? http://pastebin.ca/1562991 |
| 10:56.37 | iksik | sorry, but still I have no clue how it works... when some users uses ie. context 'foo' which includes context default. dialplan show 5000@default - should returns me my users? or dialplan show 5000@foo only? |
| 11:19.28 | *** join/#asterisk Had (n=chatzill@82-45-194-9.cable.ubr02.hari.blueyonder.co.uk) |
| 11:19.41 | Had | hello |
| 11:20.37 | Had | I have 1 question about using asterisk as gateway between sip provider and second sip system |
| 11:21.17 | Had | how can I create sip trunk for the 2nd system? should I just use sip extension? |
| 11:22.03 | Had | Will than be all caller IDs shown correctly? |
| 11:22.33 | louben | Had, do a sip friend |
| 11:22.35 | Had | and also will the DID's be shown correctly in second system? |
| 11:23.08 | louben | regarding CLIDs I know they work ok, Also should DIDs work |
| 11:23.44 | Had | OK that's fine there, that will be very easy :) thanks for your help! |
| 11:27.06 | garymc | So anyone help me get my sound working? All phone in office work great. Now Im trying to get my remote softphone to work. My softphone registers to my Asterisk PBX I can call the office extensions from my softphone but we cant hear each other. Anyone know anything about this? |
| 11:36.00 | *** join/#asterisk maour (n=gnu@unaffiliated/maour) |
| 11:37.45 | garymc | right i gotta go for now be back later |
| 11:37.52 | *** join/#asterisk lawrence (i=kvirc@80.73.70.169) |
| 11:38.00 | lawrence | hi all! |
| 11:38.06 | lawrence | what about skypiax? |
| 11:38.20 | lawrence | Is this project still alive? |
| 11:39.10 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 11:44.58 | coppice | yes it is. I don't know how well it works with asterisk, but the version for freeswitch seems to be working for people |
| 11:45.20 | lawrence | and what difference between FS and * |
| 11:45.23 | lawrence | ? |
| 11:45.46 | *** join/#asterisk ltd_wk (i=z@patwk.transact.net.au) |
| 11:47.13 | *** join/#asterisk GRiFTaH-xtc (i=GRiFTaH-@115.128.48.96) |
| 11:51.44 | Had | http://www.anders.com/cms/266 |
| 12:02.14 | *** join/#asterisk voipmonk (n=voipmonk@dsl-67-212-7-81.acanac.net) |
| 12:02.38 | dustybin | how can one setup auth so one has to enter a 8-digit password to enter a protected context? |
| 12:02.48 | iksik | FS is better then * ? :) |
| 12:03.05 | dustybin | http://www.voip-info.org/wiki/view/Asterisk+user+authentication |
| 12:03.11 | voipmonk | dustybin: you can use vmauthenticate and voicemail.conf |
| 12:03.15 | voipmonk | or you can |
| 12:03.31 | voipmonk | build the dialplan extension as the passcode |
| 12:03.39 | voipmonk | or do a mysql dip |
| 12:03.52 | dustybin | thats a good idea |
| 12:04.07 | voipmonk | iksik: is an apple better than a kiwi? |
| 12:04.09 | dustybin | voipmonk: nothing to do with voicemail |
| 12:04.16 | voipmonk | doesnt have to be |
| 12:04.19 | dustybin | i just want the user to enter a protected menu |
| 12:04.25 | voipmonk | but VMauthenticate can still be used |
| 12:04.31 | dustybin | ok |
| 12:04.36 | *** join/#asterisk war9407 (i=war@liquidswords.org) |
| 12:04.45 | dustybin | exten => s,1,Authenticate(XXXXX) |
| 12:04.46 | dustybin | ? |
| 12:04.52 | voipmonk | no |
| 12:04.59 | voipmonk | add a context at the end |
| 12:05.05 | voipmonk | and then create the context in voicemail.conf |
| 12:05.10 | voipmonk | so it doesnt say "comedian mail" |
| 12:05.25 | dustybin | exten => s,1,Authenticate(myarea) |
| 12:05.31 | dustybin | in voicemail.conf |
| 12:05.35 | dustybin | [myarea[ |
| 12:05.36 | dustybin | ] |
| 12:05.41 | voipmonk | so exten => _XXXX,1,VMAuthenticate(${EXTEN}@blah) |
| 12:05.47 | dustybin | ohh |
| 12:05.52 | voipmonk | and in voicemail.conf u have [blah] |
| 12:05.57 | dustybin | voipmonk: is that the proper way of auth? |
| 12:05.59 | voipmonk | 1234 => 1234 |
| 12:06.10 | dustybin | or is this something you have invented? |
| 12:06.17 | voipmonk | do you put fuel in your vehicle with one leg up? |
| 12:06.21 | dustybin | :P |
| 12:06.26 | voipmonk | is that the proper way to put fuel in your car? |
| 12:06.29 | voipmonk | does it matter? |
| 12:06.51 | dustybin | not really.. but i dont see why you need to use voicemail stuff for basic auth |
| 12:06.55 | dustybin | auth should be on its own |
| 12:07.02 | iksik | voipmonk, no |
| 12:07.04 | voipmonk | then write one and go away :) |
| 12:07.10 | voipmonk | or pick another method |
| 12:07.15 | iksik | voipmonk, kiwi is much better |
| 12:07.18 | *** join/#asterisk Tim_Toady (n=moi@adsl17-44.kln.forthnet.gr) |
| 12:08.03 | dustybin | surely this is the proper way of doing it: |
| 12:08.03 | dustybin | http://www.voip-info.org/wiki/view/Asterisk+cmd+Authenticate?utm_source=voip-info&utm_medium=module&utm_campaign=recentchanges |
| 12:08.13 | dustybin | Authenticate(password[,options[,maxdigits]]) |
| 12:08.30 | voipmonk | if you could read why did you come here? |
| 12:08.32 | voipmonk | :) |
| 12:08.35 | dustybin | :( |
| 12:08.52 | dustybin | feels hurt from voipmonk comment |
| 12:09.15 | *** join/#asterisk lawrence (i=kvirc@80.73.70.169) |
| 12:09.27 | lawrence | my bad connection. |
| 12:20.23 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-us/x-fbdxkgxqsgwvhfdj) |
| 12:21.15 | dustybin | how does one edit asterisk configs using the CLI? |
| 12:21.24 | dustybin | does it have a built in text editor? |
| 12:21.30 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 12:39.56 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 12:51.03 | dustybin | I have replaced all my sounds with these: http://www.enicomms.com/cutglassivr/ |
| 12:51.10 | dustybin | i downloaded ulaw version |
| 12:51.34 | dustybin | the format of the files look like this: vm-starmain.g711u |
| 12:52.08 | dustybin | my voicemail has: allow=ulaw |
| 12:52.21 | dustybin | however, it complains that the files do not exist |
| 12:52.42 | dustybin | file.c:933 ast_streamfile: Unable to open vm-login (format 0x4 (ulaw)): No such file or directory |
| 12:53.18 | dustybin | vm-login.g711u <--- file exists in the same location |
| 12:56.43 | iksik | is there any PHP module for asterisk? |
| 12:57.51 | *** join/#asterisk SaiSoma (n=SaiSoma@adsl-074-167-136-030.sip.mob.bellsouth.net) |
| 13:05.47 | *** join/#asterisk sercik (n=ciccio@host193-109-dynamic.53-79-r.retail.telecomitalia.it) |
| 13:05.50 | sercik | hello! |
| 13:05.58 | sercik | i know have advised: "The X100P/X101P cards are poor cards" |
| 13:06.08 | sercik | i red o reilly book |
| 13:06.16 | sercik | or i'm reading.. |
| 13:06.37 | sercik | but if i want to make some try woth asterisk?... that cards are compatible with dahdi? |
| 13:06.43 | *** join/#asterisk lawrence (i=kvirc@80.73.70.169) |
| 13:09.22 | *** join/#asterisk voipmonk (n=voipmonk@67.212.7.81) |
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| 13:15.01 | lawrence | guys, does anybody know where I can find current version of skypiax for asterisk (not FS)? |
| 13:15.04 | *** join/#asterisk troubled (n=troubled@unaffiliated/troubled) |
| 13:18.27 | *** part/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com) |
| 13:19.25 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 13:22.21 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 13:23.46 | *** join/#asterisk war9407 (i=war@liquidswords.org) |
| 13:24.13 | iksik | hum, freeswitch looks really great :| |
| 13:24.53 | lawrence | :) |
| 13:25.01 | lawrence | I know... But... |
| 13:25.12 | lawrence | I looking for skypiax. |
| 13:25.36 | lawrence | May be u know about mirrors/ |
| 13:25.38 | lawrence | ? |
| 13:33.29 | [KS] | is it possible to install dahdi on kernel that has no module support? |
| 13:33.36 | [KS] | actually, I installed it, but I cannot run it |
| 13:33.44 | [KS] | says that kernel has no module support :] |
| 13:34.07 | Chainsaw | [KS]: What you are trying to accomplish is indeed impossible. |
| 13:34.26 | [KS] | you should see how I installed it |
| 13:34.35 | [KS] | I had no kernel source/headers |
| 13:34.45 | [KS] | because it's custom kernel from my rps provider :} |
| 13:34.56 | Chainsaw | So you are in a hostile environment. |
| 13:35.25 | [KS] | pretty much |
| 13:38.39 | iksik | lawrence.. hmmm mirrors of what? |
| 13:38.46 | iksik | i've just downloaded latest FS from trunk |
| 13:38.48 | iksik | fswitch@sip ~/src/freeswitch.trunk/src/mod/endpoints/mod_skypiax $ ls |
| 13:38.48 | iksik | Makefile README alsa asterisk configs kernel mod_skypiax.2008.vcproj mod_skypiax.c skypiax.h skypiax_protocol.c |
| 13:39.08 | *** join/#asterisk sezuan (i=sezuan@mobil5.vp.ip6.scheff32.de) |
| 13:40.17 | *** join/#asterisk SLCarrijo (i=SLCarrij@200.174.126.62) |
| 13:41.40 | lawrence | mirror of skypiax |
| 13:41.46 | lawrence | celliax is down ;( |
| 13:48.47 | *** join/#asterisk andresmujica1 (n=andresmu@ubuntu/member/andresmujica) |
| 13:51.46 | *** join/#asterisk voipmonk (n=voipmonk@dsl-67-212-7-81.acanac.net) |
| 14:02.51 | [TK]D-Fender | garymc: worthless debug |
| 14:06.53 | *** join/#asterisk magronez (n=eusei@unaffiliated/magrao/x-2903) |
| 14:08.46 | *** join/#asterisk blkry (n=chatzill@96.37.27.72) |
| 14:11.02 | *** join/#asterisk afink (n=afink@204.26.87.226) |
| 14:18.15 | *** join/#asterisk blueskiesokie (n=blueskie@unaffiliated/blueskiesokie) |
| 14:31.01 | *** join/#asterisk crazybyte (n=crzp@unaffiliated/crazypenguin/x-000001) |
| 14:39.34 | *** join/#asterisk af_ (n=getsmart@88.149.240.140) |
| 14:57.05 | *** join/#asterisk maour (n=gnu@unaffiliated/maour) |
| 14:58.55 | *** join/#asterisk jimw (n=chatzill@adsl-70-41-228.gnv.bellsouth.net) |
| 14:59.39 | *** join/#asterisk boomerang (n=boomeran@unaffiliated/boomerang) |
| 15:01.48 | jimw | Is there a secret handshake a SIP UA can use to discover the Asterisk version it's registering with? |
| 15:02.01 | *** join/#asterisk propellerhead (n=yogurt2u@host186.190-30-30.telecom.net.ar) |
| 15:05.19 | [TK]D-Fender | jimw: set *'s UA |
| 15:06.33 | jimw | [TK]D-Fender: Huh? I can parse the 3-word sentence, but I don't have a clue what it means :) |
| 15:06.52 | iksik | [TK]D-Fender I'm bet that You will know the answer for my question ;P |
| 15:07.10 | iksik | <iksik> hm, when some users uses ie. context 'foo' which includes context default. dialplan show 5000@default - should returns me my users? or dialplan show 5000@foo only? |
| 15:07.11 | iksik | [; |
| 15:08.55 | [TK]D-Fender | jimw: you can set the USERAGENT send in sip.conf |
| 15:09.50 | [TK]D-Fender | iksik: huh? |
| 15:09.51 | *** join/#asterisk Mango (n=Mango@96.49.69.137) |
| 15:10.01 | jimw | [TK]D-Fender: This is not my *. I'm trying to register with one provided by a commercial registrar and I need to work around specific bugs. |
| 15:10.48 | [TK]D-Fender | jimw: You want to know what version a box you DON'T control is running? No, if they don't advertise then you can't |
| 15:10.52 | iksik | [TK]D-Fender, if one context includes another one, and my user account uses first one... my user will be visible under that second one? :S |
| 15:10.55 | [TK]D-Fender | jimw: go ASK THEM |
| 15:11.19 | [TK]D-Fender | iksik: Visible? this is DIALPLAN. It has nothing to do with "visible" |
| 15:11.36 | iksik | dialplan show user_no@CONTEXT - erm.. |
| 15:11.42 | [TK]D-Fender | iksik: Dialplan is for DIALING. Each device can dial extens in the context it points to. |
| 15:12.16 | jimw | [TK]D-Fender: It is unlikely they will have a clue. I think I'm working around bug 0015102 or the fix thereof. |
| 15:13.03 | jimw | [TK]D-Fender: Well, you've answered my question anyway. No secret handshakes or fraternity rings. |
| 15:13.56 | [TK]D-Fender | jimw: jimw what bug? |
| 15:14.08 | iksik | [TK]D-Fender I know for what dialplan is... but I'm looking at extensions.conf which was generated by GUI... it looks like THERE IS NO user which uses DEFAULT context... all users are using ie. DialOut context which includes DEFAULT... and users are visible under this default one... dialplan show USER@DEFAULT works o.O |
| 15:14.18 | jimw | Jackpot registrations caused by nonce SNAFUs |
| 15:14.45 | [TK]D-Fender | iksik: How the hell is a USER visible in a context? |
| 15:15.14 | jimw | [TK]D-Fender: Except my UA simply shuts down instead of reregistering ad infinitum. |
| 15:15.24 | iksik | O.o |
| 15:21.09 | *** join/#asterisk Mango (n=Mango@96.49.69.137) |
| 15:21.20 | Mango | Sorry...if anyone responded I missed it. |
| 15:21.32 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 15:23.04 | [TK]D-Fender | Mango: If you asked anything, it never made it... |
| 15:23.44 | Mango | Oh. |
| 15:23.56 | Mango | goes to have a harsh word with his internet connection. |
| 15:24.03 | lirakis_ | umm.... lol |
| 15:24.10 | Mango | I had a module question but I think I've solved it anyway. |
| 15:25.51 | dustybin | adds some ChanSpy to Mango dialplan |
| 15:26.04 | Mango | :) |
| 15:28.16 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 15:29.30 | AeroCloud | dustybin were you trying to pass variables into AGI script? |
| 15:30.36 | AeroCloud | exten => s,1,AGI(script|variable|variable|variable|etc) |
| 15:30.43 | dustybin | THANKS!!! |
| 15:31.11 | iksik | [TK]D-Fender, http://pastebin.com/m3393b27e |
| 15:31.19 | AeroCloud | caller id and most of the variables inside asterisk are already available inside AGI |
| 15:32.17 | *** join/#asterisk mike321 (n=chatzill@141.214.234.28) |
| 15:34.59 | [TK]D-Fender | iksik: What is yourr questionh? You seem incapable of saying what the problem is as for CLI for the dialplan, pasten ACTUAL dialplan . |
| 15:35.27 | iksik | why I see user extension 6000 under default context... |
| 15:35.37 | [TK]D-Fender | iksik: Why not? |
| 15:35.44 | iksik | it have something to do withi include ? |
| 15:35.46 | iksik | with* |
| 15:36.15 | [TK]D-Fender | iksik: Look in your actual dialplan. |
| 15:37.00 | *** join/#asterisk hacked_kernel (n=hacked_k@41.234.166.14) |
| 15:37.33 | iksik | [TK]D-Fender there is only include = default (where default is empty ;) ) |
| 15:37.49 | hacked_kernel | is it possible to integrate asterisk with the ordinary land line through the modem to make and receive calls? |
| 15:37.57 | [TK]D-Fender | iksik: users.conf GENERATES dialplan <- |
| 15:38.28 | iksik | hm o.O |
| 15:38.33 | [TK]D-Fender | hacked_kernel: No, your XYZ generic crap modem is all but worthless. You'll need to get a supported interface |
| 15:39.45 | iksik | currently i'm using realtime with exactly the same configuration in extensions/users/sip/etc. and ext. 6000 is not visible here: dialplan show 6000@default |
| 15:40.32 | [TK]D-Fender | iksik: * GUI does not work with realtime, and dialplan show does not work with DB's |
| 15:40.37 | hacked_kernel | [TK]D-Fender, whats that interface? |
| 15:41.01 | [TK]D-Fender | hacked_kernel: Digium, Sangoma, etc all make PC cards for this. There are external SIP gateways, etc. |
| 15:41.25 | iksik | [TK]D-Fender, it doesn't metter if YOUR GUI works or not... my works fine... but, "extensions 6000 was not found under context default" error tells me that asterisk doesn't see it |
| 15:41.30 | [TK]D-Fender | hacked_kernel: www.telephonydepot.com , www.voipsupply.com <- go cruise to get a sample of supported hardware |
| 15:41.47 | [TK]D-Fender | iksik: Where is the error? |
| 15:41.51 | iksik | on console |
| 15:42.05 | [TK]D-Fender | iksik: I have not seen your actual diaplan, nor a dump of that specific context |
| 15:42.29 | iksik | hum |
| 15:43.15 | iksik | dialplan show @default |
| 15:43.15 | iksik | ? |
| 15:44.16 | [TK]D-Fender | iksik: WHERE IS THE ERROR? |
| 15:45.11 | iksik | [Sep 12 17:44:46] WARNING[10411]: pbx.c:3839 __ast_pbx_run: Channel 'SIP/6000-b833b028' sent into invalid extension '6005' in context 'default', but no invalid handler |
| 15:45.13 | iksik | here |
| 15:45.14 | iksik | ;) |
| 15:45.53 | iksik | exactly the same configuration, but on flatfiles works... |
| 15:46.00 | hacked_kernel | [TK]D-Fender, those interfaces are too expensive, is there a less expensive way? |
| 15:46.04 | iksik | 6005 is visible under 'default' for asterisk |
| 15:46.44 | [TK]D-Fender | hacked_kernel: Linksys SPA-3102 |
| 15:47.06 | [TK]D-Fender | iksik: I don't see that anywhere |
| 15:47.18 | [TK]D-Fender | iksik: You showed me **6000** not "6005" |
| 15:47.44 | iksik | 6000 is EXACTLY the same as 6005 |
| 15:47.55 | iksik | there is no difference between them |
| 15:48.10 | [TK]D-Fender | iksik: YOU DIDN'T FUCKING SHOW ME ANYTHING VALID |
| 15:48.20 | [TK]D-Fender | iksik: where is the COMPLETE dialplan? Where is the COMPLETE call attempt with failure? |
| 15:48.39 | iksik | i've just pasted it You... it is COMPLETE |
| 15:48.48 | [TK]D-Fender | iksik: Don't show me your car and then ask me why your TRUCK doesn't work |
| 15:49.04 | [TK]D-Fender | iksik: You showed me 6000. that is not 6005. |
| 15:49.39 | iksik | lol... If I just use SED to replace 6000 into 6005 it will be fine? |
| 15:49.44 | [TK]D-Fender | iksik: NO |
| 15:49.51 | iksik | why? ;) they are the same anyway ;]; |
| 15:49.52 | [TK]D-Fender | iksik: Show me your REAL DIALPLAN. |
| 15:51.01 | *** join/#asterisk bijit (n=chatzill@201.199.157.186) |
| 15:52.16 | [TK]D-Fender | iksik: You clearly don't want to fix this... |
| 15:52.30 | [TK]D-Fender | goes off to do something productive |
| 15:54.17 | iksik | [TK]D-Fender http://wklej.org/id/149039/ |
| 15:54.38 | bijit | can i have isdn extension? instead of trunk? |
| 15:55.34 | [TK]D-Fender | iksik: Where si the failed call? Where is the complete CLI dump of default? |
| 15:55.39 | iksik | one moment |
| 15:55.54 | *** join/#asterisk _joe (n=joseph@craptome.org) |
| 15:56.06 | *** join/#asterisk seanmh (n=johndoe@c-69-254-131-168.hsd1.nm.comcast.net) |
| 15:56.27 | *** join/#asterisk JayTee52 (n=jforde05@unaffiliated/jaytee) |
| 15:57.15 | _joe | what's up, asterisk peeps? |
| 15:57.21 | _joe | how are all you all on this fine saturday? |
| 15:57.28 | Mango | dialplan reload |
| 15:57.32 | Mango | er |
| 15:57.34 | Mango | fine thank you |
| 15:57.38 | _joe | ;) |
| 15:59.33 | Mango | almost got my dialplan the way it needs to be. |
| 15:59.37 | Mango | just tweaking MOH |
| 15:59.39 | _joe | excellent! |
| 15:59.50 | Mango | is an Asterisk n00b but liking it :) |
| 16:00.07 | Mango | Thinking about making a MOH message like Karl's. |
| 16:00.49 | iksik | [TK]D-Fender, extensions.conf http://wklej.org/id/149039/ | users.conf http://wklej.org/id/149040/ | dialplan show @default http://wklej.org/id/149041/ | here everything works... it's from flatfiles, there is no error... error is on asterisk installation which uses DB ( EXACTLY THE SAME CONF ): [Sep 12 17:44:46] WARNING[10411]: pbx.c:3839 __ast_pbx_run: Channel 'SIP/6000-b833b028' sent into invalid extension '6005' in context 'default', but no invalid handler |
| 16:01.19 | drmessano | Sup jaytee |
| 16:02.29 | [TK]D-Fender | iksik: No, there IS an error, and that is due to a CALL. |
| 16:02.50 | *** join/#asterisk retentiveboy (n=pdugas@74-95-28-37-Atlanta.hfc.comcastbusiness.net) |
| 16:03.45 | [TK]D-Fender | iksik: And nothing you have shown me has anything to do with a DB |
| 16:04.38 | iksik | is this what You want http://wklej.org/id/149043/ ? |
| 16:07.09 | [TK]D-Fender | iksik: right now you are talking about 2 different system and I'm not trusting what I'm seeing and you've been reluctant to show what little you have. |
| 16:07.49 | [TK]D-Fender | iksik: "error is on asterisk installation which uses DB" <- |
| 16:08.18 | [TK]D-Fender | iksik: I am not wasting my time working on useless crap |
| 16:08.28 | iksik | ok, I see... some people just even saying that "water is to wet" ;) If You don't want to help, just don't do it.... |
| 16:08.48 | [TK]D-Fender | iksik: You are showing me text dialplan from another fucking system. |
| 16:08.57 | iksik | it's the same dialplan |
| 16:09.05 | [TK]D-Fender | iksik: iksik Not the same = waste of time |
| 16:09.14 | iksik | like extensions 6000 and 6005 ... |
| 16:09.35 | iksik | but no... You should read it by your self AND WASTE some time |
| 16:09.43 | iksik | to see, that they are EXACTLY the same |
| 16:10.18 | [TK]D-Fender | Something is clearly different and showing bits off another system doesn't do any good |
| 16:10.31 | iksik | then tell me, what are You doing to do not waste Your time with this attitude? ;> |
| 16:10.37 | riddlebox | arghh trying to setup flash operator panel, and cannot get rid of it asking for a password to transfer, or click to dial |
| 16:11.45 | iksik | [TK]D-Fender, it looks like the differences are between possibility of "include" in dialplans from flatfiles, and missing this feature in DBs - don't know.. just guessing, and asking here |
| 16:12.41 | iksik | dialplan/context damn, I'm confused now... but it's in extensions.conf |
| 16:12.42 | [TK]D-Fender | iksik: I don't see your DB, and last I checked you can't INCLUDE inside a DB, and I don't see the DB's contents or the flat-files that support it. |
| 16:13.00 | iksik | http://wklej.org/id/149039/ |
| 16:13.03 | iksik | # |
| 16:13.03 | iksik | [DLPN_DialPlanIN] |
| 16:13.03 | iksik | # |
| 16:13.03 | iksik | include = DID_livedata_timeinterval_allthetime |
| 16:13.03 | iksik | # |
| 16:13.04 | iksik | include = default |
| 16:13.07 | [TK]D-Fender | iksik: It isn't in extensions.conf, its in a DB. You just told us that. Or are you more confused now? |
| 16:13.36 | iksik | I'm talking about place where it IS used |
| 16:14.20 | box2 | i'm talkin bout R.E.S.P.E.C.T. |
| 16:14.25 | box2 | that is what it means to me |
| 16:14.59 | iksik | <'[TK]D-Fender> iksik: I don't see your DB, and last I checked you can't INCLUDE inside a DB........... Yeap, but as You can see context/dialplans in flatfiles are using it... so dialplans/contexts in DB |
| 16:15.32 | *** join/#asterisk puzzled (n=foobar@83.83.58.178) |
| 16:15.38 | iksik | they was auto-exported into DB from flatfiles |
| 16:41.40 | garymc | [TK]D-Fender : I just got back now, why is it worthless? What else do i need to do then? |
| 16:42.13 | [TK]D-Fender | garymc: You have no audio. you need SIP DEBUG for the failed attempt, |
| 16:42.23 | [TK]D-Fender | garymc: along with sip configs |
| 16:42.38 | garymc | right, are they in etc/asterisk folder? |
| 16:43.22 | [TK]D-Fender | garymc: Yes |
| 16:43.31 | garymc | so i paste them to you |
| 16:43.33 | garymc | :) |
| 16:44.03 | box2 | ipkall, run by the CIA? |
| 16:46.43 | garymc | do i need to show all my sip conf files? I see no sip debug file |
| 16:48.49 | garymc | Ok I type "sip debug" in asterisk CLI do I need to make a call with that on to get some info? |
| 16:51.37 | garymc | ok im making a call to voicemail as there is no one in the office to take the call does this matter? As I cant hear the voicmail sound either? |
| 16:55.21 | bijit | create isdn exten for video conference asterisk is this possible in asterisk? |
| 16:55.57 | garymc | [TK]D-Fender : http://pastebin.ca/1563300 heres some of the sip debug ? Lots came on the screen I think alot of it Putty doesnt show because there was so much? |
| 16:56.57 | *** join/#asterisk droemel (n=droemel@p548EF919.dip.t-dialin.net) |
| 16:57.07 | [TK]D-Fender | garymc: Yes you need to look at the call. a COMPLETE call |
| 16:57.17 | [TK]D-Fender | garymc: and include your configs as requested. |
| 16:57.32 | garymc | Im sure this call went to voicemail but i just got silence |
| 16:57.37 | garymc | does that not help? |
| 16:57.39 | [TK]D-Fender | bijit: No |
| 16:57.48 | [TK]D-Fender | garymc: COMPLETE call |
| 16:58.04 | garymc | ok ill wait till someone is back in the office about 30 mins |
| 16:58.15 | bijit | thanks [TK]D-Fender |
| 17:00.38 | *** join/#asterisk netpro25_ (n=mmanning@64-238-176-253.ksg.apt.gru.net) |
| 17:02.27 | netpro25_ | Hey guys I have a question regarding SIP over the internet. I am wondering if using SIP over a VPN would help reduce some off the packet loss that I am experiencing. For some reason my phone conversations are abruptly terminating at random intervals. |
| 17:02.38 | *** join/#asterisk edguy3 (n=edguy3@c-98-221-27-224.hsd1.nj.comcast.net) |
| 17:02.45 | netpro25_ | *of |
| 17:02.58 | drmessano | Why would a VPN fix that? |
| 17:03.30 | netpro25_ | drmessano: well the vpn would connect my home network to the server. Figured VPN would correct any packet issues |
| 17:05.07 | netpro25_ | drmessano: I am unsure as to any other way to fix the issue |
| 17:06.28 | netpro25_ | maybe if I setup an asterisk server on this end convert to IAX2 and send that over the internet |
| 17:08.22 | [TK]D-Fender | netpro25_: Whats to stop VPN packets from disappearing? |
| 17:08.47 | [TK]D-Fender | netpro25_: packets are packets. Doesn't matter how you wrap the package when they keep falling off the truck... |
| 17:08.50 | *** join/#asterisk manxpower (n=EWieling@63.sub-70-222-111.myvzw.com) |
| 17:09.05 | netpro25_ | [TK]D-Fender: did not know VPN uses TCP |
| 17:09.25 | netpro25_ | which would be bad, I am assuming |
| 17:09.32 | manxpower | that totally depends on the VPN |
| 17:09.33 | [TK]D-Fender | netpro25_: TCP for voice? Expect jiterry results, but that might help |
| 17:09.51 | manxpower | IPSec, for example, uses neither TCP nor UDP |
| 17:10.19 | [TK]D-Fender | manxpower: I smell SSL VPN |
| 17:10.31 | netpro25_ | so what is a good way to track down abrupt termination of phone calls? Asterisk logs? |
| 17:10.44 | manxpower | [TK]D-Fender: that has got to be the worst kind of VPN |
| 17:11.11 | manxpower | netpro25_: "abrupt termination of calls" is one of the hardest things to figure out in asterisk |
| 17:11.29 | netpro25_ | manxpower: damn |
| 17:11.42 | netpro25_ | manxpower: where should I start |
| 17:12.04 | garymc | Why does putty not show all the debug info? What ssh program can i use in windows will enable me to look back and copy everythig that happened in the Asterisk CLI? |
| 17:12.11 | manxpower | Well, diagram a call. i.e. SIP Phone -> Asterisk -> Digium analog card -> telco or whatever you have. |
| 17:12.35 | manxpower | garymc: use putty. |
| 17:12.43 | garymc | iam |
| 17:12.50 | manxpower | then you can do what you want to do |
| 17:12.57 | garymc | but after so long the first bit of text disapears |
| 17:13.10 | manxpower | garymc: are you using the "screen" program? |
| 17:13.10 | netpro25_ | SIP Phone -> Router -> Internet -> Asterisk Server -> Vitelity |
| 17:13.28 | garymc | where its a black terminal screen |
| 17:13.30 | manxpower | netpro25_: Is your connection to vitelity using SIP or IAX? |
| 17:13.34 | netpro25_ | sip |
| 17:13.44 | *** join/#asterisk rofcorp (n=rfletche@190.140.47.123) |
| 17:13.51 | manxpower | netpro25_: when do the calls terminate? 2 seconds, 30 seconds, random? |
| 17:14.01 | netpro25_ | like 10 minutes |
| 17:14.14 | rofcorp | how to configure Skype for asterisk to make outgoing calls? |
| 17:14.15 | netpro25_ | 5-10 minutes |
| 17:14.30 | netpro25_ | rofcorp: that is a good question, google |
| 17:14.33 | *** join/#asterisk ircd (i=sam@unaffiliated/samferry) |
| 17:14.49 | garymc | sorry yes all the call to voice mail shows from start to finish |
| 17:14.49 | manxpower | netpro25_: do you have canreinvite=no you should not need it since according to your diagram you don't have any NAT involved |
| 17:14.52 | [TK]D-Fender | garymc: set a bigger scrollback buffer |
| 17:14.52 | netpro25_ | rofcorp: as far as I know its not supported |
| 17:14.56 | rofcorp | i already google it, thats why im here. |
| 17:15.22 | netpro25_ | manxpower: router does NAT |
| 17:15.52 | manxpower | rofcorp: nobody here uses skype. any and all skype stuff is proprietary. I know it exists, I know of nobody using it. |
| 17:15.52 | netpro25_ | or are you reffering to the vitelity config |
| 17:16.40 | manxpower | netpro25_: chances are you'll have to enable SIP debug and figure out why it's dropping the call |
| 17:16.52 | iksik | what is a difference between using "=" and "=>" in extensions.conf? |
| 17:16.59 | netpro25_ | manxpower: I have careinvite=no for vitelity |
| 17:17.05 | manxpower | iksik: nothing. it's cosmetic |
| 17:17.10 | netpro25_ | manxpower: so I can get rid of that |
| 17:17.29 | manxpower | netpro25_: the symptom of needing canreinvite=no is calls dropping before the 60 second mark |
| 17:17.36 | netpro25_ | ah ok |
| 17:17.37 | iksik | manxpower, oh, ok |
| 17:17.57 | rofcorp | maxpower, so why asterisk developed a module for skype? |
| 17:18.02 | hesco | After returning from an AGI script with a result of 0, the *CLI> is telling me: "Huh? Child handler, but nobody there?" Any ideas what that might mean? |
| 17:18.12 | manxpower | rofcorp: Asterisk did not. Digium did. contact them |
| 17:18.43 | rofcorp | ok, digium. |
| 17:18.43 | manxpower | rofcorp: what is the URL for the sykpe/Asterisk info you found? |
| 17:18.47 | rofcorp | the same thing. |
| 17:18.59 | manxpower | no, they are not the same thing. |
| 17:19.11 | rofcorp | they developed asterisk. |
| 17:19.14 | netpro25_ | http://blogs.digium.com/2009/02/23/skype-for-asterisk-update/ |
| 17:19.23 | manxpower | rofcorp: they develop many other things too |
| 17:19.24 | rofcorp | and now developed the sfa. |
| 17:19.48 | rofcorp | I invite you to try the SFA module, its very interesting. |
| 17:20.52 | rofcorp | the SFA is now released, and commercial. that post is old. |
| 17:21.17 | manxpower | there you go |
| 17:22.59 | rofcorp | ok, maybe you could help me. |
| 17:23.15 | rofcorp | the syntax to make an outgoing call using SFA is |
| 17:23.16 | rofcorp | Dial(Skype/[<originator>@]<destination>) |
| 17:23.27 | manxpower | awesome |
| 17:23.40 | rofcorp | how i can make a call file using this command? |
| 17:24.24 | manxpower | I'd have to read the docs, but I think it would be Channel: Skype Data: restofthestuff |
| 17:25.00 | rofcorp | ok, i will try it. |
| 17:28.36 | *** join/#asterisk retentiveboy (n=pdugas@74-95-28-37-Atlanta.hfc.comcastbusiness.net) |
| 17:28.51 | *** part/#asterisk rofcorp (n=rfletche@190.140.47.123) |
| 17:30.46 | garymc | [TK]D-Fender : heres the sip debug call http://pastebin.ca/1563338 |
| 17:30.51 | garymc | what you make of this? |
| 17:32.26 | garymc | again made a call from extension 202 to 201. 201 is in theoffice 202 is at my house. The call was answered we both spoke but couldnt hear each other. ended clal thats the output above |
| 17:32.54 | [TK]D-Fender | garymc: ensure that each device has "canreinvite=no" |
| 17:33.34 | garymc | each deive, as in freepbx extension thingy |
| 17:33.53 | manxpower | garymc: you'll have to translate from Asterisk terms to FreePBX terms. |
| 17:35.38 | garymc | [TK]D-Fender im pretty sure they both dont but im double checking |
| 17:35.57 | manxpower | Asterisk terms: Add canreinvite=no in the sip.conf entry for the device that is behind NAT. FreePBX Terms: click on the toaster looking thingy. |
| 17:38.55 | garymc | [TK]D-fender : All extensions are callreinvit=no |
| 17:39.03 | garymc | callreinvite=no |
| 17:39.29 | [TK]D-Fender | garymc: What do you have forwarded to *? |
| 17:39.39 | garymc | what you mean |
| 17:39.59 | garymc | ports? |
| 17:40.45 | garymc | [TK]D-Fender : I dont know what you mean "what do i have forwarded?" |
| 17:41.01 | [TK]D-Fender | PORTS |
| 17:41.17 | garymc | yeah ports are jeez the ones you gave me last night |
| 17:41.43 | garymc | cant remeber off the top off my head, udp 5060 |
| 17:41.49 | garymc | 4562 or something |
| 17:41.56 | garymc | and 10000-20000 |
| 17:41.59 | garymc | all udp |
| 17:43.10 | hesco | "Huh? Child handler, but nobody there?" I'm guessing no one here has any ideas what that might mean? |
| 17:43.58 | garymc | [TK]D-Fender : You think it could be my ports on the router not opened properly? As I been back the office today to make sure they were all there |
| 17:45.47 | [TK]D-Fender | garymc: Test Zoiper directly and only to voicemail. Do you hear the prompts? does it record you? |
| 17:46.06 | garymc | Ive tested . I hear nothing and it doesnt record me |
| 17:46.12 | garymc | it works when im in the office |
| 17:46.19 | garymc | with my laptop i have here now |
| 17:46.36 | garymc | as soon as im off site though it still registers but no sound |
| 17:46.58 | manxpower | ~answers |
| 17:46.59 | infobot | from memory, 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 |
| 17:47.03 | garymc | I can call extensions make them ring but silence all the way. I do hear a ringing sound in the softphone though |
| 17:47.04 | manxpower | garymc: read the NAT link |
| 17:48.12 | p3nguin | If I dial from one SIP phone to another SIP phone and begin a conversation, can Asterisk be turned off completely (computer powered off) and the two phones will continue to communicate? |
| 17:49.42 | manxpower | p3nguin: no. |
| 17:49.43 | garymc | p3nguin : no thats a question lol |
| 17:49.51 | garymc | even i know that wont work |
| 17:49.54 | *** join/#asterisk twanny796 (n=chatzill@195.158.126.145) |
| 17:50.22 | p3nguin | What does it mean for Asterisk to be "out of the media path?" |
| 17:50.36 | manxpower | p3nguin: they will continue to communicate (if the call is reinvited) until a SIP message has to be sent then all hell will break lose. |
| 17:51.38 | *** join/#asterisk twanny796 (n=chatzill@195.158.126.145) |
| 17:52.19 | twanny796 | zaptel driver for 2.4.34 ? |
| 17:53.11 | garymc | [TK]D-Fender : Did you see anything else in my debug? |
| 17:54.14 | garymc | my sip.conf is controlled by freepbx. It tells me not to touch anything in there? |
| 17:54.29 | [TK]D-Fender | garymc: Check firswalls, double check forwarding, and your provider could be interfering |
| 17:54.35 | garymc | It says to use sip_custom.conf? |
| 17:54.46 | manxpower | garymc: ask on #freePBX they know where what goes |
| 17:55.10 | manxpower | ~freepbx |
| 17:55.12 | 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 |
| 17:55.15 | manxpower | I don't need to remind you ./ |
| 17:55.42 | garymc | [TK]D-Fender : Ive triple checked the ports and firewalls. I also opened ports for ftp on another server (works fine) and ssh on the asterisk box. They where TCP ports though and not UDP |
| 17:56.38 | garymc | I know manxpower.... I dont know what id be asking them though because id only be using the gui to set the sip.conf anyway |
| 17:56.41 | [TK]D-Fender | garymc: pastebin your sip.conf and everything linked to is masking only passwords |
| 17:57.00 | garymc | ok |
| 17:57.16 | manxpower | garymc: it's not going to work if you are only opening up TCP |
| 17:57.54 | garymc | no you misunder stand me manxpower I only opened ssh port 22 TCP and the other ports for SIP where UDP |
| 17:57.58 | *** join/#asterisk chandoo (n=chandoo@ool-4353b978.dyn.optonline.net) |
| 17:58.15 | manxpower | garymc: ports 5060/UDP and 10000-20000/UDP? |
| 17:58.33 | garymc | manxpower : yep that exactly |
| 17:58.38 | manxpower | good |
| 17:59.42 | netpro25_ | garymc: what is 10k - 20k used for? |
| 17:59.49 | netpro25_ | ports* |
| 18:00.03 | garymc | SIP calls I beleive |
| 18:00.23 | netpro25_ | that explains my QOS problem |
| 18:00.28 | manxpower | AUDIO! |
| 18:00.39 | netpro25_ | did not set those ports for QOS priority |
| 18:00.42 | manxpower | Audio uses RTP, signaling uses SIP |
| 18:01.19 | garymc | [TK]D-Fender my sip.conf doesnt make sense to me, there isnt much in it really, heres the pastebin http://pastebin.ca/1563374 |
| 18:01.42 | garymc | Maybe my RTP.conf file is wrong? |
| 18:01.57 | [TK]D-Fender | 13:56]<[TK]D-Fender>garymc: pastebin your sip.conf and everything linked to is masking only passwords <--------------- |
| 18:02.00 | [TK]D-Fender | garymc: and that too |
| 18:04.51 | p3nguin | The correct way to express port range 10000-20000 in iptables is "--dport 10000:20000", right? |
| 18:06.10 | *** part/#asterisk jimw (n=chatzill@adsl-70-41-228.gnv.bellsouth.net) |
| 18:07.30 | p3nguin | garymc: It could be. By default (without specifying the ports), the port range is 5000 - 31000. |
| 18:07.33 | garymc | [TK]D-Fender : This is all the files with stuff in them. Title just above each file. Its all as one pastebin. Hope this is ok http://pastebin.ca/1563378 |
| 18:07.58 | *** join/#asterisk seanmh (n=johndoe@c-69-254-131-168.hsd1.nm.comcast.net) |
| 18:08.22 | garymc | p3nguin : Im not sure i just opened those ports in my Mac Airport Router in the office |
| 18:08.31 | garymc | The ones I was told too |
| 18:08.41 | p3nguin | Your rtp.conf specifies 10000-20000, so you're good there. |
| 18:09.05 | [TK]D-Fender | garymc: Do you have forwarding on your remote side? |
| 18:10.26 | garymc | you mean in my router at home? |
| 18:10.56 | p3nguin | Do you know the difference between local and remote? |
| 18:12.10 | manxpower | forwarding only has to be set up on the router that provides NAT to the Asterisk server |
| 18:12.42 | p3nguin | Is your asterisk server at home? If not, then you don't need to keep asking about your router at home. |
| 18:12.43 | garymc | I dont have forwarding setup on my home router as i thought that defeats the object |
| 18:12.57 | garymc | only at the office on the router connected to the asterisk |
| 18:14.12 | garymc | Wow I cant beleive it i think i have the [TK]D-fender stuck for words with this one ;P |
| 18:14.35 | p3nguin | s/stuck/annoyed/ |
| 18:14.58 | garymc | p3nguin : you confuse me |
| 18:15.11 | garymc | where are you from? |
| 18:15.29 | p3nguin | USA |
| 18:15.32 | [TK]D-Fender | garymc: He's from Sed... |
| 18:15.45 | [TK]D-Fender | garymc: Its a nice place to visit but they won't let you live there |
| 18:16.00 | *** join/#asterisk netpro25_ (n=mmanning@64-238-176-253.ksg.apt.gru.net) |
| 18:16.01 | [TK]D-Fender | garymc: And no, I'm not sure what the issue is right now. |
| 18:16.04 | garymc | i dont know many places in the us |
| 18:16.12 | netpro25_ | is there a way to debug SIP for a particular IP? |
| 18:16.16 | [TK]D-Fender | garymc: "iptables --list" <- from your server Linux CLI |
| 18:16.27 | garymc | ok hold on |
| 18:16.33 | [TK]D-Fender | p3nguin: Lost on this one... |
| 18:16.50 | p3nguin | What? |
| 18:18.47 | garymc | it doesnt show anythign i dont think [TK]D-Fender i typed "iptables --list" in the server cli and it returned this http://pastebin.ca/1563392 |
| 18:18.53 | p3nguin | You're lost in his problem or I was lost in something that I wasn't paying attention to? |
| 18:19.26 | p3nguin | You have no firewall rules! |
| 18:19.34 | garymc | I think its the only genuine problem ive come here with in 2 weeks, and ive got him stumped ;P |
| 18:19.53 | garymc | its gonna be something stupid though i know, something simple |
| 18:19.56 | p3nguin | Is that firewall on the asterisk box or a separate computer? |
| 18:20.00 | [TK]D-Fender | garymc: What router are you using at the office? |
| 18:20.17 | garymc | I have a BT one with DMZ on for now |
| 18:20.17 | *** join/#asterisk skymeyer (n=skymeyer@mailout.itconnect.be) |
| 18:20.21 | garymc | then i have |
| 18:20.38 | *** join/#asterisk rj45 (n=rj45@c-24-0-166-121.hsd1.pa.comcast.net) |
| 18:20.48 | drmessano | DMZ = fail |
| 18:20.51 | garymc | a Macintosh TIMECapsule (Airport) router pluged into that giving all the comps their internet |
| 18:20.55 | [TK]D-Fender | garymc: "BT one" says nothing |
| 18:20.57 | p3nguin | garymc: Is that firewall on the asterisk box or a separate computer? |
| 18:21.07 | [TK]D-Fender | garymc: Your core netwoking is now suspect |
| 18:21.29 | garymc | [TK]D-Fender its our providers own make. But im not using that for port forwarding |
| 18:21.40 | [TK]D-Fender | garymc: and if you are even using WiFi from * to your core networking, then that must first be removed. |
| 18:21.51 | garymc | im not using wifi |
| 18:21.56 | p3nguin | If it is at the perimeter, you HAVE TO use it for forwarding of some type. |
| 18:22.08 | *** join/#asterisk antoranz (n=quassel@190.157.129.60) |
| 18:22.09 | garymc | I open ports on the TimeCapsule |
| 18:22.15 | garymc | works for SSH and FTP |
| 18:22.15 | [TK]D-Fender | garymc: then your forwarding may be worthless. You could be block BY that device |
| 18:22.20 | antoranz | Hi, guys! |
| 18:22.26 | garymc | which one the BT one? |
| 18:22.30 | [TK]D-Fender | yes |
| 18:22.35 | p3nguin | I'm sure that your router at the perimeter is providing NAT. |
| 18:22.35 | antoranz | I'm doing basic experiments with asterisk/SIP |
| 18:22.47 | garymc | Ive removed firewall on that device |
| 18:22.54 | antoranz | I tried with kphone and I'm able to connect but I can't with ekiga |
| 18:23.10 | antoranz | in tshark I see a 405 error when registering |
| 18:23.10 | [TK]D-Fender | garymc: My trust level is appropriately low |
| 18:23.24 | [TK]D-Fender | garymc: And teh device could simply be screwing with things |
| 18:23.30 | garymc | [TK]D-Fender : Trust of me? |
| 18:23.53 | garymc | yes possibly but, ssh and ftp are working great |
| 18:24.00 | [TK]D-Fender | garymc: of what steps you've taken or that they should be taken as being able to affect the outcome. |
| 18:24.02 | netpro25_ | Anyway to debug the RTP packets? |
| 18:24.11 | [TK]D-Fender | garymc: Yes, and shit routers deal with TCP better |
| 18:24.29 | antoranz | I followed these instructions but I get the same error: http://wiki.ekiga.org/index.php/Ekiga_as_an_Asterisk_client |
| 18:24.39 | garymc | right ok, well its the MAC device thats dealing with TCP and UDP |
| 18:25.14 | garymc | but you saying the BT HUB thats plugged into my main broadband socket would be stopping the UDP cos it is shitty? |
| 18:25.43 | garymc | even though It has firewall etc all turned off? |
| 18:26.42 | *** join/#asterisk SirColin (n=SirColin@my83-216-68-241.mynow.co.uk) |
| 18:27.28 | p3nguin | MAC device? I think devices have MACs, but MACs aren't devices. |
| 18:27.46 | garymc | p3nguin : I meant Macintosh |
| 18:27.53 | p3nguin | oh, Mac. |
| 18:28.02 | p3nguin | MAC is something else when dealing with networking. |
| 18:28.02 | *** join/#asterisk galeras (n=galeras@186.82.239.70) |
| 18:28.05 | garymc | TIME CAPSULE : |
| 18:28.09 | garymc | yeah sorry about that |
| 18:28.39 | garymc | ok [TK]D-Fender so we come to the conclusion my Broaband router is the problem? |
| 18:28.56 | p3nguin | I think I would start over, removing any stupid DMZ settings, then correctly configure any port forwardings that is necessary. |
| 18:28.58 | garymc | The Telco companys own one |
| 18:29.17 | [TK]D-Fender | garymc: A conclusion might come when you rip it out and confirm that it is the culprit |
| 18:29.25 | [TK]D-Fender | garymc: right now it is just highly suspect |
| 18:30.02 | garymc | Ok so it could be the settings on the BT Business hub, thats what they call it. They are the biggest Telco company in uk, infact the own all lines |
| 18:30.21 | garymc | so im sure others are using this |
| 18:30.26 | garymc | Ok gotta take missus to work |
| 18:30.29 | garymc | be back later |
| 18:30.31 | garymc | bye |
| 18:37.25 | drmessano | DMZ on a Comcast SMC gateway works well to put a router behind it, not so much other devices |
| 18:39.12 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 18:39.48 | *** join/#asterisk retentiveboy (n=pdugas@74-95-28-37-Atlanta.hfc.comcastbusiness.net) |
| 18:41.51 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 18:50.57 | digilink | hi all, I'm having some trouble with setting up an OpenVOX TDM400 card. I have an FXO card in slot 1, and an FXS card in slot 2. However, when I run dahdi_genconf, it populates /etc/dahdi/system.conf with fxsks=1 and fxoks=2. Shouldn't it be the other way around? |
| 18:52.02 | *** join/#asterisk sercik (n=ciccio@host93-28-dynamic.52-79-r.retail.telecomitalia.it) |
| 18:52.18 | sercik | good day to all channel! |
| 18:52.37 | sercik | i need to open some port to make my asterisk available from outside? |
| 18:52.43 | Tim_Toady | digilink its correct, fxo card has fxs signaling and the other way arround |
| 18:52.59 | digilink | oh ok |
| 18:53.07 | digilink | I misunderstood how it worked then |
| 18:53.44 | digilink | now for whatever reason I can't get dahdi to load with asterisk |
| 18:54.00 | sercik | dahdi is one asterisk is another |
| 18:54.19 | russellb | asterisk must know that you have a cheap clone card in your machine :-p |
| 18:54.38 | digilink | well it previously worked fine |
| 18:54.40 | *** join/#asterisk eXistenZ (n=pectic@unaffiliated/existenz) |
| 18:54.57 | digilink | I was trying to make the FXS port work and thats when the problems started :( |
| 18:55.02 | eXistenZ | Hello, is it possible to connect asterisk to some electric switch system? |
| 18:55.58 | sercik | I'm already disliked by everyone on this channel? |
| 18:56.06 | netpro25_ | So I am losing audio after some time in a conversation but the SIP packets are still there and they indicate everything is fine but the phone goes dead as far as audio is concerned. Connection is maintained though. |
| 18:56.36 | Tim_Toady | sercik u need atleast 5060/udp for sip |
| 18:56.45 | sercik | ty Tim_Toady |
| 18:56.51 | netpro25_ | This is happening randomly after at least 10 minutes |
| 18:57.00 | Tim_Toady | and some more ports for rtp, check ur asterisk/rtp.conf |
| 18:57.46 | digilink | oh crap, I just realized my phone cord was unplugged from the card |
| 18:57.53 | digilink | lol FXO is working again |
| 18:58.40 | sercik | Tim_Toady to receive calls from another pc with a softphone installed udp is enough? |
| 18:59.10 | Tim_Toady | yes |
| 18:59.18 | *** join/#asterisk maour (n=gnu@unaffiliated/maour) |
| 18:59.19 | sercik | thanks again :) |
| 19:00.30 | sercik | if someone with a softphone wants to call a client connected to my asterisk which syntax is needed? |
| 19:00.55 | *** join/#asterisk sezuan (i=sezuan@mobil5.vp.ip6.scheff32.de) |
| 19:01.22 | *** join/#asterisk TJNII (n=TJNII@207.189.199.58) |
| 19:01.29 | sercik | P.S. softphone in another place of world... |
| 19:03.22 | *** join/#asterisk phunyguy (n=phunyguy@h69-130-69-136.kgldga.dsl.dynamic.tds.net) |
| 19:03.48 | drmessano | user@host |
| 19:03.58 | drmessano | and allowguest |
| 19:04.28 | Tim_Toady | well, that might work but its dangerus :P |
| 19:04.29 | *** join/#asterisk TJNII (n=TJNII@207.189.199.58) |
| 19:04.37 | drmessano | dangerous? |
| 19:05.00 | Tim_Toady | allowing guests is dangerus |
| 19:05.30 | drmessano | Only if you're dialplan sucks |
| 19:05.33 | drmessano | your* |
| 19:06.21 | sercik | ok |
| 19:06.28 | Tim_Toady | its better to have more layers of security than just the dialplan |
| 19:07.48 | drmessano | If your setup is that fragile that allowguest poses a risk, you shouldn't be using Asterisk |
| 19:08.03 | sercik | thank you to all for reply |
| 19:08.18 | sercik | maybe i was not clearly... |
| 19:08.18 | Tim_Toady | anyway thats a long talk |
| 19:08.44 | drmessano | If you have any thoughts of a utopian world of SIP <> SIP calling , you need to be able to allow anonymous calls |
| 19:08.51 | sercik | i mean: i want to receive calls not only from analog line but also from other voip client |
| 19:09.06 | drmessano | sercik: Understood you just fine |
| 19:09.17 | drmessano | Other phone calls USER@HOST on your box |
| 19:09.31 | drmessano | allowguest=yes |
| 19:09.33 | drmessano | Done |
| 19:10.03 | *** join/#asterisk war9407 (i=war@liquidswords.org) |
| 19:10.25 | sercik | perfect.. only it seems strange to me that this is not normal.. so the right for you is to receive calls only from analog?? |
| 19:11.27 | drmessano | The default is to not allow anonymous SIP calls, which is perfectly sane as a default.. |
| 19:11.42 | Tim_Toady | no, i mean to set sip users with passwords etc for ur sip clients |
| 19:12.02 | *** join/#asterisk JonCup (n=JonCup@72.34.90.74) |
| 19:12.16 | JonCup | Any one here tried the asterisk voice changer? |
| 19:12.31 | Tim_Toady | sercik u better start reading this: http://www.asteriskdocs.org/ |
| 19:12.51 | drmessano | He doesnt want it as an extension with no auth |
| 19:12.58 | sercik | i can assure you that i'm reading much... |
| 19:13.12 | sercik | but i also ask here :) |
| 19:13.15 | *** part/#asterisk eXistenZ (n=pectic@unaffiliated/existenz) |
| 19:13.38 | sercik | at least.... until noone kick me ;) |
| 19:14.05 | [TK]D-Fender | kicks sercik |
| 19:14.13 | drmessano | There is a big difference between allowing anonymous calls and giving unauthed access to your full dialplan |
| 19:14.56 | sercik | i'm so much nagging ? :( |
| 19:15.19 | [TK]D-Fender | sercik: so go set up another SIP client |
| 19:16.03 | sercik | do you mean to receive calls from outside? |
| 19:17.23 | hesco | <PROTECTED> |
| 19:17.24 | [TK]D-Fender | sercik: You are the one being vague about what you want to do |
| 19:17.49 | [TK]D-Fender | hesco: Maybe * has them LOCKED |
| 19:17.59 | [TK]D-Fender | hesco: can ROOT remove them? |
| 19:18.08 | hesco | yes, root can |
| 19:18.24 | hesco | any idea how to unlock them? |
| 19:18.39 | sercik | maybe the file are created with wrong permission? |
| 19:19.24 | hesco | ok, running another test to see. |
| 19:19.34 | *** join/#asterisk viq (n=viq@unaffiliated/viq) |
| 19:21.35 | hesco | indeed they are being created as root:root, not root:dialer as I had thought. |
| 19:21.52 | sercik | so dialer can't delete it |
| 19:21.55 | hesco | The directory has perms: drwxrwsr-x 2 root dialer 80 2009-09-12 15:20 outgoing |
| 19:22.31 | hesco | doesn't that sticky bit mean files created in that directory are supposed to inherit the group ownership of the directory itself? |
| 19:23.06 | TJNII | What is creating the files |
| 19:23.38 | hesco | a module being run by an agi script being run by asterisk which is running as root here |
| 19:24.39 | TJNII | Solution one: Run * as asterisk:asterisk or asterisk:dialer. Solution two: put a chgrp in your script. |
| 19:26.15 | hesco | just for general principal, I'd love to have asterisk run as other than root. How involved is that? |
| 19:26.24 | TJNII | Not very. |
| 19:26.24 | *** join/#asterisk Deeewayne (n=dwayne@c-71-228-179-90.hsd1.al.comcast.net) |
| 19:26.24 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 19:26.50 | hesco | I ssupect having my script do the chgrp will prove more practical for meeting my current deadline, though. |
| 19:26.51 | TJNII | There are options for it in asterisk.conf |
| 19:27.17 | TJNII | But you have to chown all the files if you've been running it as root |
| 19:28.34 | hesco | I'm guessing this is what I'm looking for? ;astctlowner = root and ;astctlgroup = apache |
| 19:29.15 | hesco | 'all the files', does that mean everything owned by root at /usr/local/asterisk-1.6 ??? |
| 19:30.05 | TJNII | I was wrong. Those are for the control file. The uid and gid options are passed to * on the command line. They go in the init scrips on my Debian box. |
| 19:34.17 | hesco | I'm also running debian, though my * is built from source. /etc/init.d/asterisk includes: #AST_USER="asterisk" and #AST_GROUP="asterisk" and a weighty warning about 'substantial work' on the environment, which I guess explains why I have not done this yet |
| 19:34.55 | *** join/#asterisk cosmicwombat (n=cosmicwo@69.7.44.68) |
| 19:35.05 | TJNII | Pfft. The substantial work is just a bunch of chowns. |
| 19:39.24 | hesco | logs, spools, binary (perhaps). anything else? |
| 19:41.07 | TJNII | Look at the directories listed in asterisk.conf |
| 19:41.17 | hesco | thanks. |
| 19:41.22 | TJNII | That, if I am not mistaken, should tell you all of them |
| 19:42.14 | hesco | I'm about ready to test the chown on my .call files in the spool. If this works, the 'substantial work' can wait until after I deliver this dialer application. |
| 19:42.26 | hesco | But its long been on my wish list |
| 19:50.22 | *** join/#asterisk tamiel (n=tamiel@ip-7.net-81-220-254.rev.numericable.fr) |
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| 19:56.54 | *** join/#asterisk ManxPower (n=EWieling@63.sub-70-222-111.myvzw.com) |
| 19:57.21 | ManxPower | anyone know of a male that does voice prompts like Allison does? |
| 19:58.07 | box2 | Graham Bell |
| 19:58.19 | box2 | teehee |
| 19:58.38 | box2 | that would actually be neat |
| 19:59.07 | *** join/#asterisk puzzled (n=foobar@53533AB2.cable.casema.nl) |
| 20:04.20 | ManxPower | once you figure out the right the right search terms, there's plenty of info out there. |
| 20:04.26 | ManxPower | asterisk male voice |
| 20:08.10 | netpro25_ | I think I found my SIP problem. Seems as though SIP is causing my routers CPU usage to peak. |
| 20:08.14 | *** join/#asterisk _ShrikE (n=_ShrikE@74.185.215.60) |
| 20:08.26 | netpro25_ | Anyone experience this? |
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| 20:09.29 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 20:12.10 | sercik | hey! i have aloowguest=yes in my sip.conf but i obtain this error when i try to call with an unregistered softphone: |
| 20:12.10 | sercik | Call from '' to extension 'clark' rejected because extension not found. |
| 20:12.30 | sercik | i use this command sip:clark@10.0.0.1 |
| 20:12.48 | sercik | the same command works perfectly when i'm registered to asterisk. |
| 20:13.16 | sercik | 10.0.0.1 is the ip running asterisk |
| 20:16.40 | ManxPower | netpro25_: your router sucks. 8-| |
| 20:16.58 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 20:17.18 | ManxPower | I have an older Linksys router that reboots if you have too many connections open. |
| 20:17.33 | box2 | sercik: you have to have allowguest and your dailplan in the same context |
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| 20:19.13 | box2 | such as [default] or [conference_room] |
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| 20:25.34 | riddlebox | do any of you guys use the flash operator panel? |
| 20:28.33 | TJNII | hates flash |
| 20:29.56 | *** join/#asterisk matata (n=bassem@Wikipedia/Bassem-JARKAS) |
| 20:32.09 | ManxPower | riddlebox: In my experience long running Flash applications tend to be unstable. |
| 20:32.41 | ManxPower | The biggest problem I found with it is maintaining all the information in the config of FOP for adds/moves/changes |
| 20:32.42 | riddlebox | is there any other operator panel type application? |
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| 20:37.31 | sercik | good night to everyone, i go to bed. |
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| 23:28.44 | momelod | greetings channel |
| 23:29.21 | momelod | can anyone tell me if the rxgain and txgain settings have any effect over T-1 interfaces or are these settings only for FXO ports? |
| 23:46.48 | _ShrikE | momelod: yes they do |
| 23:54.48 | momelod | _ShrikE: oh |
| 23:55.25 | momelod | can i specify a negative value? i ask b/c im hearing my own voice echo back and me and the person on the other end says my voice is very load |
| 23:55.37 | _ShrikE | yes |
| 23:55.53 | momelod | any idea what the valid range of values is? |
| 23:56.06 | momelod | i cant find an example online using negative values |
| 23:56.56 | momelod | the person on the other end doesn't report hearing any echo |