| 00:00.30 | *** join/#asterisk vinhdizzo (~vinh@dhcp-053179.ics.uci.edu) |
| 00:03.50 | leifmadsen | huh |
| 00:03.51 | leifmadsen | odd |
| 00:03.58 | leifmadsen | and it does it on 1.6.2? |
| 00:04.06 | leifmadsen | (that could be a bug you should file) |
| 00:04.27 | Kris2k | I think this bug has been persistant since ilbc got pulled-out of the src tree |
| 00:04.30 | Kris2k | so 1.2.x ? |
| 00:04.47 | Kris2k | or was it 1.4.x ? dont remember |
| 00:07.01 | Kris2k | http://forums.digium.com/viewtopic.php?t=19830 |
| 00:07.03 | Kris2k | interesting |
| 00:07.17 | Kris2k | I guess I'll file a bug |
| 00:11.50 | pdtpatrick | Question - if asterisk was to crash .. which of the log files would have that in? /var/log/asterisk/??? |
| 00:15.46 | *** join/#asterisk ariel_ (~chatzilla@pdpc/supporter/active/abatista) |
| 00:16.20 | Kris2k | you might want to look at /var/log/asterisk/full (providing its configured) which will contain some hints |
| 00:22.04 | scalex000 | hello, do you know how to add a tag in sip hearder |
| 00:22.39 | Kris2k | scalex000: core show application SIPAddHeader |
| 00:22.47 | scalex000 | ok |
| 00:23.30 | *** join/#asterisk dmz (~dmz@64.203.235.49.dyn-cm-pool-34.pool.hargray.net) |
| 00:30.00 | scalex000 | do you know which field is used to send URI MAP |
| 00:32.14 | Kris2k | scalex000: sorry, no... the last thing I vaguely remember is a VXML_URL that can be defined for VXML aware phones to push-them a page when the phone receives a call. |
| 00:33.39 | scalex000 | kris2k,ok, I have problem when I dial from asterisk to nortel, not recognize this number is for private network, nortel think is a public number |
| 00:34.23 | Kris2k | scalex000: ah so you're trunking through a nortel system to get onto the pstn? |
| 00:34.45 | scalex000 | Kris2k, no |
| 00:36.05 | Kris2k | scalex000: whats the call flow? |
| 00:36.09 | scalex000 | Kris2k, I try setup all I need in Nortel, but I when I put the number in private receive # not in public, the nortel dont know which extension Im calling |
| 00:36.48 | scalex000 | Kris2k, I mean I set all setting in nortel |
| 00:36.51 | Kris2k | scalex000: normally in nortel world, you're supposed to define a virtual tie line for every single DN you would have in the system, since it will not expose its internal extensions to a public trunk |
| 00:37.29 | scalex000 | Kris2k, yes target line |
| 00:37.36 | Kris2k | I must say, last time I played with a nortel, it was about 10 years ago when trunking two BCM boxes via H323 |
| 00:37.59 | scalex000 | Kris2k,I try it but I can't so I setup with sip |
| 00:38.13 | Kris2k | and it was important to define a virtual tie-line for each DN, since it was the only way to expose the internal DNs of the system to a trunk |
| 00:38.41 | Kris2k | is your asterisk connected to the nortel system via a trunk interface, or an extension interface? |
| 00:39.06 | scalex000 | Kris2k, the problem is when I dialed from asterisk to nortel they think this call come from public |
| 00:39.26 | scalex000 | Kris2k, VOIp Trunk |
| 00:40.46 | scalex000 | Kris2k, public network 10 digit, private network length 4 diget |
| 00:40.51 | scalex000 | Kris2k, public network 10 digit, private network length 4 digit |
| 00:41.25 | *** join/#asterisk Smirker (~x@14-202-69-225.static.tpgi.com.au) |
| 00:41.33 | Kris2k | scalex000: right, and did you expose your extension through a virtual tie-line, and try calling it? It should work |
| 00:42.23 | scalex000 | Kris2k, when u say virtual tie-line what its mean |
| 00:42.45 | Smirker | ok guys. i'm using ReceiveFAX under res_fax_spandsp over ulaw (tried alaw too). if i call the number, i hear the fax noises. if i try to fax to the number, i get the following error after about 5 seconds: WARNING[20397]: res_fax.c:1474 receivefax_t38_init: channel 'SIP/trunk-00000000' timed-out during the T.38 negotiation. |
| 00:42.59 | Smirker | googled it to hell. hitting my head against the table. |
| 00:43.01 | Smirker | any ideas? |
| 00:43.05 | Kris2k | scalex000: its a relationship (normally used to map a DID to an extension) |
| 00:43.17 | scalex000 | yes |
| 00:43.27 | scalex000 | Kris2k, target line |
| 00:43.40 | Kris2k | okay so, from asterisk, call a target line |
| 00:43.47 | scalex000 | Kris2k, target line, have to option to put public received number, and private |
| 00:44.02 | Kris2k | scalex: no clue, been too long |
| 00:44.06 | Kris2k | try public |
| 00:44.13 | scalex000 | Kris2k, on public work |
| 00:44.25 | scalex000 | Kris2k, but on private nop |
| 00:44.42 | Kris2k | figured, because the trunk interface is exposed on the 'public' side of the PBX |
| 00:45.19 | *** part/#asterisk ariel_ (~chatzilla@pdpc/supporter/active/abatista) |
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| 00:45.39 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 00:45.56 | scalex000 | Kris2k, this mean they program incorrect |
| 00:46.33 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
| 00:46.55 | Kris2k | scalex000: I'm no nortel expert, just did it once with H323 10+ years ago |
| 00:47.11 | scalex000 | Kris2k, ok |
| 00:48.35 | scalex000 | Kris2k, so do you suggest to use h323 not sip |
| 00:48.52 | Smirker | sooo. this fax question hey! |
| 00:55.32 | *** join/#asterisk avb (~avb@190.122.106.167) |
| 00:56.13 | avb | guys, still fighting with SS7 |
| 00:56.35 | avb | using chan_ss7, monitor see audio in the channel, but client gets dead air |
| 00:57.03 | avb | with both zaptel and dahdi |
| 00:57.28 | avb | no audio, but channel is up |
| 00:58.33 | scalex000 | Kris2k, by the way do you know how to send a password? |
| 01:03.33 | *** join/#asterisk adolfomaltez (~taro@190.86.150.215) |
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| 01:16.09 | *** part/#asterisk CRCinAU (~CRCinAU@2002:cb38:f71b:1:badb:adc0:ffee:1) |
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| 01:16.26 | CRCinAU_ | I'm tinkering with the call queues in Asterisk 1.8, and I'm trying to figure out how to ring, say, Phone 1, then after x seconds of no answer etc, ring phone 1 & 2, etc |
| 01:16.41 | CRCinAU_ | then maybe after another x seconds, ring phone 1 & 2 & 3 |
| 01:17.17 | CRCinAU_ | I would have thought the penalties in queues.conf would have worked - but it seems like no way to increase the penalty via a queue rule |
| 01:18.04 | CRCinAU_ | ie: penaltychange => 30,+1 |
| 01:20.08 | gruvfunk_away | mchou: "Asterisk 1.8.4.4-1digium1~lucid built by pbuilder @ nighthawk on a x86_64 running Linux on 2011-06-29 14:38:45 UTC" |
| 01:20.45 | mchou | gruvfunk_away: I just rebuilt (on armel) and the problem persists |
| 01:22.08 | *** join/#asterisk nightrid3r (~nightrid3@91.176.50.186) |
| 01:22.57 | mchou | gruvfunk: something funky going on armel maybe |
| 01:23.26 | gruvfunk | ahh |
| 01:23.32 | gruvfunk | arm |
| 01:36.55 | *** join/#asterisk DEAD_BEEF (~DEAD_BEEF@unaffiliated/dead-beef/x-2609176) |
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| 02:17.34 | jeremy_g | everyone sleeping |
| 02:23.41 | leifmadsen | yes |
| 02:26.37 | mchou | gruvfunk: might have found the solution: http://forums.digium.com/viewtopic.php?f=1&t=76909 |
| 02:26.54 | mchou | gruvfunk: compiling it now |
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| 02:43.05 | mchou | gruvfunk: ok, that was it. |
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| 04:47.06 | Antonjs | I'm having an extremely strange problem: A call on an incoming trunk isn't sending (or asterisk isn't receiving) RTP/DTMF if the call is directed to an IVR, but it works great if the call is forwarded to an extension. From what I can see, the SIP negotiation in each case is happening in exactly the same way |
| 04:47.12 | Antonjs | Anyone have any ideas? |
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| 05:07.55 | Wiretap | Antonjs, you need to allow it to ring for a bit before you answer it seems |
| 05:09.06 | Antonjs | I thought that too, but I tried forwarding it through a ring group with a five second timeout to the IVR - still does the same thing |
| 05:09.56 | Antonjs | with the extension added to the ring group, if I pick up before the timeout audio works great, if I leave it to forward to the IVR it still doesn't work |
| 05:11.14 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-focvjxqirbmagons) |
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| 05:31.03 | Antonjs | Aha, if I switch the ring group to use MoH instead of just a ring then it works |
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| 05:34.49 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 05:34.51 | schmidts | good morning |
| 05:38.32 | *** join/#asterisk hrolf (~hrolf@static-host202-61-49-9.link.net.pk) |
| 05:42.13 | vty | schmidts, hola |
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| 06:03.15 | Antonjs | doesn't work forwarding directly through to a conference either |
| 06:03.26 | Antonjs | even with pre-ring and a delay on the incoming trunk |
| 06:09.42 | *** join/#asterisk flyman_ (~flyman_@chello084114147007.7.15.vie.surfer.at) |
| 06:12.50 | *** join/#asterisk irroot (~irroot@197.170.205.237) |
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| 06:32.40 | *** join/#asterisk jsidhu (u1050@gateway/web/irccloud.com/x-suikjeprzbgsprbu) |
| 06:34.17 | jsidhu | can someone help me out please, I've got a Rhino R2T1 with a PRI from AT&T and it just stopped working. AT&T insists the PRI circuit is fine. ... someone set it up long time ago and I've never had to mess around with it.. any starting point/direction would be appreciated |
| 06:47.48 | *** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com) |
| 06:50.51 | schmidts | i HATE isdn pbx :( customer of me having 3 ISDN - Voip atas called the police 5 times but i didnt see any of these calls and it looks like his PBX route these calls in a very different way :( |
| 06:52.18 | irroot | morning shmidts yes old tech has a way of breaking things |
| 06:53.04 | irroot | open front pannel unseen put a screwdriver in and move it about randomly till you see smoke ... leave |
| 06:56.21 | tzafrir_laptop | schmidts, blame the ATAs :-p |
| 07:00.56 | schmidts | i have configured the ATAs they are very fine ;) its the stupid PBX they use |
| 07:02.49 | jsidhu | anyone have experience with the Rhino R2T1-e? |
| 07:03.52 | *** join/#asterisk Farrukh (~Farrukh@202.61.49.9) |
| 07:04.27 | Farrukh | is there any one who can help me with degium cards in UK |
| 07:05.47 | *** join/#asterisk gmaruzz (~gmaruzz@static-217-133-80-112.clienti.tiscali.it) |
| 07:10.04 | jimbo_uk | Hi Farrukh |
| 07:10.12 | jimbo_uk | possibly if it's a quick one |
| 07:10.14 | irroot | jimbo_uk morning mate |
| 07:10.15 | jimbo_uk | whats up |
| 07:10.19 | jimbo_uk | hey dude |
| 07:10.28 | jimbo_uk | i might have found a fly in our ointment! |
| 07:11.11 | jimbo_uk | remember i said we were seeing issues with the hangup event happening when the call was answered? |
| 07:11.29 | jimbo_uk | this causes issues because it's the h event that triggers the NOT_INUSE state |
| 07:11.44 | jimbo_uk | well i found this from a bur report (old admittedly) |
| 07:11.50 | jimbo_uk | The hangup event you're seeing is the Local channel stepping out of the way for a native bridge once the call gets setup with the intended "real" channel for the Agent callback. It's not ideal behavior, but this is how it's been and will be for 1.2 and 1.4. |
| 07:12.24 | jimbo_uk | maybe not relevant because we're not using agent callback, but harks of some reality because of the local channel use. |
| 07:13.24 | *** join/#asterisk tamiel (~tamiel@213.30.183.226) |
| 07:13.27 | jimbo_uk | bweschke Resolution open => not fixable |
| 07:13.31 | jimbo_uk | hmm.... |
| 07:14.24 | irroot | jimbo_uk put a /n on the end of the local channel |
| 07:14.50 | irroot | or process the channel in the h exten ie check if its a local chan if so ignore |
| 07:18.58 | jimbo_uk | ok |
| 07:19.26 | jimbo_uk | can't do this in the h exten because i need to record the 'real' hangup event to reset DEVICE_STATE |
| 07:20.23 | jimbo_uk | let me try the /n |
| 07:23.40 | irroot | yip you can if the channel is not Local/.... skip the devstate |
| 07:23.58 | jimbo_uk | ? |
| 07:24.04 | jimbo_uk | the channel is local :-) |
| 07:24.19 | jimbo_uk | <PROTECTED> |
| 07:24.19 | jimbo_uk | <PROTECTED> |
| 07:24.19 | jimbo_uk | <PROTECTED> |
| 07:24.19 | jimbo_uk | <PROTECTED> |
| 07:24.21 | Farrukh | jimbo can you help me with call progress tones for UK |
| 07:24.32 | irroot | yes but you dialing a SIp channel |
| 07:24.53 | irroot | in the hangup only bother with sip channels |
| 07:24.58 | jimbo_uk | Farruk, yes, but about to go onto conference call - hang about. |
| 07:25.05 | Farrukh | np |
| 07:25.17 | jimbo_uk | go on.... |
| 07:25.28 | irroot | is heading to office in a bit |
| 07:26.33 | jimbo_uk | i can't see it's possible to map the SIP and local channels |
| 07:26.44 | irroot | ah it is there |
| 07:26.59 | jimbo_uk | [ Context 'jptest' created by 'pbx_config' ] |
| 07:27.00 | jimbo_uk | <PROTECTED> |
| 07:27.00 | jimbo_uk | <PROTECTED> |
| 07:27.00 | jimbo_uk | <PROTECTED> |
| 07:27.00 | jimbo_uk | <PROTECTED> |
| 07:27.00 | jimbo_uk | <PROTECTED> |
| 07:27.05 | jimbo_uk | ouch that's messy! |
| 07:27.06 | irroot | ~pb |
| 07:27.06 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 07:27.39 | irroot | ~trixbox |
| 07:27.39 | infobot | [trixbox] SH1TB0X. Basically a CRAPPY, closed-source distro. STAY AWAY! |
| 07:27.42 | jimbo_uk | http://pastebin.com/wjXKfbzq |
| 07:27.52 | jimbo_uk | haha! |
| 07:27.53 | jimbo_uk | I agree |
| 07:27.58 | jimbo_uk | shit stay away! |
| 07:28.32 | jimbo_uk | i do a lot of 'rescues' for peoply using trixbox |
| 07:28.38 | jimbo_uk | nice trick though. |
| 07:29.51 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:61b5:5c78:7631:ea3e) |
| 07:32.52 | irroot | yeah fleapbx and trixbox keeps us busy .... our distro goes in and live in 15min for format install and 10-20min for config less if autoconfig phones |
| 07:33.28 | jimbo_uk | nice |
| 07:35.49 | Noah7372 | Anyone know any good extensions too add? Like some a radio station w/ music playing, or global news, or something? |
| 07:37.38 | irroot | off for eye test and office ... |
| 07:39.22 | jimbo_uk | cool |
| 07:39.30 | *** join/#asterisk Guizmo (~Guizmo@unaffiliated/guizmo) |
| 07:39.43 | jimbo_uk | \/n seems to work - good find |
| 07:39.59 | jimbo_uk | (christ irc is a piece of crap) |
| 07:40.18 | jimbo_uk | thanks again irroot... |
| 07:41.26 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 07:41.58 | jimbo_uk | Farrukh - hey |
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| 08:11.04 | merlin8282 | Hi all ! I've a question concerning codecs: we have many snom360 and one analog phone. When I enable for example g722 in sip.conf, then the snoms can't call the analog phone anymore. |
| 08:11.25 | Farrukh | jimbo r u there |
| 08:11.45 | merlin8282 | asterisk says WARNING[8801]: channel.c:3210 ast_request_uniqueid: No translator path exists for channel type Zap (native 76) to 4096 |
| 08:11.48 | merlin8282 | and WARNING[8801]: app_dial.c:1256 dial_exec_full: Unable to create channel of type 'Zap' (cause 58 - Bearer capability not available) |
| 08:12.55 | merlin8282 | How can I solve this problem ? |
| 08:16.42 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 08:21.42 | *** join/#asterisk irroot (~irroot@pbx.distrotech.co.za) |
| 08:22.34 | irroot | is at mission control ... |
| 08:25.11 | *** join/#asterisk vikapi (~vikapi@124.125.34.134) |
| 08:25.24 | vikapi | has any1 tried asterisk with x100p?? |
| 08:25.33 | vikapi | using bsnl connection? |
| 08:36.01 | *** join/#asterisk gravin (~gravin@175.139.239.3) |
| 08:38.47 | tzafrir_laptop | vikapi, I suppose it will work |
| 08:38.54 | tzafrir_laptop | do you have any problems? |
| 08:38.56 | vikapi | tryin to configure x100p with bsnl in india!!! using asterisk 1.4.42 |
| 08:39.06 | vikapi | i will paste my current configuration.. |
| 08:39.12 | tzafrir_laptop | vikapi, any reason not to use a newer version? |
| 08:39.26 | tzafrir_laptop | ~pb |
| 08:39.26 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 08:39.42 | vikapi | tried with asterisk 1.6 and dahdi |
| 08:39.56 | vikapi | since didn work, i jus thought of usin different versions.. |
| 08:39.58 | tzafrir_laptop | what specific 1.6 ? 1.6.2 ? |
| 08:40.14 | tzafrir_laptop | Do you actually use Zaptel and not DAHDI? |
| 08:40.46 | *** join/#asterisk arapaho (~arapaho@pierre.infomaniak.ch) |
| 08:41.54 | vikapi | when tryin with asterisk 1.6.x (the most recent release -- dont remember the exact version) |
| 08:42.00 | vikapi | when using 1.6 i used dahdi |
| 08:42.04 | tzafrir_laptop | latest is 1.8 |
| 08:42.05 | vikapi | http://pastebin.com/CQk32hsY |
| 08:42.28 | vikapi | u want me to try in tat?? |
| 08:42.36 | vikapi | can u check my existin conf?? |
| 08:45.03 | vikapi | tzafrir_laptop, any hint??? |
| 08:46.06 | *** join/#asterisk gravin (~gravin@175.139.239.3) |
| 08:48.18 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 08:52.12 | tzafrir_laptop | vikapi, zaptel is outdated |
| 08:52.34 | tzafrir_laptop | The latest version of zaptel is DAHDI. Please don't use Zaptel |
| 08:52.55 | tzafrir_laptop | If this is a new isntall, go with DAHDI and 1.8 (latest) |
| 08:53.37 | irroot | man this is some funny stuff .... guys in the office all playing fishing on there droid phones |
| 08:56.05 | gravin | lol |
| 08:56.25 | vikapi | tzafrir_laptop, i did the install using source compile.. |
| 08:56.30 | irroot | need bigger space someone will get a black eye soon not sure if workmans compensation isurance covers this yet |
| 08:57.00 | vikapi | tzafrir_laptop, update from 1.4.x to 1.8.x is possible straight away?? |
| 08:57.20 | vikapi | tzafrir_laptop, or i screwed up my linux box completely??? :) |
| 08:57.21 | tzafrir_laptop | Is this an existing system or a new install? |
| 08:57.38 | vikapi | it was a new install.. |
| 09:00.53 | vikapi | tzafrir_laptop, do i ve to reinstall the base system? |
| 09:00.55 | vikapi | ?? |
| 09:01.53 | tzafrir_laptop | generally you can install 1.8 on top of 1.4 |
| 09:02.05 | tzafrir_laptop | Shouldn't be a problem |
| 09:05.41 | vikapi | tzafrir_laptop, wat abt dahdi?? |
| 09:05.51 | vikapi | tzafrir_laptop, server is installd with zaptel now.. |
| 09:06.52 | tzafrir_laptop | install dahdi (from source) on top of it |
| 09:07.01 | tzafrir_laptop | (before installing asterisk) |
| 09:07.49 | vikapi | tzafrir_laptop, i dnt ve to install libpri for making x100p work correct? |
| 09:08.12 | tzafrir_laptop | no. libpri is not required for analog cards such as the x100p |
| 09:08.52 | vikapi | ok |
| 09:15.40 | *** join/#asterisk Sertys (~sertys@89.252.247.42) |
| 09:36.41 | *** join/#asterisk lost_soul (~noymfb@cpe-67-249-131-153.twcny.res.rr.com) |
| 09:45.06 | *** join/#asterisk felimwhiteley (~quassel@46.7.101.58) |
| 09:54.02 | *** join/#asterisk felimwhiteley (~quassel@46.7.101.58) |
| 09:57.12 | *** join/#asterisk felimwhiteley (~quassel@46.7.101.58) |
| 10:02.55 | *** join/#asterisk sawgood (~sawgood@173-13-158-27-sfba.hfc.comcastbusiness.net) |
| 10:04.09 | vikapi | tzafrir_laptop, when we give dahdi show channels in CLI, i get no output.. jus check http://pastebin.com/0t52BQGa |
| 10:04.34 | vikapi | is this normal? |
| 10:04.44 | tzafrir_laptop | what's the output of lsdahdi ? |
| 10:04.44 | vikapi | or should it give the channel configured |
| 10:05.02 | tzafrir_laptop | it should at least show "pseudo" |
| 10:05.42 | vikapi | well it's not showin pseudo.. btw, how will be the output of pseudo looks like?? |
| 10:08.41 | Farrukh | anyone sungoma cards |
| 10:08.55 | Farrukh | I need Some advice |
| 10:10.49 | *** join/#asterisk ph8 (~ph8@unaffiliated/ph8) |
| 10:15.14 | *** join/#asterisk vikapi (~vikapi@124.125.34.134) |
| 10:17.25 | *** join/#asterisk TimeRider (~steve@host-92-27-131-175.static.as13285.net) |
| 10:18.40 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-163.hsd1.nj.comcast.net) |
| 10:25.02 | vikapi | tzafrir_laptop, lsdahdi gives |
| 10:25.02 | vikapi | ### Span 1: WCFXO/0 "Wildcard X100P Board 1" (MASTER) |
| 10:25.02 | vikapi | <PROTECTED> |
| 10:25.11 | vikapi | is this normal? |
| 10:25.22 | *** join/#asterisk orn (~orn@rtr1.sh23.sip.is) |
| 10:25.36 | tzafrir_laptop | vikapi, × ×©×××ש×××ת ××§× |
| 10:26.00 | tzafrir_laptop | vikapi, yes, only it's not configured with asterisk |
| 10:26.12 | tzafrir_laptop | now you need a proper chan_dahdi.conf |
| 10:26.20 | tzafrir_laptop | What version of asterisk do you use? |
| 10:26.37 | vikapi | downloading asterisk 1.8.4.. |
| 10:26.54 | tzafrir_laptop | let me know when it is installed |
| 10:26.58 | vikapi | currently using asterisk 1.4.52 |
| 10:27.05 | vikapi | sorry 1.4.42 |
| 10:27.07 | tzafrir_laptop | It is slightly easier to configure than 1.4 |
| 10:27.23 | vikapi | ok.. |
| 10:28.51 | Guizmo | tzafrir_laptop, what about languages ? for fr, I don't see fr sounds in centos 5. Is it placed somewhere else ? |
| 10:29.05 | Guizmo | (for asterisk 1.8) |
| 10:29.49 | tzafrir_laptop | just grab that tarball and put it yourself |
| 10:30.33 | tzafrir_laptop | Note that the Asterisk "fr" are actually Quibeck ones. I was told by some that it is not exactly the same |
| 10:31.05 | tzafrir_laptop | I figure you can find a number of other fr sound sets for asterisk |
| 10:31.13 | Guizmo | ah ok thanks. was still trying to play fr instead of en :) |
| 10:31.21 | Guizmo | so I didn't hear them yet |
| 10:31.26 | tzafrir_laptop | Did you set the language? |
| 10:31.29 | Guizmo | yes |
| 10:31.58 | Guizmo | I tried to replace the en directory with the fr sound, but now nothing is played |
| 10:32.16 | *** join/#asterisk cerberus_za (~coert@41-134-110-162.dsl.mweb.co.za) |
| 10:34.48 | cusco | hi folks |
| 10:35.39 | cusco | when a call comes in: Set(void=${QUEUE_VARIABLES(Lusomundo)}); |
| 10:35.53 | cusco | if(${QUEUEHOLDTIME} > 249){ playback(call-again); } |
| 10:36.06 | cusco | else{ Queue(lusomundo); } |
| 10:36.18 | cusco | now... |
| 10:36.41 | cusco | no calls in queue lusomundo and queueholdtime is always 170 |
| 10:36.43 | cusco | why? |
| 10:37.04 | cusco | :/ |
| 10:38.04 | stix | Hi guys. I have an application running in loop in asterisk. Can I kill it somehow without killing asterisk? |
| 10:57.12 | *** join/#asterisk saxa (~sasa@189.26.255.43) |
| 10:57.23 | vikapi | tzafrir_laptop, installation of asterisk1.8.4 completed. |
| 10:57.27 | vikapi | went well.. |
| 10:58.57 | *** join/#asterisk TimeRider (~steve@host-92-27-131-175.static.as13285.net) |
| 10:59.37 | Lantizia | Lo has anyone here got mysql cdr working using the odbc method? if so I don't suppose you could share your odbc file? tried everything i can think of |
| 11:01.56 | vikapi | tzafrir_laptop, when i do asterisk -vvvvvvvvvvvvvvvvvvvgc it gives me segmentation fault.. core dumped.. |
| 11:02.38 | vikapi | tzafrir_laptop, http://pastebin.com/ZAubyFG7 |
| 11:03.01 | vikapi | tzafrir_laptop, anyidea?? |
| 11:06.16 | *** join/#asterisk irroot (~irroot@pbx.distrotech.co.za) |
| 11:12.15 | tzafrir_laptop | vikapi, try removing all modules from /usr/lib/asterisk/modules and re-run 'make install' |
| 11:15.09 | vikapi | tzafrir_laptop, workd..u r genius.. |
| 11:17.39 | vikapi | tzafrir_laptop, can i use the previous conf of chan_dahdi.conf of 1.4.42? |
| 11:17.55 | Sertys | guys, how do i pass the callerid between two asterisks? |
| 11:18.19 | tzafrir_laptop | vikapi, generally - yes |
| 11:18.23 | Sertys | it's like i keep getting the default callerid(the accountnumber) on the receiving side |
| 11:18.33 | tzafrir_laptop | Sertys, how are those two connected? |
| 11:18.41 | Sertys | well, they're sip peers |
| 11:18.56 | tzafrir_laptop | Just set the callerid |
| 11:19.24 | Sertys | i tried that in so many ways |
| 11:19.49 | Sertys | i dial from agi and use the setcallerid function |
| 11:19.58 | Sertys | tried SET |
| 11:20.36 | Sertys | my receiving side is defaulting to a2billing btw |
| 11:20.36 | tzafrir_laptop | check ${CALLERID(all)} before calling |
| 11:26.12 | vikapi | tzafrir_laptop, i restored the previous configuration, now i get the output http://pastebin.com/7QZFZ17G when doin show channels from CLI. but when i try to call the PSTN line from my mobile, im not gettin any verbose in the CLI!!! |
| 11:26.58 | tzafrir_laptop | vikapi, please don't shout |
| 11:27.06 | tzafrir_laptop | (!!! and such is shouting) |
| 11:27.22 | tzafrir_laptop | core set verbose 3 |
| 11:27.23 | vikapi | tzafrir_laptop, oh no..didn't knw tat.. |
| 11:27.38 | vikapi | tzafrir_laptop, sorry if tat offended u... |
| 11:27.42 | tzafrir_laptop | now, what do you see in the asterisk CLI when a call comes in? |
| 11:27.48 | tzafrir_laptop | no problem |
| 11:28.49 | vikapi | tzafrir_laptop, still nothing in CLI... |
| 11:32.02 | vikapi | http://pastebin.com/9sDYWHzz and tried.. but still nothing.. |
| 11:32.29 | vikapi | tzafrir_laptop, is there a way to check if the signalling and other parameters used are correct??? |
| 11:32.48 | vikapi | tzafrir_laptop, like a test which checks the available signalling.. |
| 11:32.50 | vikapi | ?? |
| 11:33.24 | tzafrir_laptop | What do you see in lsdahdi? |
| 11:36.27 | irroot | idiots policy is that there can be no non windows machines on network so they setting up a windows server with a vm for asterisk |
| 11:36.40 | irroot | please explain why ?? |
| 11:38.35 | vikapi | lsdahdi |
| 11:38.36 | vikapi | ### Span 1: WCFXO/0 "Wildcard X100P Board 1" (MASTER) |
| 11:38.36 | vikapi | <PROTECTED> |
| 11:39.20 | *** join/#asterisk Smirker (~x@60-241-222-233.static.tpgi.com.au) |
| 11:40.34 | tzafrir_laptop | vikapi, are you sure you call to the right number? |
| 11:40.47 | tzafrir_laptop | can you try an outgoing call? |
| 11:41.36 | tzafrir_laptop | If you connect an analog phone instead of that card, and call to htat number, does the phone ring? |
| 11:48.26 | vikapi | tzafrir_laptop, yes.. |
| 11:48.30 | vikapi | it does ring. |
| 11:49.14 | tzafrir_laptop | what does: dahdi show channels show? |
| 11:50.24 | vikapi | infact, to the pass port of x100p, i ve connected a phone, and it is ringing.. |
| 11:50.26 | vikapi | *CLI> dahdi show channels |
| 11:50.26 | vikapi | <PROTECTED> |
| 11:50.27 | vikapi | <PROTECTED> |
| 11:50.27 | vikapi | <PROTECTED> |
| 11:50.49 | vikapi | *CLI> dahdi show channels |
| 11:50.49 | vikapi | <PROTECTED> |
| 11:50.49 | vikapi | <PROTECTED> |
| 11:50.49 | vikapi | <PROTECTED> |
| 11:51.06 | vikapi | oops..http://pastebin.com/i3eZrjvw |
| 11:53.48 | tzafrir_laptop | dialplan show s@default |
| 11:54.01 | *** join/#asterisk gmaruzz (~gmaruzz@static-217-133-80-112.clienti.tiscali.it) |
| 11:58.39 | [netman] | I am using Dial() with 'g' option, but when DIALSTATUS = CANCEL, I only can execute the dialplan from the 'h' extension. How can I make another Dial() then? |
| 11:58.53 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 11:58.53 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 11:59.19 | vikapi | tzafrir_laptop, http://pastebin.com/60za51sc |
| 11:59.50 | tzafrir_laptop | ok |
| 11:59.58 | *** join/#asterisk pk1r (~pk1r@fiber-087-195-226-092.solcon.nl) |
| 12:00.15 | vikapi | now i got an incomin call from a number. and cli showd up.. |
| 12:00.19 | pk1r | hello there. having troubles with iax trunks here |
| 12:00.28 | pk1r | seeing this error: chan_iax2.c:906 jb_warning_output: Resyncing the jb. last_delay -41, this delay -4294901801, threshold 1004, new offset 4294901801 |
| 12:00.31 | tzafrir_laptop | the phone is properly connected, right? You see '(battery)' in lsdahdi |
| 12:00.41 | vikapi | http://pastebin.com/7kbaTzRW |
| 12:01.00 | pk1r | choppy sounds, so guess this error has something to do with it |
| 12:01.20 | tzafrir_laptop | vikapi, so all's well? |
| 12:02.06 | pk1r | asterisk 1.4.39.1/dahdi 2.4.1 using astribank2 hardware |
| 12:02.54 | vikapi | tzafrir_laptop, how unfortunate im..i was all this time tryin to call this analog fone using my mobile fone.. |
| 12:03.10 | vikapi | tzafrir_laptop, when call was made from another analog phone, CLI is showing up.. |
| 12:03.26 | vikapi | tzafrir_laptop, why isn't not working with a cell phone??? wats so special in tat? |
| 12:03.48 | pk1r | this is only happening in meetme confernces |
| 12:05.26 | pk1r | tried the internal timer setting as found googling, but didn't help |
| 12:05.39 | pk1r | anybody here any idea perhaps? |
| 12:06.14 | leifmadsen | pk1r: turn off adaptive jitterbuffering |
| 12:06.18 | leifmadsen | (make it static) |
| 12:06.45 | leifmadsen | adaptive jitterbuffering doesn't really "work" right (at least in 1.4 and 1.6.2 afaik) |
| 12:07.10 | leifmadsen | it may be better in 1.8 and/or 1.10 because I know The_Boy_Wonder (dvossel) was doing some work there when he was working with ConfBridge() |
| 12:07.24 | pk1r | leifmadsen which config file do i find that in? iax.conf? |
| 12:07.27 | leifmadsen | yes |
| 12:07.36 | leifmadsen | look for the jitterbuffer related settings |
| 12:07.54 | leifmadsen | you could just turn off JB for now to test |
| 12:08.02 | leifmadsen | if it's better, try static, if that's OK, then you're gold |
| 12:09.00 | pk1r | ok, going to give it a try. back in a few minutes with results. thank you |
| 12:09.21 | leifmadsen | hope it helps |
| 12:12.20 | vikapi | tzafrir_laptop, oh no.. its not workin now.. im goin mad here. :) |
| 12:19.17 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 12:20.20 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 12:21.34 | *** join/#asterisk felimwhiteley (~quassel@46.7.101.58) |
| 12:21.40 | pk1r | lefmadsen it appears to be helping! got 3 agents in calls right now, no choppy. |
| 12:22.17 | pk1r | leifmadsen should i just change no to static? |
| 12:26.20 | leifmadsen | pk1r: if you *need* jitterbuffer then sure, but if not, then there is no point in introducing an artificial delay :) |
| 12:28.08 | *** join/#asterisk jc319 (~jc318@78-86-169-203.dsl.cnl.uk.net) |
| 12:28.40 | *** join/#asterisk ariel_ (~chatzilla@pdpc/supporter/active/abatista) |
| 12:30.40 | pk1r | ok, i'll keep it NO then :) |
| 12:30.51 | pk1r | thank you for your time |
| 12:32.11 | vikapi | outbound dial using SIP,http://pastebin.com/qpbAdmWg |
| 12:34.02 | vikapi | tzafrir_laptop, let me connect the server with another analog line and try.. |
| 12:34.51 | tzafrir_laptop | app_dial.c:2041 dial_exec_full: Unable to create channel of type 'Zap' (cause 66 - Channel not implemented) |
| 12:35.04 | tzafrir_laptop | Use "DAHDI" instead of "Zap" in your dialplan |
| 12:35.28 | vikapi | ya... |
| 12:35.38 | vikapi | ya...i jus fixed it.. |
| 12:35.41 | vikapi | my bad.. |
| 12:35.43 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 12:35.47 | vikapi | it works.. outbound works.. |
| 12:38.46 | *** join/#asterisk Aven (~user@95.52.229.140) |
| 12:39.05 | *** join/#asterisk wonderworld (~ww@port-92-201-70-142.dynamic.qsc.de) |
| 12:40.32 | *** join/#asterisk fasta (1334259@xs8.xs4all.nl) |
| 12:41.55 | fasta | I want to have completel control over asterisk, which API do I need for that, and where is it documented? |
| 12:42.32 | WIMPy | fasta: vi |
| 12:43.40 | fasta | WIMPy: are you saying that whatever language Asterisk is implemented in (likely C) has no documentation other than some source level comments? |
| 12:43.43 | leifmadsen | fasta: look for AMI (Asterisk Manager Interface) |
| 12:43.56 | leifmadsen | fasta: for source code documentation, yes, see the source |
| 12:44.12 | leifmadsen | russellb wrote a chapter in a book about Asterisk architecture.... wonder if I can find that |
| 12:44.13 | WIMPy | fasta: Are you asking for configuration or realtime control? |
| 12:44.24 | leifmadsen | fasta: your statement is a big vague |
| 12:44.30 | leifmadsen | there are many "control" levels |
| 12:44.45 | fasta | leifmadsen: I said, 'complete control'. |
| 12:44.52 | *** join/#asterisk bchia (~Adium@nat/digium/x-geevrrzribdmzhsl) |
| 12:45.07 | fasta | leifmadsen: so, if there is anything which cannot be done via some 'command' interface, I don't want that level of control. |
| 12:45.35 | WIMPy | Ok, thn you don't want vi for the config files, but for the source code. |
| 12:45.46 | leifmadsen | fasta: look at the source then |
| 12:45.52 | *** join/#asterisk timahvo1 (~rogue@41.223.57.75) |
| 12:46.00 | fasta | Ok, I was under the impression it had been developed further. |
| 12:46.13 | fasta | Thank you for the information. |
| 12:46.19 | leifmadsen | fasta: http://www.venturevoip.com/news.php?rssid=2563 |
| 12:46.37 | leifmadsen | I'm not sure what you mean.... there will never be an interface as fine grained as the source itself |
| 12:47.16 | leifmadsen | if you're looking for a programmatic way of interfacing with asterisk, you want AGI and AMI, or if you want to build control for call flows, you want dialplan, or if you want to build control for users, then you want an auto-attendant |
| 12:47.26 | leifmadsen | there are many levels and interfaces depending on what you're doing and for whom |
| 12:47.52 | leifmadsen | and direct 1-to-1 relationship for all source code controls to external API controls is never going to happen |
| 12:55.37 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 13:01.52 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-daucepwiplssqtrb) |
| 13:02.46 | *** join/#asterisk lost_soul (~noymfb@cpe-67-249-131-153.twcny.res.rr.com) |
| 13:07.09 | *** join/#asterisk irroot (~irroot@pbx.distrotech.co.za) |
| 13:19.46 | *** join/#asterisk tekzilla (~jon@hmbg-5f763de5.pool.mediaWays.net) |
| 13:20.27 | fasta | leifmadsen: Have you ever seen the documentation for e.g. Boost? |
| 13:20.32 | Katty | hi |
| 13:20.37 | fasta | leifmadsen: clearly, people do document some software. |
| 13:21.12 | leifmadsen | fasta: I have done a bit of Asterisk documentation myself |
| 13:21.13 | Katty | i haz a bagel |
| 13:21.19 | leifmadsen | I have not looked at Boost |
| 13:21.38 | leifmadsen | Katty: I be le jealous |
| 13:21.56 | Katty | go get breakfast |
| 13:22.15 | leifmadsen | meh |
| 13:22.18 | leifmadsen | I had cereal |
| 13:22.35 | Katty | was it boring cereals |
| 13:23.31 | anonymouz666 | module unload res_cereals.so then |
| 13:24.14 | leifmadsen | Katty: Cheerios (honey nut) |
| 13:24.59 | irroot | hey there katty |
| 13:25.21 | anonymouz666 | send a fax to irroot |
| 13:26.03 | fasta | leifmadsen: that document is very useful, btw. |
| 13:27.55 | Katty | leifmadsen: yum. |
| 13:27.57 | Katty | irroot: hello. |
| 13:30.11 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 13:30.59 | irroot | me has finally changed the data store from the old asterisk 1.0 family/key/value for user features ... |
| 13:32.20 | *** part/#asterisk ahfeel (~ahfeel@sd-16412.dedibox.fr) |
| 13:37.23 | avb | my problem in SS7 appeared in not SS7 place :) |
| 13:37.49 | avb | seems asterisk or chan_dahdi or codecs are not playing well x64 environment |
| 13:38.30 | *** join/#asterisk TimeRider (~steve@host-92-27-131-175.static.as13285.net) |
| 13:39.04 | avb | once i was making a call via digital card, i was unable to get audio even from playback () in dialplan |
| 13:39.31 | avb | today i should get reinstalled server |
| 13:39.58 | leifmadsen | irroot: did you update it to work with SQLite3? :) |
| 13:40.48 | *** join/#asterisk sbszulu (~dundubala@41.16.237.23) |
| 13:40.58 | irroot | liefmadsen using pgsql / func_odbc its not the internal database |
| 13:41.33 | *** join/#asterisk scalex000 (~chatzilla@186.6.151.91) |
| 13:41.42 | leifmadsen | irroot: ahhhh gotcha |
| 13:42.07 | irroot | would have 10 calls to func_odbc for user settings now with array and new schema only one |
| 13:42.32 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 13:42.34 | irroot | big problem was in me gui had to touch 41 files |
| 13:42.47 | irroot | kept me quite for few daus |
| 13:42.54 | irroot | s/daus/days/ |
| 13:43.34 | atheos | irroot, you've replaced the asterisk db with pgsql? I'm interested in hearing more about that, have you documented it? |
| 13:43.59 | *** join/#asterisk RickB17 (~RickB17@rn1pat.recoverynetworks.com) |
| 13:44.04 | irroot | atheos not a replacement i use func_odbc and postgres to do the same |
| 13:44.24 | irroot | i basically dont use astdb for anything other than the internal bits |
| 13:44.43 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:44.43 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:44.48 | atheos | ok, misunderstood you, Thanks irroot. |
| 13:47.43 | RickB17 | is anyone able to help me by taking a look at my crash issue? https://issues.asterisk.org/jira/browse/ASTERISK-18085 |
| 13:47.44 | CaptainPants | [ASTERISK-18085] [Status: Triage] Program terminated with signal 11, Segmentation fault. (Asterisk Crashes when dialing certain extensions with the same macro) - https://issues.asterisk.org/jira/browse/ASTERISK-18085 |
| 13:55.16 | *** part/#asterisk merlin8282 (~merlin828@AStrasbourg-554-1-246-62.w81-51.abo.wanadoo.fr) |
| 13:56.24 | irroot | rickb17 cant reproduce it here sorry |
| 13:56.25 | *** join/#asterisk lcat (~lcat@187.45.253.242) |
| 13:56.43 | RickB17 | do you have any recommendations? is there any additoinal debugging i can do? |
| 13:56.58 | RickB17 | it happens everytime i dial a couple of my extensions |
| 13:57.39 | irroot | but there is something that seems odd |
| 13:58.20 | irroot | the [:Dial.... |
| 13:58.30 | irroot | the [ bracket here |
| 13:58.51 | irroot | it should not be |
| 13:59.32 | RickB17 | for this one? http://pastebin.com/t4XBmGda |
| 14:00.02 | irroot | yeah |
| 14:00.35 | irroot | the [:...] in the documentation implies what is in the [] is optional the [] should not be included |
| 14:01.00 | RickB17 | do you mind making that change to the pastebin i jsut sent so i can see? |
| 14:01.25 | RickB17 | i'm intending to check the db to see if CFIMNUM =0 or not |
| 14:01.26 | irroot | Execute what is after the ? if the condition matches and after the : if supplied when condition does not match |
| 14:01.52 | *** join/#asterisk seanbright (~sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 14:02.17 | irroot | ExecIf($[${CFIMNUM} != 0]?Dial(${EPHONES}&SIP/${CFIMNUM}@flowroute,32,kt):Dial(${EPHONES},32,kt)) |
| 14:02.38 | irroot | http://pastebin.com/cZ130W0e |
| 14:02.54 | RickB17 | thanks. I will give that a try right now |
| 14:03.13 | RickB17 | i appreciate your help. |
| 14:03.32 | irroot | pleasure was looking at it the other day but got sidetracked |
| 14:04.42 | *** join/#asterisk gravin (~gravin@202.168.70.115) |
| 14:09.52 | *** join/#asterisk ffs (~garland@unaffiliated/ffs) |
| 14:10.45 | RickB17 | irroot: i think you may have resolved my issue. My next question is; why would this cause the daemon to crash? |
| 14:11.24 | *** join/#asterisk byronsmith (~smithers@84.12.253.146) |
| 14:11.38 | Kobaz | you need [] around a logical expression |
| 14:12.22 | RickB17 | a crash seems a bit extreme for a dialplan logic error. |
| 14:12.47 | byronsmith | hi all, just downloaded the latest svn asterisk trunk and upon compiling I am now receiving the following error |
| 14:12.49 | byronsmith | <PROTECTED> |
| 14:12.49 | byronsmith | astdb2sqlite3.c: In function âinit_stmtâ: |
| 14:12.49 | byronsmith | astdb2sqlite3.c:137: warning: implicit declaration of function âsqlite3_prepare_v2â |
| 14:12.49 | byronsmith | astdb2sqlite3.c: In function âdb_openâ: |
| 14:12.49 | byronsmith | astdb2sqlite3.c:174: warning: implicit declaration of function âsqlite3_open_v2â |
| 14:12.50 | byronsmith | astdb2sqlite3.c:174: error: âSQLITE_OPEN_READWRITEâ undeclared (first use in this function) |
| 14:12.50 | byronsmith | astdb2sqlite3.c:174: error: (Each undeclared identifier is reported only once |
| 14:12.50 | RickB17 | but i do understand my issue now. I will have to review my other strings to make sure they are structured properly |
| 14:12.51 | byronsmith | astdb2sqlite3.c:174: error: for each function it appears in.) |
| 14:12.51 | byronsmith | astdb2sqlite3.c:174: error: âSQLITE_OPEN_CREATEâ undeclared (first use in this function) |
| 14:12.52 | byronsmith | astdb2sqlite3.c:174: error: âSQLITE_OPEN_NOMUTEXâ undeclared (first use in this function) |
| 14:12.52 | byronsmith | make[1]: *** [astdb2sqlite3.o] Error 1 |
| 14:12.53 | byronsmith | make: *** [utils] Error 2 |
| 14:12.57 | irroot | rickb17 you believe punishment should fit the crime and capital is bit much |
| 14:13.01 | irroot | ~pb |
| 14:13.02 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 14:13.08 | byronsmith | whoops |
| 14:13.11 | byronsmith | sorry |
| 14:14.00 | RickB17 | thanks again. I'll update the issue and close it out. |
| 14:14.25 | irroot | pleasure |
| 14:14.47 | byronsmith | http://pastebin.com/n7uBMjFL |
| 14:14.49 | byronsmith | :) |
| 14:16.00 | irroot | byronsmith mmm you have good sqllite implementation |
| 14:16.21 | byronsmith | was standard CentOS install |
| 14:17.13 | *** join/#asterisk BMJ (~bjohns@c-24-126-158-110.hsd1.ga.comcast.net) |
| 14:17.14 | *** mode/#asterisk [+o BMJ] by ChanServ |
| 14:17.20 | BMJ | asterisk-scf |
| 14:17.25 | BMJ | whoops |
| 14:17.28 | leifmadsen | :) |
| 14:18.37 | leifmadsen | byronsmith: can you file that as an issue? I want to assign it to twilson |
| 14:19.54 | byronsmith | sure, can you point me in the right direction? |
| 14:20.16 | byronsmith | nevermind found it:) |
| 14:20.53 | leifmadsen | :) |
| 14:21.00 | irroot | byronsmith sqlite3 -version and also you have the "-dev" for sqllite installed if there |
| 14:22.17 | byronsmith | yes devel is installed, was first thing I tried. sqlite3 -version 3.3.6 |
| 14:23.10 | byronsmith | qlite-devel.i386 3.3.6-5 installed |
| 14:23.14 | irroot | indeed |
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| 14:31.34 | *** join/#asterisk jmls (~Julian@host217-36-208-155.in-addr.btopenworld.com) |
| 14:31.41 | jmls | afternoon all |
| 14:31.50 | jmls | 1.8 newbie, longtime 1.4 user . |
| 14:32.02 | *** join/#asterisk d-_-b- (~d-_-b-@2607:f370:9999:dead:5ab0:35ff:fef7:6be3) |
| 14:32.02 | jmls | so, wtf does "app_dial_gosub_virtual_context" have to do with anything ? |
| 14:32.03 | jmls | ;) |
| 14:32.18 | jmls | Executing [s@app_dial_gosub_virtual_context:1] NoOp("SIP/ae641618-1403-088c-e011-c7a403756c5a-00000099", "") in new stack |
| 14:33.10 | jmls | there is a gosub on the dial command ( ,U(foo)) |
| 14:33.18 | psilikon | dynamic realtime is the shiznit |
| 14:33.28 | jmls | it executes foo,s,1 and then displays that message |
| 14:34.56 | irroot | what dialplan you using ?? you on some gui like ~trixbox |
| 14:35.12 | jmls | no, I'm a vi man ;) |
| 14:36.38 | wdoekes2 | jmls: if you don't Return() from foo, you get that |
| 14:36.51 | wdoekes2 | or if you do, I don't remember |
| 14:36.52 | jmls | hmm. |
| 14:36.52 | jmls | exten => s,n,Return() |
| 14:37.13 | jmls | you *have* to return from a gosub, though , don't you ? |
| 14:37.16 | wdoekes2 | either way, you don't need a return, but you may need the return_status_context_thingy variable |
| 14:37.29 | jmls | oh. |
| 14:37.33 | jmls | thinks |
| 14:37.44 | jmls | sure I read you need to Return() from gosub |
| 14:37.56 | jmls | tries to find the evidence to back his claim .... |
| 14:37.56 | wdoekes2 | you don't need to return from this one.. it resumes with the original dial |
| 14:38.20 | jmls | right - so not quite the same as a Gosub() command |
| 14:38.24 | jmls | okies |
| 14:38.57 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 14:39.39 | jmls | yup. removed the Return() strange message is now MIA |
| 14:39.40 | jmls | ;) |
| 14:39.42 | jmls | thanks |
| 14:39.43 | *** join/#asterisk adyn (~adyn@pdpc/supporter/active/adyn) |
| 14:39.51 | irroot | jmls http://pastebin.com/6c3Wwddw |
| 14:41.12 | jmls | ta. Doesn't mention that you *don't* need Return(), though .. |
| 14:41.39 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:41.39 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 14:42.47 | irroot | yeah liefmadsen will add it to the documentation :P |
| 14:42.53 | jmls | heh |
| 14:43.03 | jmls | go leifmadsen 8-) |
| 14:43.49 | wdoekes2 | hehe |
| 14:44.46 | *** join/#asterisk OlafsenM (~mark.olaf@85.114.39.66) |
| 14:44.50 | OlafsenM | hi guys |
| 14:44.53 | OlafsenM | need help |
| 14:45.44 | leifmadsen | irroot: your keyboard must be odd :) |
| 14:45.49 | OlafsenM | my telco wont play early audio (I just hear silence) |
| 14:46.00 | OlafsenM | they swear it's all ok on their side |
| 14:46.08 | irroot | liefmadsen only on even days |
| 14:46.17 | OlafsenM | because they have another customer which is using the same feature |
| 14:46.18 | leifmadsen | irroot: it's leifmadsen :) |
| 14:46.39 | irroot | i miss spell my own name sometimes appologies |
| 14:46.42 | leifmadsen | :) |
| 14:46.54 | leifmadsen | OlafsenM: you're not trying to answer the call ahead of time are you? |
| 14:47.03 | OlafsenM | none |
| 14:47.09 | OlafsenM | *nope |
| 14:47.22 | OlafsenM | i'm just doing: Progress() |
| 14:47.30 | OlafsenM | Playback(test,noanswer) |
| 14:49.12 | leifmadsen | odd |
| 14:49.18 | leifmadsen | I have nothing else to contribute :) |
| 14:49.37 | byronsmith | leifmadsen https://issues.asterisk.org/jira/browse/ASTERISK-18097 - hope I did it right! |
| 14:49.38 | CaptainPants | [ASTERISK-18097] [Status: Triage] svn revision 326678 of the Asterisk Trunk fails to compile on SQLITE3 error - https://issues.asterisk.org/jira/browse/ASTERISK-18097 |
| 14:54.12 | leifmadsen | byronsmith: so far so good! :) |
| 14:54.58 | *** part/#asterisk avb (~avb@190.122.106.167) |
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| 15:01.10 | psilikon | Is there a character limit on a meetme room name? |
| 15:02.22 | psilikon | I tried generating a random 80 char alphanum as a meetme room but when I dial into it I get disconnected immediately yet the cli reports no errors. |
| 15:05.52 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 15:07.01 | *** join/#asterisk ShadwDrgn (arcane@75-131-200-102.dhcp.gwnt.ga.charter.com) |
| 15:07.34 | ShadwDrgn | I am getting an error with the init.d script for starting asterisk |
| 15:07.36 | ShadwDrgn | ulimit: open files: cannot modify limit: Operation not permitted |
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| 15:26.55 | mickecarlsson | /join #asterisk-testing |
| 15:27.15 | jmls | mickecarlsson: error ;) |
| 15:27.29 | mickecarlsson | Jupp, one space to many |
| 15:27.47 | jmls | :-D |
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| 15:40.12 | wonko | ok, doing the Solaris -> OI "upgrade" |
| 15:40.15 | wonko | let's see how this goes |
| 15:40.49 | Qwell | OI? |
| 15:43.47 | wonko | oops |
| 15:43.49 | wonko | wrong channel |
| 15:43.53 | wonko | sorry for the noise |
| 15:43.57 | wonko | (OpenIndiana, btw) |
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| 16:06.39 | scalex000 | hello, I have a connection using h323, the i can't hear the otherside and the otherside either |
| 16:11.02 | Guizmo | is it possible to connect directly an ISDN line BRI in one of the port of a B410P ? |
| 16:12.01 | scalex000 | any idea, connection its good |
| 16:12.43 | scalex000 | any idea, not errors appear, but I can't hear the otherside |
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| 16:22.14 | *** join/#asterisk dasgoodyah (~sha@bas1-montreal19-1177814296.dsl.bell.ca) |
| 16:23.03 | dasgoodyah | Looking for some help with my asterisk setup. I have no audio being transmitted between internal extensions and external calls |
| 16:25.05 | dasgoodyah | willing to pay paypal for some help... |
| 16:25.15 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 16:25.32 | paulc | You're probably better off paying someone in here, than PayPal - they have enough money already ;-) |
| 16:25.46 | paulc | Is your Asterisk box on a public IP or behind a firewall/router? |
| 16:25.46 | dasgoodyah | lol |
| 16:26.06 | dasgoodyah | i have a hosted asterisk box from rentpbx.com |
| 16:26.14 | dasgoodyah | running elastix |
| 16:26.32 | paulc | dasgoodyah: I'll be right back - people at my desk for a sec |
| 16:28.50 | leifmadsen | I suggest getting help from support at rentpbx.com :) |
| 16:29.57 | dasgoodyah | if i can't get this fixed myself i'll do that for sure |
| 16:30.16 | dasgoodyah | but i'm a little stubborn with technical things, always wanting to fix it myself |
| 16:30.59 | dasgoodyah | could this be a nat issue? Even thought the pbx is not on my local network |
| 16:31.10 | leifmadsen | I suggest starting with 'sip set debug' and find out if things are setup to go where they should, along with 'rtp set debug' |
| 16:31.24 | leifmadsen | if the PBX is not on your local network it's most certainly (and probable) that it is a NAT issue |
| 16:32.14 | *** join/#asterisk irroot (~irroot@41.49.126.35) |
| 16:32.57 | dasgoodyah | I've done a sip debug but i didnt get much info, just that the call has been connected and nothing else after until i hung up the call |
| 16:33.46 | dasgoodyah | How would port forwarding go, My local ip > the asterisk box? |
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| 16:48.57 | scalex000 | leifmadsen, what can I do when I can hear both side using h323 |
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| 17:07.33 | ssureshot | should the firmware load on an astribank if it doesn't see the T1 ? |
| 17:09.43 | jaytee | Is it possible for me to have two sip accounts that share the same voicemail mailbox? |
| 17:10.02 | irroot | jaytee yes indeed |
| 17:10.23 | jaytee | so just use the same mailbox=XXX@default in each sip account? |
| 17:10.23 | irroot | mailbox= in sip.conf |
| 17:10.33 | irroot | snap |
| 17:10.58 | jaytee | irroot, thanks. Thought that would work but just wanted confirmation. |
| 17:11.19 | p3nguin | It is also possible for two sip accounts to share two voice mail boxes. |
| 17:11.41 | p3nguin | (which is how I would prefer to do it) |
| 17:13.06 | irroot | jaytee the mailbox= function is purely for the notification on the phone ie light |
| 17:13.33 | irroot | and has been broken recently |
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| 17:55.05 | jimbo_uk | guys, anyone know of a way to rewrite the CDR dst fieldwithout just looping the call back to the local channel and supressing theoriginal call with nocdr() |
| 18:00.04 | jimbo_uk | shit easy. CDR(dst)=blah |
| 18:03.03 | jimbo_uk | hmm not so easy |
| 18:03.05 | jimbo_uk | <PROTECTED> |
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| 18:36.50 | irroot | jimbo_uk jambo ... cant mess the dst too easy but not to hard either :P |
| 18:39.12 | leifmadsen | jimbo_uk: the real answer is "use CEL" |
| 18:39.31 | leifmadsen | otherwise, ya, you need to make sure the dialplan is executing what you want written |
| 18:40.02 | irroot | you can put a "Goto" where you dialing to go to the dst you want and then dial from there |
| 18:40.13 | irroot | in the same context even |
| 18:47.21 | jmls | mmm. has anyone implemented desktop sharing with a sip client ? |
| 18:47.23 | *** join/#asterisk gruvfunk (~chatzilla@cpe-68-172-221-157.hvc.res.rr.com) |
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| 18:53.19 | gruvfunk | greets |
| 18:53.47 | gruvfunk | since there are way more people in here than in #switchvox -- is Asterisk code at the core of Switchvox? can I edit files manually the same way? |
| 18:54.06 | ShadwDrgn | does anyone know of a really reliable SIP trunking service that offers unlimited channels, inbound AND outbound calling, and united states phone numbers. Toll free numbers would be a bonus. |
| 18:54.47 | gruvfunk | Voip.ms |
| 18:55.03 | gruvfunk | ShadwDrgn: ^^ |
| 18:55.08 | ShadwDrgn | looking now |
| 18:55.29 | gruvfunk | they offer unlimited channels on their toll free did |
| 18:55.47 | ShadwDrgn | outgoing and incoming calls? |
| 18:57.18 | ectospasm | gruvfunk: it's Asterisk underneath, but good luck getting any manual changes to work. As such, it's totally unsupported and you will void your warranty for trying. |
| 18:57.38 | gruvfunk | ectospasm: interesting... wow.. thank you |
| 18:57.43 | gruvfunk | basically, no go, right? |
| 18:58.13 | ectospasm | yes. there are a lot of things compiled into binary form, it wouldn't be a simple matter to add functionality you seek. |
| 18:58.27 | gruvfunk | ShadwDrgn: yes, no restriction on channels, merely pay for the usage (per minute) |
| 19:00.07 | gruvfunk | (on toll free DID) |
| 19:00.32 | gruvfunk | regular US DID comes with 25 channels, but you can purchase more to add |
| 19:02.40 | *** join/#asterisk fireman_biff (~biff@65.48.133.102) |
| 19:03.13 | *** join/#asterisk qop (~H@200.94.69.114) |
| 19:04.41 | qop | hello. is there some way to know which extensions.conf is being used? I just need to add more extensions, had 3 and need 10. Added them over Elastix but I get 403 bad auth when login with a sip phone |
| 19:05.16 | qop | oh and... opened the extensions.conf and can not see anything related to the extensions we already have |
| 19:05.55 | navaismo | @qop extensions.conf have the dialplan for devices look into sip.conf and iax.conf |
| 19:06.12 | qop | navaismo: ty :) |
| 19:06.22 | fireman_biff | Hi, any idea why I would get "Everyone is busy/congested at this time" for an IAX2 trunk with no calls and no limit on max channels, which shows up as OK in the output of asterisk -rx 'iax2 show peers' ? (asterisk 1.4.22) |
| 19:06.22 | navaismo | @qop if you are using web based administartion like trixbox or elastix you need to edit via web gui |
| 19:06.31 | fireman_biff | I'm just trying to dial an extension on the other pbx |
| 19:07.11 | navaismo | @qop do not edit conf files only those have *_custom.conf otherwise you can broke your system |
| 19:07.59 | qop | navaismo: uhm, well, I though that adding the extensions through it would suffice.. maybe I am assuming wrongly? But I am getting 403 and assume that I have to edit something else too |
| 19:08.05 | qop | ok |
| 19:08.37 | navaismo | @fireman_biff maybe you can search in the full log on both servers for more detailed error. |
| 19:09.24 | *** part/#asterisk esaym153 (~esaym153@cpe-67-11-147-73.satx.res.rr.com) |
| 19:10.13 | navaismo | @qop the 403 indicates that authorizacion name or password doesnt match |
| 19:10.16 | qop | navaismo: nvm, I forget to think before asking some times :p |
| 19:10.20 | navaismo | @qop with your peer |
| 19:10.29 | fireman_biff | hmm... why didnt i think of that... thanks navaismo |
| 19:10.37 | *** join/#asterisk albertoandrade (~albertoan@187.112.123.80) |
| 19:10.45 | qop | navaismo: yes.. I just opened the sip.conf custom and indeed can not see the new users |
| 19:10.56 | brainiac | Does anyone know what command(s) will show the voltage level on the ports on a TDM2400P? |
| 19:11.05 | *** join/#asterisk vinhdizzo (~vinh@dhcp-053179.ics.uci.edu) |
| 19:11.05 | qop | therefore no auth is possible for an unexistent user |
| 19:12.22 | navaismo | @qop the file for sip and iax peers in a fbpx based system its sip_additional and iax_additional |
| 19:13.15 | ssureshot | oh man,, I think dahdi source needs to have fxload as a package required.. lol I just fought that crap for 3 hours trying to figure out why my firmware wouldn't load for an astribank |
| 19:14.07 | navaismo | @qop maybe with allowguest=yes but is insecure |
| 19:15.34 | qop | navaismo: into sip_additional.conf ? |
| 19:15.52 | qop | navaismo: security is not an issue here |
| 19:16.12 | navaismo | @qop no you need to download the module SIP settings from webGUI |
| 19:16.31 | fireman_biff | I was able to get it to work by setting "requirecalltoken=no" on the receiving end, is there any security risk with that? |
| 19:17.03 | qop | navaismo: ah ok, I remember that |
| 19:21.20 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-146.cablep.bezeqint.net) |
| 19:21.51 | *** part/#asterisk irroot (~irroot@197.171.220.165) |
| 19:29.08 | *** join/#asterisk jc319 (~kvirc@78-86-169-203.dsl.cnl.uk.net) |
| 19:35.47 | *** join/#asterisk wcselby (~wcselby@pdpc/supporter/professional/wcselby) |
| 19:35.50 | wcselby | o/ |
| 19:36.23 | *** join/#asterisk vikapi (~vikapi@124.125.34.134) |
| 19:44.18 | *** join/#asterisk Aven (~user@95.52.229.140) |
| 19:47.28 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-145.cablep.bezeqint.net) |
| 19:48.55 | *** join/#asterisk Defraz (~Defraz@184-155-137-27.cpe.cableone.net) |
| 19:51.12 | mateu | is there an easy way to write a different CDR dst field than the extension name? |
| 19:53.40 | wcselby | mateu- userfield? |
| 19:54.06 | mateu | I was going to go that route, but it's already taken and it seems I don't have custom field support available. |
| 19:54.30 | wcselby | are you using 1.8.x? if so you could look at CEL logging.... |
| 19:54.44 | mateu | 1.6.x unfortunately |
| 19:55.55 | navaismo | 1.6 support accountcode amaflags and userfiled to "writeable" |
| 19:57.04 | mateu | hmm, maybe I can use amaflags (have to see if replace DOCUMENTATION with something else presents a problem) |
| 20:03.10 | jimbo_uk | hey |
| 20:03.14 | jimbo_uk | anyone good with AGI? |
| 20:04.42 | wcselby | jimbo_uk- depends on what you need? |
| 20:04.58 | jimbo_uk | i'm passing |
| 20:05.18 | jimbo_uk | EXEC SET CDR(userfield)=2011-07-07 18:11:11 |
| 20:05.26 | jimbo_uk | of course the space is cocking it up |
| 20:05.32 | jimbo_uk | is is as simple as doing a |
| 20:05.36 | jimbo_uk | EXEC SET CDR(userfield)="2011-07-07 18:11:11" |
| 20:06.08 | jimbo_uk | because if so i need to figure out how to get a " into a string in c without it thinking it's a terminator |
| 20:06.14 | jimbo_uk | (not the arnie variety) |
| 20:06.32 | wcselby | hmmm, are you using an AGI library ? |
| 20:06.38 | jimbo_uk | yes the C one |
| 20:06.45 | wcselby | i haven't worked with that |
| 20:07.07 | jimbo_uk | called cagi |
| 20:07.19 | *** join/#asterisk scalex000 (~chatzilla@186.6.70.68) |
| 20:07.26 | jimbo_uk | -- AGI Script Executing Application: (SET) Options: (CDR(userfield)=2011-07-0720:42:54) |
| 20:07.33 | scalex000 | hello |
| 20:07.36 | jimbo_uk | this is what is being set |
| 20:07.38 | scalex000 | I need help |
| 20:07.43 | wcselby | i would try the quote route, if that doesn't work, try setting it in a variable first then passing the variable, without spaces |
| 20:07.59 | scalex000 | Im using h323 but i can't hear the other side |
| 20:08.13 | jimbo_uk | if i remove the space it's ok |
| 20:08.19 | wcselby | scalex000- can the other side hear you? |
| 20:08.21 | jimbo_uk | i think the AGI is seeing is as two options |
| 20:08.38 | *** join/#asterisk timahvo1 (~rogue@41.223.57.72) |
| 20:08.50 | scalex000 | wcslby, nop |
| 20:09.08 | scalex000 | wcslby, from where i need to check |
| 20:09.56 | wcselby | scalex000 - give us more details? how are you making the h323 call? are you using a SIP phone? is NAT involved anywhere in the process? what version of h323 are you using, which version of asterisk, etc |
| 20:10.42 | scalex000 | ok, im using a sip phone, without nat |
| 20:10.45 | scalex000 | let me see |
| 20:11.13 | scalex000 | h323 1.18.0 |
| 20:11.43 | scalex000 | Asterisk is 1.6.2.18 |
| 20:11.51 | mateu | jimbo_uk: exten => s,n,Set(CDR(userfield)=2011-07-07 18:11:11) works for me (but that's from the dialplan) |
| 20:13.25 | wcselby | scalex000 - maybe give us the CLI output, and if there's any h323 debug (I don't know, I don't use it) |
| 20:15.23 | scalex000 | http://pastebin.com/hBGTTvTy |
| 20:15.32 | scalex000 | wcslby, http://pastebin.com/hBGTTvTy |
| 20:15.43 | scalex000 | wcslby, look ok |
| 20:19.26 | wcselby | scalex000 - does the other end ever answer, and does asterisk show this? |
| 20:19.37 | scalex000 | yes |
| 20:19.57 | wcselby | well, please show a complete call in the cli log then, not just half of one |
| 20:20.02 | scalex000 | wcslby, yes, ring in the another side but can't hear me |
| 20:30.27 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 20:30.27 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 20:30.31 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 20:31.51 | jimbo_uk | fyi "ko ko" works |
| 20:32.17 | jimbo_uk | yes dialplan is fine. agi didn't like the space - had to " " it |
| 20:32.24 | jimbo_uk | night night all |
| 20:34.58 | scalex000 | wcslby, ok |
| 20:36.30 | *** join/#asterisk vikapi (~vikapi@124.125.34.134) |
| 20:37.01 | vikapi | command, core show channels should atleast give a pseudo channel info correct?? |
| 20:37.02 | scalex000 | wcslby, do you need debug |
| 20:37.06 | scalex000 | wcslby, do you need debug? |
| 20:37.17 | vikapi | when does this command doesn't give any output.. im facing tat now.. |
| 20:38.03 | scalex000 | wcslby, http://pastebin.com/pcDWKp0Q |
| 20:38.22 | *** join/#asterisk ahfeel (~ahfeel@sd-16412.dedibox.fr) |
| 20:38.27 | ahfeel | Hi all |
| 20:38.41 | vikapi | dahdi_cfg -vvvv shows "1 channels to configure" |
| 20:39.11 | qop | omg omg omg... I tried to add some extensions with elastix and then using the sip_additional.conf now I do not have the option to select extension in elastix :( |
| 20:39.29 | qop | reverted sip_additional.conf to original and restarted but nothing :( |
| 20:39.42 | scalex000 | wcslby, http://pastebin.com/0FY6TSW5 here debug |
| 20:42.04 | qop | <PROTECTED> |
| 20:42.33 | wcselby | qop - check #elastix or the elastix support channel (I don't know what this is off the top of my head) |
| 20:43.43 | wcselby | scalex000 - what h323 conf file are you using? |
| 20:44.18 | qop | wcselby: http://picpaste.com/222222-FNNYnBwI.png |
| 20:44.29 | qop | wcselby: it is sip extension |
| 20:45.16 | scalex000 | wcslby, http://pastebin.com/a0vVdzcw |
| 20:45.47 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 20:47.45 | ahfeel | I'm trying to detect through the Manager API Events when an originated call has answered and when is hasn't, but I can't find any hint in the events |
| 20:47.53 | ahfeel | anybody has an hint for me ? |
| 20:48.09 | wcselby | qop - I'm not sure what your picture was supposed to be showing me, but as I said, you need to follow up with normal elastix support channels, not asterisk support. |
| 20:48.18 | qop | if any ideas I'd really appreciate it |
| 20:48.53 | wcselby | scalex000 - do you have any g729 licenses? also, have you tried binding to your specific ip address instead of 0.0.0.0 and then reload? only thing I can think of off the top of my head. |
| 20:49.03 | wcselby | scalex000 - what type of nortel are you trying to connect to? |
| 20:49.04 | scalex000 | wcslby, yes |
| 20:49.11 | scalex000 | wcslby, BCM 200 |
| 20:49.15 | qop | wcselby: yes... 17 users in #elastix :s I was trying to show the little thing marked as "extension" |
| 20:49.19 | qop | ty |
| 20:49.42 | wcselby | qop - I'm not trying to be an ass, I just have no idea how to help since I don't use elastix |
| 20:49.56 | qop | wcselby: yes I know :0 |
| 20:51.01 | scalex000 | wcslby, thank you very much, I love you |
| 20:51.07 | wcselby | scalex000 - according to voip-info.org, you need to "set the system type to "Other" when chosing the H.323 trunking protocol" - I'm assuming this is on the nortel side of the trunk |
| 20:51.13 | wcselby | scalex000 - is it working? |
| 20:51.23 | scalex000 | wcslby, bindingaddress |
| 20:51.29 | wcselby | scalex000 - cool |
| 20:51.46 | wcselby | i thought it may have something to do with that, you were sending 127.0.0.1 in some of the debug packets |
| 20:52.40 | navaismo | @qop As I sayed before in freepbx systems dont edit conf files everything needs to be edited from GUI |
| 20:52.50 | paulc | ~book |
| 20:52.51 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342. Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/, or ~buybook |
| 20:53.43 | navaismo | @qop try replacing the sip_additional with the original and perform amportal stop then amportal kill and finally amportal start and see if your extensions go back |
| 20:53.46 | qop | navaismo: what happens if you edit manually? I saw a legend about the file being override |
| 20:54.04 | qop | navaismo: but looks like elastix is somehow unable to read it |
| 20:54.28 | scalex000 | wcslby, do yo know another option or how to use tos_audio, cos_audio |
| 20:54.43 | navaismo | if you edit manually it override the gui configuration and viceversa, if you need to add a device via conf file you must use the _custom.conf |
| 20:55.20 | *** part/#asterisk ahfeel (~ahfeel@sd-16412.dedibox.fr) |
| 20:56.46 | qop | navaismo: I used sip_additional.conf |
| 20:57.14 | wcselby | scalex000 - no I haven't played with those, sorry |
| 20:58.06 | navaismo | @qop in the edited sip_additional.conf you have all your devices (old and new devices?) |
| 20:58.32 | navaismo | @qop in the asterisk cli do a sip relaod an try to register one |
| 20:59.09 | qop | navaismo: all I saw there was extensions, so I added new ones... I have already reverted the file to its original state tho |
| 21:01.01 | wcselby | qop - what are you using to edit the files? |
| 21:01.35 | qop | I can not even access /index.php?menu=pbxconfig :s |
| 21:01.41 | qop | wcselby: emacs |
| 21:03.05 | qop | odd... now every time that I click a new tab I am requested for my login... |
| 21:03.53 | qop | navaismo: restarting asterisk would do that? |
| 21:03.55 | navaismo | @qop do you edited other file? |
| 21:03.59 | wcselby | qop - do you have a backup of the file you edited before you edited it? |
| 21:04.00 | qop | or asterisk.reload |
| 21:04.05 | navaismo | nop |
| 21:04.12 | qop | navaismo: not a single thing else |
| 21:04.14 | navaismo | amportal restart better |
| 21:05.57 | navaismo | I edited the sip_additional adding other extension when reload via GUI the FPBX delete the new extension, so If you restore the original file and execute amportal stop and then amportal start |
| 21:06.42 | qop | STOPPING ASTERISK Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
| 21:06.57 | qop | file exists but is blank |
| 21:07.17 | navaismo | yes its a "normal" message when using fpbx based systems |
| 21:08.16 | qop | ok... so, restoring old file and restart should do... but it doesnt... |
| 21:08.41 | Qwell | Maybe you aught to try #freepbx? |
| 21:09.52 | qop | ok |
| 21:09.53 | navaismo | in the GUI disappear all extensions? |
| 21:12.02 | qop | navaismo: checking... |
| 21:12.46 | qop | navaismo: I can not know, since I need to be able to access the PBX tab where those are |
| 21:13.06 | navaismo | you can't acces? |
| 21:13.41 | *** join/#asterisk [Outcast] (~anonymous@pool-108-7-70-195.bstnma.fios.verizon.net) |
| 21:14.32 | qop | navaismo: no... but hold on the db might be the problem... |
| 21:14.42 | navaismo | what are the output for /var/log/httpd/acces_log and /var/log/httpd/error_log |
| 21:14.44 | navaismo | ok |
| 21:18.42 | scalex000 | wcslby, using g729 the sound is not good with h323 |
| 21:19.53 | wcselby | doesn't surprise me |
| 21:19.59 | wcselby | g729 is a compressed codec |
| 21:20.06 | wcselby | does it sound better with g711? |
| 21:20.12 | qop | navaismo: I just see the "httpd: Could not reliably determine the server's fully qualified domain name, using 127.0.0.1 for ServerName" |
| 21:20.34 | qop | navaismo: I get DB Error: connect failed when 72.16.32.21/admin |
| 21:20.41 | *** join/#asterisk CaptainPants (~CaptainPa@nat/digium/x-zglkdyvdcsbufscd) |
| 21:20.48 | Qwell | qop: #freepbx, please. |
| 21:21.00 | qop | Qwell: yes |
| 21:23.47 | scalex000 | wcslby, YES |
| 21:26.12 | vikapi | tryin to configure x100p..when giving, "core show channels" in CLI, it shows 0 channels.. analog line is connected to x100p..any idea on this? |
| 21:26.41 | wcselby | vikapi - dahdi show status |
| 21:26.53 | wcselby | core show channels shows any channels that are currently in use |
| 21:27.16 | vikapi | *CLI> dahdi show status |
| 21:27.17 | vikapi | Description Alarms IRQ bpviol CRC4 Fra Codi Options LBO |
| 21:27.17 | vikapi | Wildcard X100P Board 1 OK 0 0 0 CAS Unk 0 db (CSU)/0-133 feet (DSX-1) |
| 21:27.28 | vikapi | wcselby, but it should atleast show pseudo channels rite? |
| 21:28.03 | wcselby | vikapi - only if it's currently in use |
| 21:28.34 | navaismo | @vikapi chan_dahdi is configured? |
| 21:28.41 | wcselby | were you looking for different results? |
| 21:29.23 | vikapi | wcselby, today when i configurd in the same server with 1.8.4, it showd me pseudo channels.. i had to re-install the server but when configuring now, it doesn't show me tat |
| 21:30.01 | wcselby | were you using the channels at the time? |
| 21:30.03 | vikapi | navaismo, http://pastebin.com/GxZkufsc |
| 21:30.13 | vikapi | wcselby, no.. |
| 21:30.13 | wcselby | why are you looking for pseudo channels, is probably the better question |
| 21:31.01 | vikapi | wcselby, good question..actually,it was weird, when i saw it first and dissappearin in the new installation..puzzled.. |
| 21:31.29 | wcselby | vikapi - so you're having no other issues than you don't see the same output now? |
| 21:33.12 | vikapi | wcselby, 1 thng i doubt..when i dial into the x100p, and in extension.conf,i have said to dial in to a sip extension..but after call is received, and when hung up in ekiga side, my mobile(caller) doesn't get hung up.. |
| 21:33.28 | vikapi | wcselby, bcoz of misconfiguration?? |
| 21:33.40 | wcselby | not necessarily |
| 21:34.02 | wcselby | i don't recall all the reasons for that particular error right now though |
| 21:34.15 | wcselby | i know it's been talked about on the mailing list several times though |
| 21:34.19 | wcselby | search through that |
| 21:34.36 | Qwell | It's a terrible card. Failing hangup detection is normal. |
| 21:35.30 | vikapi | wcselby, ok.. |
| 21:36.29 | navaismo | set busydetect to yes and try |
| 21:36.40 | Qwell | umm, no |
| 21:36.54 | Qwell | busydetect is not related to hanging up at all... |
| 21:37.16 | vikapi | wcselby, jus check this paste..http://pastebin.com/nqCh1JYM its the CLI output when on call and hangup.. |
| 21:37.27 | vikapi | jus see 2 hangup occurances.. looks weird.. |
| 21:37.52 | Qwell | That's because you're doing a Hangup in the h exten for some silly reason. |
| 21:40.12 | wcselby | Hangup() doesn't need to be called in the 'h' exten, you put it at the end of the calling exten |
| 21:41.26 | vikapi | wcselby, & Qwell, rite.. i screwed up the setup.. |
| 21:41.35 | vikapi | let me try 1ce again |
| 21:41.47 | vikapi | navaismo, tried and working.. |
| 21:41.48 | navaismo | @Qwell interesting, so why when we have troubles with hanging up and activate the busydetect it works, on the side of server of course?? |
| 21:42.04 | Qwell | navaismo: Placebo effect. |
| 21:42.22 | navaismo | ¬¬ |
| 21:44.14 | navaismo | @Qwell [quote]busydetect: If enabled, Asterisk will analyze the audio coming in on the line during a call or a dial attempt to attempt to recognize busy signals. This is useful on analog trunk interfaces both to detect a busy signal when dialing out, and for detecting when the person has hung up. See also Disconnect Supervision. Be sure that you don't use this on digital interfaces like QuadBri cards and so on. Otherwise you will run in "broken calls" |
| 21:44.14 | navaismo | <PROTECTED> |
| 21:44.38 | Qwell | citation? |
| 21:44.55 | navaismo | yep |
| 21:45.02 | navaismo | from http://www.voip-info.org/wiki/view/Asterisk+config+zapata.conf |
| 21:45.10 | Qwell | voip-info wiki is garbage |
| 21:45.24 | navaismo | but its usefull |
| 21:45.27 | Qwell | It has extremely incorrect information. Case in point - the stuff you just linked. |
| 21:47.56 | navaismo | ok now from chan_dahdi: ; On trunk interfaces (FXS) and E&M interfaces (E&M, Wink, Feature Group D |
| 21:47.56 | navaismo | ; etc, it can be useful to perform busy detection either in an effort to |
| 21:47.56 | navaismo | ; detect hangup or for detecting busies. This enables listening for |
| 21:47.56 | navaismo | ; the beep-beep busy pattern. |
| 21:47.59 | navaismo | ; |
| 21:48.01 | navaismo | ;busydetect=yes |
| 21:48.03 | navaismo | ; |
| 21:48.05 | navaismo | ; If busydetect is enabled, it is also possible to specify how many busy tones |
| 21:48.07 | navaismo | ; to wait for before hanging up. |
| 21:48.26 | wcselby | navaismo - you're not helping your case |
| 21:48.28 | navaismo | so can you explain me the real function for busydetect for a better practice |
| 21:48.35 | Qwell | It detects busy signals. |
| 21:49.00 | wcselby | Qwell - why would you think that? ;-) |
| 21:50.02 | navaismo | wcselby-- ooh a smartty comment |
| 21:50.13 | wcselby | sorry |
| 21:50.19 | wcselby | i was just being silly |
| 21:50.45 | *** join/#asterisk TimeRider (steve@188-220-33-123.zone11.bethere.co.uk) |
| 21:51.31 | navaismo | obviously it detect busy pattern but what is the purpose? |
| 21:51.47 | Qwell | So that you can hang up with you get a busy signal. |
| 21:52.05 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 21:52.29 | navaismo | me or asterisk? |
| 21:52.51 | Qwell | I don't know. Is that config in chan_dahdi.conf, or navaismo.conf? |
| 21:53.28 | navaismo | mmmm |
| 21:54.29 | *** join/#asterisk Eitan (~Eitan@adsl-99-22-192-148.dsl.lsan03.sbcglobal.net) |
| 21:54.52 | Eitan | i think it might have been in here, someone told me they had created a TTS script for an IVR that pretty much pulled the voice from the TTS for OSX... |
| 21:55.09 | navaismo | so mr sarcasm, you cannot answer with the right answer, I made a question cause you wrote my interpretation is bad, and you cannot answer me without sarcasm |
| 21:55.37 | navaismo | if I ask is because i think you can point me in the right direction |
| 21:56.46 | scalex000 | hello, im using h323, and dmtfmode=rfc2833 but i can dial any number or extension |
| 21:56.48 | wcselby | asterisk will detect the busy signal on the dahdi channel and hangup the call, if you've got busydetect set to yes. |
| 21:57.07 | navaismo | thx wcselby |
| 21:57.20 | wcselby | one of the unintended side effects of this is that a lot of telco's will put a busy signal on the line after a call has been hungup if the line is still being held open for whatever reason |
| 21:57.42 | wcselby | thus, busydetect sometimes will hangup calls that asterisk otherwise doesn't detect, due to shoddy hardware or whatever |
| 21:57.55 | wcselby | now, it's tuime to go eat, so i'll be back later |
| 21:57.58 | wcselby | time* |
| 22:00.51 | qop | navaismo: :D |
| 22:01.13 | qop | navaismo: it was mysql db that was down doe to no space left |
| 22:01.41 | *** join/#asterisk dasgoodyah (~sha@bas1-montreal19-1177816043.dsl.bell.ca) |
| 22:02.16 | navaismo | jeje |
| 22:03.23 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-daucepwiplssqtrb) |
| 22:03.38 | qop | yeah :s |
| 22:04.44 | vikapi | from where can i get asterisk irc logs? |
| 22:05.06 | navaismo | @qop and now everything ins working again? |
| 22:05.25 | *** join/#asterisk _jdccdevel (~chatzilla@S0106000a95e61848.ed.shawcable.net) |
| 22:05.53 | qop | navaismo: yum remove vlc wine ElectricFence.... why on earth did the other admi installed so much crap there..?? |
| 22:06.00 | qop | navaismo: yes all workin :) |
| 22:06.44 | qop | tnx for the help |
| 22:07.17 | navaismo | @qop anytime |
| 22:09.43 | *** part/#asterisk navaismo (~navaismo@187.170.7.174) |
| 22:19.20 | *** join/#asterisk ahfeel (~ahfeel@sd-16412.dedibox.fr) |
| 22:23.18 | *** join/#asterisk fhmiv (~fhmiv@adsl-065-005-242-247.sip.ags.bellsouth.net) |
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| 22:34.02 | *** join/#asterisk Farrukh (softmaster@119.155.42.234) |
| 22:34.14 | Farrukh | hi |
| 22:34.43 | Farrukh | anybody who can help with aserisk in UK |
| 22:34.50 | *** join/#asterisk The_REV (~The_REV@c-76-21-1-80.hsd1.ca.comcast.net) |
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| 22:35.44 | Farrukh | please someone help me |
| 22:36.17 | p3nguin | ~ask |
| 22:36.17 | 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. |
| 22:37.27 | *** part/#asterisk fireman_biff (~biff@65.48.133.102) |
| 22:37.31 | Farrukh | i am stuck with disconnect tone on my analog card I receive continus tone from BT but dont know where to configure |
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| 22:39.22 | Farrukh | anybody ..... |
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| 22:49.14 | Farrukh | can anybody help pl |
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| 22:54.16 | Farrukh | ok i have solved myself , thanks for your support |
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| 23:11.57 | byronsmith | hi guys, recently did an upgrade from 1.4.39 to 1.6.2.19 and upon issuing a reload randomly the asterisk process crashes out with /usr/sbin/safe_asterisk: line 118: 4602 Segmentation fault (core dumped) nice -n $PRIORITY ${ASTSBINDIR}/asterisk -f ${CLIARGS} ${ASTARGS} >&/dev/${TTY} < /dev/${TTY} |
| 23:12.23 | byronsmith | doesnt happen all the time either |
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| 23:20.20 | byronsmith | guess I'm on my own with this one hey? |
| 23:28.13 | *** join/#asterisk ferdna (~yup@cpe-67-10-220-35.elp.res.rr.com) |
| 23:28.25 | p3nguin | Stop running "reload" all willy-nilly. |
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| 23:35.15 | byronsmith | how else would you suggest I reload my system then? |
| 23:35.52 | p3nguin | What does "reload my system" actually mean? |
| 23:36.52 | p3nguin | In the years I have been running Asterisk, I have never once needed to run "reload" on any of them. |
| 23:36.55 | byronsmith | I've recompiled to a lower version of the 1.6.2 tree, from 1.6.2.19 to Asterisk 1.6.2.18.1 and my problem is not more. I can issue as many reloads as I want and it's stable |
| 23:37.41 | byronsmith | i realise there are more granluilar ways to reload however I've never had an issue issuing a complte reload until .19 |
| 23:38.16 | byronsmith | anyways, i seem to have resolved this now thankfully! but many thanks for your help. |
| 23:38.54 | p3nguin | File a bug report if you think it's a problem in the current version that needs addressed. |
| 23:39.43 | byronsmith | i shall do so but it's 00:39 and thee old eye lids are giving way. btw you say not once in all your years have you run reload..? so how do you do it then? |
| 23:39.56 | byronsmith | how do you apply config changes with a reload of some sort? |
| 23:40.22 | byronsmith | am i missing a trick!? |
| 23:40.28 | p3nguin | Depends on what I had to reconfigure. If I changed sip.conf, I run sip reload. If I changed extensions.conf, I run dialplan reload. |
| 23:40.45 | byronsmith | ok sure but they are still reloads of some description |
| 23:41.46 | byronsmith | i shall try be more granular in my reloading habits in future, advice taken on board! many thanks |
| 23:43.01 | p3nguin | You can also reload certain individual modules with module reload <module name>. |
| 23:44.06 | byronsmith | yes, very useful indeed, only found out about that one a few weeks ago |
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| 23:47.12 | byronsmith | p3nguin you heard of Thirdlane? would you happen to know how he issues a 'reload' after pressing the 'reload' link on his web gui? |
| 23:47.52 | p3nguin | I wouldn't have any idea; I don't do GUIs. |
| 23:48.41 | p3nguin | Depending on what the button does, it very well may run 'reload' in Asterisk. |
| 23:50.06 | byronsmith | still I think there is an issue in .19 because i would imagine running a complete 'reload' shouldn't be crashing out like this. if it wasn't meant to be done they surely wouldn't allow it |
| 23:50.33 | byronsmith | but i will raise a ticket tomorrow, many thanks for your time. much appreciated |
| 23:50.42 | byronsmith | cheers |
| 23:50.46 | p3nguin | Windows has a "reboot" mechanism, but that doesn't make it necessary to use it all the time. |
| 23:51.29 | byronsmith | no but you have to because it's windows |
| 23:51.30 | byronsmith | heh |
| 23:51.38 | p3nguin | Point taken. |
