| 00:03.24 | *** join/#asterisk talntid (n=swarm@66.208.251.166) |
| 00:05.59 | *** join/#asterisk dssman (n=no@CPE001d7e602900-CM0011aec52a9c.cpe.net.cable.rogers.com) |
| 00:06.24 | dssman | I have a question, do incomming and outgoing get defined in the same context? |
| 00:08.12 | [TK]D-Fender | dssman, depends if you point everything to that context |
| 00:10.47 | *** join/#asterisk ahbritto (n=guest@adsl-68-125-197-181.dsl.pltn13.pacbell.net) |
| 00:10.56 | AdamWest | it can, but typically should not |
| 00:11.23 | AdamWest | contexts can be thought of like security containers that control what someone can do |
| 00:11.32 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 00:13.41 | *** join/#asterisk nighty^ (n=nighty@210.188.173.246) |
| 00:15.56 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
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| 00:35.46 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 00:41.45 | rickross | anyone running * 1.6.0-beta6 ? |
| 00:44.13 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
| 00:46.37 | *** join/#asterisk jameswf-home (n=james@ip72-204-221-181.ph.ph.cox.net) |
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| 00:53.56 | *** join/#asterisk mmurdock (n=tga@c-98-202-154-226.hsd1.ut.comcast.net) |
| 00:54.11 | jameswf-home | knows he is not catholic... working on easter bah |
| 00:57.31 | jameswf-home | â â â â â â â |
| 00:57.40 | file | ooh, old Digium polos at the bottom of my dresser... sweet |
| 00:57.54 | jameswf-home | ebay? |
| 00:58.06 | Corydon76-dig | Happy zombie jesus day |
| 00:58.33 | file | pfft |
| 00:58.42 | Corydon76-dig | (He'll be eating your brain, now) |
| 00:58.54 | ManxPower | AARRRGGGHH!! AARRRGGGHHH!!! |
| 00:59.13 | outtolunc | we are pitiful <G> |
| 00:59.40 | Corydon76-dig | ManxPower: actually, describes what the problem is with a good many people... Zombie Jesus ate their brains... |
| 01:00.06 | ManxPower | <-- devout atheist |
| 01:01.48 | [TK]D-Fender | <-- moderate extremist |
| 01:01.57 | *** join/#asterisk tengulre (n=tengulre@124.42.50.9) |
| 01:02.18 | *** join/#asterisk osiris (n=osiris@c-71-205-29-230.hsd1.mi.comcast.net) |
| 01:02.49 | Corydon76-dig | <---- passionate apathist |
| 01:02.54 | outtolunc | <-- too broke to have gone anywhere (but to the movies earlier) |
| 01:03.23 | jameswf-home | the dislexic agnostic wasnt sure if there was a dog |
| 01:08.24 | *** join/#asterisk obnauticus (n=obnautic@c-67-160-183-109.hsd1.wa.comcast.net) |
| 01:12.08 | drmessano | I hope MegaJesus flattens your cities.. you people make me proud |
| 01:12.52 | jameswf-home | I like cute baby jesus |
| 01:14.44 | *** join/#asterisk sun_moon (n=RaviRaja@ip72-206-113-190.om.om.cox.net) |
| 01:14.54 | jameswf-home | : I like to think of Jesus as a mischievous badger. |
| 01:15.17 | jameswf-home | Dear Lord baby Jesus, or as our brothers in the south call you, "J |
| 01:15.29 | *** part/#asterisk sun_moon (n=RaviRaja@ip72-206-113-190.om.om.cox.net) |
| 01:15.32 | jameswf-home | *ezus |
| 01:15.59 | jameswf-home | Dear tiny infant Jesus... |
| 01:16.00 | jer | jameswf-home, funny movie... more funny when it first came out though |
| 01:16.23 | jameswf-home | yeah but probably one of the greatest sceins |
| 01:16.30 | jer | officially gives up on this PRI |
| 01:16.35 | jer | let it become someone else's problem |
| 01:16.51 | *** join/#asterisk FireMac (n=firemac@CPE000d88ae88b9-CM0011ae8bb0ee.cpe.net.cable.rogers.com) |
| 01:18.01 | drmessano | ZOMG |
| 01:18.17 | drmessano | The Effing ad on Trixbox.org looks like something from a grunge video |
| 01:21.38 | drmessano | Uhhh |
| 01:21.45 | drmessano | This crap is disturbing |
| 01:21.59 | *** join/#asterisk osiris (n=osiris@c-71-205-29-230.hsd1.mi.comcast.net) |
| 01:22.00 | drmessano | Trixbox Appliance with "Sexy Box" above it |
| 01:22.24 | drmessano | Thats gonna end badly |
| 01:26.19 | jameswf-home | someone commented it looks like they have michael jackson |
| 01:26.24 | Greek-Boy | i have set my astrundir to be /var/run/asterisk |
| 01:26.32 | Greek-Boy | but when i use stop gracefully |
| 01:26.52 | Greek-Boy | it looks for the pid file in /var/run/asterisk.pid |
| 01:29.02 | obnauticus | ln -s /var/run/asterisk/ /var/run/asterisk.pid |
| 01:29.03 | obnauticus | lol |
| 01:29.27 | jameswf-home | Greek-Boy: look lower in asterisk.conf I think its set in 2 places |
| 01:29.31 | Greek-Boy | it shouldn't have to be that way lol |
| 01:30.29 | *** join/#asterisk Kumbang (n=dsp@rusnas.paume.itb.ac.id) |
| 01:30.49 | jameswf-home | I cant remember off hand but I rremember doing it in to places |
| 01:31.15 | obnauticus | two* |
| 01:31.37 | jameswf-home | yeah two too |
| 01:31.41 | jameswf-home | :) |
| 01:32.08 | jameswf-home | oh are you using freepbx? |
| 01:32.44 | obnauticus | throws up |
| 01:34.07 | jameswf-home | Thats what it was if you run freepbx it is defined in amportal.conf and asterisk.conf .... after changing it in asterisk.conf you may have to do a killall asterisk and restart it |
| 01:41.09 | Greek-Boy | Asterisk died with code 1. |
| 01:43.01 | *** join/#asterisk ngvoice (n=linxrout@58.186.179.125) |
| 01:43.22 | ngvoice | hello there |
| 01:43.29 | ngvoice | i want to do IVR dial plan like this : incomming call, Press one for blah , Press two for blah.. |
| 01:43.36 | ngvoice | and after press one |
| 01:43.44 | ngvoice | another press 1 for this option , press two for this option |
| 01:43.51 | ngvoice | <PROTECTED> |
| 01:44.06 | mmurdock | ngvoice: The Book has a great example for doing that. |
| 01:44.17 | ngvoice | can anyone please point out |
| 01:44.35 | mmurdock | ~thebook |
| 01:44.35 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
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| 01:53.20 | Katty | anyone know of a nice php based calendar thingy i can add to my server? (kinda like gmail calendar) |
| 03:04.41 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 03:04.41 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.0-beta6, 1.4.18.1, 1.4.19-rc3, 1.2.27 (2008/03/18), *-Addons 1.4.6, 1.6.0-beta2 (2008/02/21), Zaptel 1.4.9.2 (2008/02/28), Libpri 1.4.3 (2007/12/13) -=- #asterisknow or #asterisk-gui for AsteriskNOW (asterisknow.org) -=- #switchvox for Switchvox (switchvox.com) -=- #freepbx for FreePBX/trixbox -=- Beware of zombies. |
| 03:05.02 | ngvoice | hic |
| 03:05.09 | ngvoice | can anyone help please |
| 03:05.13 | ngvoice | i want to do IVR dial plan like this : incomming call, Press one for blah , Press two for blah.. |
| 03:05.17 | ngvoice | and after press one |
| 03:05.20 | ngvoice | another press 1 for this option , press two for this option |
| 03:05.25 | ngvoice | <PROTECTED> |
| 03:06.32 | rkeene | Read the book. |
| 03:06.48 | obnauticus | lopl |
| 03:06.55 | ngvoice | i read the book |
| 03:06.57 | ngvoice | aldready |
| 03:07.01 | ngvoice | still does not work out |
| 03:07.45 | drmessano | Then read it more |
| 03:08.22 | ngvoice | never mind |
| 03:09.57 | [TK]D-Fender | ngvoice, "core show application waitexten" , "core show function TIMEOUT" |
| 03:10.09 | jameswf-home | ~ch5 |
| 03:10.10 | jbot | Read about extensions DialPlans etc.. in Chapter 5 of Asterisk: The Future of Telephony 2nd Edition http://www.oreilly.com/catalog/9780596510480/ |
| 03:10.14 | jameswf-home | yay bot |
| 03:10.14 | [TK]D-Fender | ngvoice, "core show application background" |
| 03:10.36 | jameswf-home | ~ch6 |
| 03:10.36 | jbot | Read about Advanced extensions DialPlans etc.. in Chapter 6 of Asterisk: The Future of Telephony 2nd Edition http://www.oreilly.com/catalog/9780596510480/ |
| 03:11.44 | obnauticus | ~fleamarket |
| 03:11.45 | jbot | Fleamarket its just like, its just like a mini mall http://www.youtube.com/watch?v=ULgwbvj768E |
| 03:14.12 | *** join/#asterisk xtr (i=94752345@216.19.191.191.novuscom.net) |
| 03:18.01 | *** join/#asterisk lowlevel (n=Stuart@lowlevel.ca) |
| 03:25.38 | *** join/#asterisk carlos_ (n=carlos@ool-457640b7.dyn.optonline.net) |
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| 03:25.57 | carlos_ | Can someone please help me ? |
| 03:26.39 | riddlebox | cant help you if you dont say what you need help with |
| 03:26.54 | carlos_ | how can I see if my nic is set to full duplex |
| 03:27.21 | carlos_ | if its not set to full duplex how can I change it on AsteriskNow |
| 03:27.45 | carlos_ | Im havind a horrible time with sound on my LAN |
| 03:27.52 | carlos_ | having* |
| 03:28.13 | [TK]D-Fender | carlos_, describe the complete call with poor quality in detail. |
| 03:28.19 | riddlebox | http://ots.iit.edu/network/linux_fullduplex.php |
| 03:28.38 | carlos_ | let me check that! appreciate it. |
| 03:29.15 | [TK]D-Fender | carlos_, and I highly doubt thats an issue.... |
| 03:33.58 | carlos_ | Im sorry. on cli whats the command to edit that /etc/sysconfig ........... file ? |
| 03:35.19 | *** join/#asterisk coppice (n=chatzill@175.203.17.210.dyn.pacific.net.hk) |
| 03:38.52 | *** join/#asterisk mmurdock (n=tga@c-98-202-154-226.hsd1.ut.comcast.net) |
| 03:39.59 | mmurdock | Has anyone here worked with asterisk manager call origination? |
| 03:40.01 | [TK]D-Fender | carlos_, take your pick of any CLI text editor. |
| 03:40.19 | [TK]D-Fender | carlos_, pico, nano, vi, vim, emacs, etc... |
| 03:40.42 | [TK]D-Fender | mmurdock, Ask a qpecific question and you might get a better answer |
| 03:41.00 | mmurdock | I'm having an issue trying to track the calls progress after the call has been initiated. |
| 03:41.42 | mmurdock | I do not see a way to link the status messages with the initated call. |
| 03:42.09 | [TK]D-Fender | mmurdock, what "status messages" and track via what? |
| 03:42.46 | mmurdock | The status messages would be the messages that show up in the Asterisk Manager |
| 03:43.23 | mmurdock | And I want to track a calls status that I initiated through the Asterisk Manager. |
| 03:43.50 | mmurdock | Does that make any sense? |
| 03:44.05 | [TK]D-Fender | mmurdock, Then set an identifying channel variable. |
| 03:45.44 | mmurdock | I don't follow. |
| 03:46.15 | coppice | oh, a leader :-) |
| 03:46.49 | mmurdock | I send variables when I initiate the call, but they are not returned in the Asterisk Manager. |
| 03:46.57 | mmurdock | coppice: ha ha |
| 03:47.01 | mmurdock | coppice: :) |
| 03:48.24 | *** join/#asterisk brunner (n=chris@75-143-111-62.dhcp.aubn.al.charter.com) |
| 03:48.28 | brunner | are there any voip services that call the international number and then call you and connect the two? |
| 03:48.49 | [TK]D-Fender | mmurdock, What you do is take the channel name of those messages and query active channels to see if its for the one you're tracking. |
| 03:55.49 | mmurdock | [TK]D-Fender: I think I got it. I do a getvar action from the Asterisk Manager against a channel that shows up and query one of the variables I use when I originate the call. |
| 03:56.16 | [TK]D-Fender | mmurdock, Congratulations, take the plushie of your choice from the back rack. |
| 03:56.44 | mmurdock | [TK]D-Fender: To bad the actionid does not follow through for the whole call. That would be cleaner. |
| 03:59.47 | *** part/#asterisk brunner (n=chris@75-143-111-62.dhcp.aubn.al.charter.com) |
| 04:02.48 | [TK]D-Fender | mmurdock, Well you have your means.... |
| 04:04.57 | mmurdock | [TK]D-Fender: eh? |
| 04:05.37 | [TK]D-Fender | mmurdock, You have a way to do what you were looking for. It may not be pretty, but its usable. |
| 04:06.54 | *** join/#asterisk lowtone2 (n=lowtone@ip98-165-136-222.ph.ph.cox.net) |
| 04:07.21 | mmurdock | [TK]D-Fender: Agreed, thanks for your help. |
| 04:11.15 | carlos_ | thanks tk I will try pico editor |
| 04:12.18 | jameswf-home | uses nano |
| 04:12.40 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-866dc84b75d032a4) |
| 04:12.59 | jameswf-home | lowtone2: an az local w00t |
| 04:15.02 | coppice | has anyone written femto yet? |
| 04:15.25 | carlos_ | hey guys im using the asterisknow package and do not know how the heck to get into the shell. I read something like booting from the cd in rescue mode but that doesnt sound right any help ? |
| 04:15.39 | carlos_ | i mean go to the shell from CLI |
| 04:15.55 | [TK]D-Fender | carlos_, Please join #asterisknow for support with your distro |
| 04:16.05 | carlos_ | ok thanks. |
| 04:16.16 | carlos_ | join #asterisknow |
| 04:16.32 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
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| 04:28.54 | jameswf-home | I would have said boot to single |
| 04:39.57 | *** join/#asterisk tengulre11 (n=tengulre@124.42.50.9) |
| 04:42.01 | jameswf-home | have to get stupid blackberry wapped |
| 04:42.09 | jameswf-home | *swapped |
| 04:43.43 | coppice | swiped |
| 04:43.49 | coppice | swatted |
| 04:44.03 | coppice | don't blackberries rot rather quickly? |
| 04:44.08 | obnauticus | Yes. |
| 04:44.21 | obnauticus | they do not have a very thick protien shell around the sugars inside them |
| 04:44.47 | obnauticus | therefore they rot faster after they die |
| 04:46.06 | obnauticus | Is asterisk the most popular open telephony platform? |
| 04:48.39 | *** join/#asterisk shital (n=shital@122.167.101.164) |
| 04:50.49 | obnauticus | <PROTECTED> |
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| 04:58.30 | coppice | There's also the Dante scale |
| 04:59.50 | obnauticus | !!!3!!! |
| 05:03.39 | *** join/#asterisk quigon (n=matias@190.3.121.15) |
| 05:04.00 | obnauticus | coppice i like wtfs/min |
| 05:05.47 | obnauticus | herm |
| 05:05.55 | obnauticus | is there an erlang projection graphing plugin for asterisk. |
| 05:06.37 | coppice | erlang projection? you mean like firing him from a cannon, or something? |
| 05:06.38 | Nasra | can you post the link for future of telephony in pdf format...so I can have as my favorite... |
| 05:06.55 | jameswf-home | ~book |
| 05:06.56 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 05:07.01 | obnauticus | coppice i assume you don't know what an erlang is? |
| 05:07.18 | coppice | you assume wrong, grasshopper |
| 05:07.32 | Nasra | hey thanks alot |
| 05:07.58 | coppice | I'm British. I know all there is about queuing |
| 05:08.31 | obnauticus | I'm half chinese, I know all there is about science, math, and technology. |
| 05:08.38 | obnauticus | And the dynamics behind the 4.01GPA |
| 05:08.52 | obnauticus | And deviding by zero thery. |
| 05:09.14 | obnauticus | And being able to play the piano, violin, viola, and chello simultaneously, |
| 05:09.20 | coppice | which half is chinese? |
| 05:09.25 | obnauticus | not the cock. |
| 05:09.34 | [TK]D-Fender | 's money is on the left half... |
| 05:09.54 | coppice | my half chinese son has also been fortunate in that respect :-) |
| 05:10.01 | obnauticus | lol |
| 05:10.03 | obnauticus | im chinese and irish |
| 05:10.04 | obnauticus | :P |
| 05:10.14 | obnauticus | one screwed up combo if you ask me |
| 05:10.33 | [TK]D-Fender | alrighty... time to check out... later all |
| 05:11.18 | jameswf-home | I am German/Irish/English.... so I am a self hating drunk who has bad teeth and drinks warm beer thats so thick you need a spoon.... |
| 05:11.58 | Corydon76-dig | jameswf-home: me, too, except that I also have some Scottish |
| 05:12.14 | obnauticus | ooo sexy. |
| 05:12.42 | obnauticus | man these earcups on my Sony MDR V700's are freaking nasty |
| 05:12.47 | Corydon76-dig | So I have all those attributes, plus I enjoy music for the sheer annoyance it has on other people |
| 05:13.10 | coppice | ooh, a Britney fan |
| 05:13.21 | jameswf-home | Corydon76-dig: a self hating drunk who has bad teeth and drinks warm beer thats so thick you need a spoon and he wears a dress :).... |
| 05:13.39 | obnauticus | you too?1 |
| 05:13.55 | Corydon76-dig | I like warm beer... Cold negates a good amount of the flavor |
| 05:14.36 | Corydon76-dig | well, not warm, really. Room temperature. It's still 30 degrees cooler than body temp |
| 05:15.53 | obnauticus | http://www.newegg.com/Product/Product.aspx?Item=N82E16826402010 |
| 05:15.55 | obnauticus | omg. |
| 05:18.15 | jameswf-home | I want http://www.thinkgeek.com/gadgets/watches/6a17/ |
| 05:18.44 | obnauticus | I have the wiFi t-shirt |
| 05:19.17 | obnauticus | http://www.thinkgeek.com/gadgets/tools/7ef8/ |
| 05:19.19 | obnauticus | OMG |
| 05:21.34 | obnauticus | can |
| 05:21.45 | obnauticus | can't one just like, get a crimper and an end and do that much easier. |
| 05:23.28 | coppice | crimper? are hairdressers good at making cables? |
| 05:23.40 | obnauticus | ... |
| 05:23.45 | obnauticus | stops talking to coppice. |
| 05:23.54 | jameswf-home | I have an RJ48 superlooper.... these are handy cause you put tm inline and loop up both ends at the same time |
| 05:24.23 | obnauticus | I have a Sun SoE adapter to Cisco soE adapter |
| 05:24.29 | obnauticus | ...adapter. |
| 05:25.05 | obnauticus | and it's for sun blades iirc. |
| 05:27.14 | jameswf-home | I missed the history f the internet yesterday.... I wanted to see al gore |
| 05:43.17 | jameswf-home | why do people sing along at concerts... dude I didnt pay $100 a ticket to hear you I paid for the band let them sing and STFU |
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| 06:03.45 | newbie___ | hi all, i am trying to connect to a voice provider using sip. all i am given is a username, password and ip address. i tested with Linksys ATA and is workind fine, i tested with xten and no problem. but when use * to dial out, i get no tone. any ideas ? |
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| 06:41.35 | *** mode/#asterisk [+ooo bkruse Corydon76-dig Corydon76-vcch] by irc.freenode.net |
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| 06:43.40 | obnauticus | drmessano <34 |
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| 07:25.54 | ice_croft | hi all |
| 07:26.17 | ice_croft | my planet vip-150 has some strange behavoir |
| 07:26.57 | ice_croft | it works well, then suddenly cant dial any number with SIP/2.0 407 Proxy Authentication Required |
| 07:27.52 | ice_croft | it works well for some time, then starts raisin errors again |
| 07:28.01 | ice_croft | what is it |
| 07:28.03 | ice_croft | ? |
| 07:28.13 | ice_croft | sip show peers shows OK |
| 07:33.20 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 08:16.00 | MooingLemur | anyone ever try to get linksys/sipura phones to provision over HTTPS? |
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| 08:58.03 | ngbzxc | hi people.. how are u? |
| 08:59.17 | ngbzxc | some one know where i can get tuning asterisk tutorial ? |
| 08:59.22 | ngbzxc | some one know where i can get tuning asterisk tutorial ? |
| 09:01.20 | *** join/#asterisk mvanbaak (n=michiel@vanbaak.xs4all.nl) |
| 09:05.48 | Nasra | !book |
| 09:25.34 | Nasra | !o'reilly |
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| 10:23.47 | *** join/#asterisk dimgr (n=dimgr@athedsl-30662.home.otenet.gr) |
| 10:24.01 | dimgr | hello i have a question ;) |
| 10:25.20 | dimgr | I have an IP phone at home which is working fine with a voip provider. If i set up an asterisk pbx at home using 1 fxo card that will connect to that ip pohone what are the benefits that i will notice using asterisk pbx ( with 1 phone only) than without asterik |
| 10:25.51 | mvanbaak | all the neat stuff you can do with asterisk |
| 10:27.06 | dimgr | ax-100 is the card i will get .. |
| 10:28.22 | dimgr | but can you takjke full advantage of asterik if you only use 1 phone ? |
| 10:30.18 | mort_gib | dimgr: "Full advantage" -Like you want to do a call centre in your house ;-) |
| 10:30.30 | dimgr | yes why not;) |
| 10:30.55 | mort_gib | Exactly! |
| 10:31.30 | dimgr | http://www.voip-info.kinghost.com.br/wiki/view/AX-100P.html |
| 10:31.34 | mort_gib | Like in my case... I can't get a lan line in my house, rural area and all |
| 10:31.50 | mort_gib | But I can get a GSM gateway, and an UTMS gateway |
| 10:32.16 | mort_gib | So my home asterisk gives me 4 handsets, in different areas of the house. |
| 10:32.27 | *** join/#asterisk acidfu (n=acidfu@59.94.237.191) |
| 10:32.29 | acidfu | hi guys! |
| 10:32.40 | acidfu | I'm looking for a linux sip phone, someone have a suggestion ? |
| 10:32.50 | dimgr | so assuming there are 4 phones in the house but only one is connected to asterisk ... if you call my house will you hear the mpe on hold from asterisk or not? |
| 10:32.54 | mort_gib | It never uses the GSM gatweway, unless UMTS is down, but accepts incoming calls on the GSM gateway |
| 10:33.20 | dimgr | mpe = mp3;) |
| 10:33.53 | acidfu | dimgr, the 3 others are connecte to what ? |
| 10:33.57 | acidfu | conected* |
| 10:33.59 | acidfu | damn |
| 10:34.18 | dimgr | wall phone plugs;) |
| 10:35.15 | acidfu | you have an fxo into your pbx ? |
| 10:35.26 | acidfu | connected to the wall phone plugs ? |
| 10:35.42 | dimgr | i will get an fxo card |
| 10:36.07 | acidfu | so I will hear the song if the pbx answer |
| 10:36.22 | acidfu | its like if you pickup the phone and you start singing |
| 10:36.43 | JT | forget about the AX-100P |
| 10:36.48 | JT | another X100P clone |
| 10:36.52 | acidfu | yes, thats junk |
| 10:36.58 | JT | all X100P related stuff is junk really |
| 10:37.42 | dimgr | find me a good pstn card then with 1 fxo port |
| 10:37.59 | acidfu | ok pay me 50$/hour for doing your work |
| 10:38.00 | acidfu | ;) |
| 10:38.01 | JT | forget about them if 1 port is your requirement |
| 10:38.08 | JT | buy a linksys SPA-3102 |
| 10:38.12 | dimgr | ;) |
| 10:38.13 | acidfu | dimgr, you need an fxo/fxs card |
| 10:38.23 | JT | you don't need a card |
| 10:38.27 | JT | an ATA is fine |
| 10:38.29 | acidfu | dimgr, if you are poor, lazy, and hopeless |
| 10:38.36 | acidfu | what can we help you :/ |
| 10:38.40 | acidfu | who |
| 10:38.58 | acidfu | how |
| 10:38.58 | acidfu | damn |
| 10:39.40 | dimgr | when i say cheap i mean a reasonable price |
| 10:41.01 | acidfu | search tdm fxo fxs |
| 10:41.02 | acidfu | on google |
| 10:41.13 | JT | as i said |
| 10:41.19 | JT | by an SPA-3102 or similar |
| 10:42.13 | dimgr | i want astetrisk ;) |
| 10:42.28 | JT | and? |
| 10:42.32 | JT | it works with asterisk |
| 10:42.36 | JT | ~thebook |
| 10:42.37 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 10:44.09 | *** join/#asterisk iso (n=allen@dante.smarterits.com) |
| 10:51.24 | FreezeS | hey guys |
| 10:51.42 | FreezeS | is there a way for asterisk to send sip keepalive messages to its registered peers ? |
| 10:52.07 | FreezeS | in order to detect if a client has dissapeared in the middle of the call |
| 10:53.55 | viperdude | free |
| 10:54.31 | viperdude | FreezeS: you can detect if the RTP stream has stopped |
| 10:54.49 | dimgr | Digium TDM 400P 1 FXS 1 FXO |
| 10:54.58 | FreezeS | viperdude: what if the other part keeps sending an RTP stream ? |
| 10:55.31 | FreezeS | or the rtp stream doesn't pass through asterisk at all ? |
| 10:55.45 | viperdude | FreezeS: I believe it works one way |
| 10:56.48 | viperdude | http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+rtptimeout |
| 10:58.54 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 10:59.35 | FreezeS | viperdude: I knew about that, but I was wondering if there is something like it for SIP |
| 11:00.16 | JT | damn that guy was stubborn |
| 11:00.18 | viperdude | well are you trying to detect if the UA is there while on a call or while hung up? |
| 11:00.26 | JT | told him the cheapest best quality option, he ignored it |
| 11:00.27 | JT | ah well |
| 11:00.48 | FreezeS | I'm trying to detect if the UA is there at any given time |
| 11:01.13 | FreezeS | we had a few software glitches and got some calls left hanging for days because of that |
| 11:01.19 | acidfu | but SIP is only signaling, isnt ? |
| 11:01.19 | FreezeS | and the RTP timeout was enabled |
| 11:01.20 | acidfu | and then the conversation goes through rtp |
| 11:01.39 | viperdude | lol use decent software then |
| 11:02.08 | viperdude | you can use AGI to detect zombie channels and soft hangup them |
| 11:02.19 | FreezeS | viperdude: thanks for the suggestion but I can't rely on the clients behaving properly |
| 11:02.30 | FreezeS | yes, that's what we are doing now |
| 11:02.38 | FreezeS | but it's not very elegant |
| 11:03.03 | viperdude | FreezeS: IMHO don't rely on anything you dont control |
| 11:03.15 | viperdude | thats why i like the AGI solution |
| 11:03.21 | FreezeS | even the things you think you control can fail |
| 11:03.37 | viperdude | true but you also have the power to fix them |
| 11:03.38 | FreezeS | that's why I'm trying to implement multiple checks |
| 11:03.54 | FreezeS | and this SIP keepalive feature would have been golden |
| 11:07.40 | *** join/#asterisk Bonix-BR (i=Bonix-BR@217-lo1.rt2.isimples.com.br) |
| 11:07.57 | viperdude | well QUALIFY will do the job for future calls but not for current calls |
| 11:08.19 | FreezeS | what do you mean by "future calls" ? |
| 11:08.49 | viperdude | when someone trys to call in 30 mins |
| 11:09.10 | FreezeS | well, this is useless, because most probably the UA was restarted |
| 11:09.13 | viperdude | http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+qualify |
| 11:09.24 | FreezeS | and the call when it crashed is still left running |
| 11:09.31 | FreezeS | because it never sent the hangup message |
| 11:09.52 | viperdude | if rtptimeout doesn't get it then you need AGI |
| 11:10.31 | viperdude | i handle 100000 calls a week on my asterisk boxes and get that issue maybe once or twice a month |
| 11:10.32 | *** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
| 11:10.46 | FreezeS | me too |
| 11:10.46 | viperdude | hardly a issue |
| 11:11.00 | FreezeS | but that single call left hanging could cost as much as 3000 euro |
| 11:11.07 | viperdude | huh? |
| 11:11.10 | FreezeS | and that is a serious issue |
| 11:11.15 | *** join/#asterisk _ys (i=yuri@91.151.196.254) |
| 11:11.16 | viperdude | not fore me |
| 11:11.20 | viperdude | for me |
| 11:11.25 | FreezeS | :) |
| 11:11.42 | viperdude | my AGI gets em everytime |
| 11:13.50 | acidfu | you mean when the conversation is done, if someone forget to hangout the phone |
| 11:13.55 | acidfu | you want to know ? |
| 11:14.03 | acidfu | handoff |
| 11:14.06 | acidfu | hangoff |
| 11:14.20 | FreezeS | yes, but I found out about this the hard way |
| 11:14.41 | FreezeS | yes acidfu, this is the problem |
| 11:14.51 | FreezeS | if a client crashes, it never sends the hangup signal |
| 11:15.02 | FreezeS | and the call is left running indefinetly |
| 11:15.03 | acidfu | oh |
| 11:15.08 | acidfu | I see |
| 11:18.35 | acidfu | FreezeS, actually what are you doing to fix it ? |
| 11:18.54 | acidfu | you have an asterisk plugin, a watchdog that send an hangup signal ? |
| 11:19.04 | FreezeS | I have a script that checks if an unregistered user has any channels |
| 11:19.46 | acidfu | ok |
| 11:19.57 | acidfu | and ifs the case ? |
| 11:20.11 | FreezeS | if they are running for more than 10 minutes I close the channels |
| 11:20.27 | FreezeS | but it's an external application |
| 11:20.39 | FreezeS | that can fail for any reason |
| 11:21.22 | acidfu | yes |
| 11:21.28 | acidfu | at this point i'm not experienced enough to aswer this question |
| 11:21.39 | FreezeS | there should be an option like "qualify" that would hang up all the channels if the UA doesn't answer |
| 11:22.11 | FreezeS | I'll see if I could modify the sourcecode myself and submit a patch |
| 11:22.30 | acidfu | you're using the last svn code ? |
| 11:22.36 | acidfu | and the last drivers ? |
| 11:22.41 | FreezeS | no, I'm on 1.4.18-1 |
| 11:22.46 | *** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org) |
| 11:22.52 | FreezeS | it's a production server |
| 11:23.21 | *** join/#asterisk bsaxon (n=bryantsa@71-8-14-108.dhcp.leds.al.charter.com) |
| 11:24.10 | FreezeS | I really hate they don't have on asterisk.org a list with the changes from the last version |
| 11:24.18 | FreezeS | I need to download it and check the file there |
| 11:24.20 | FreezeS | sucks |
| 11:24.36 | acidfu | there's probably a changelog somehwere |
| 11:25.00 | FreezeS | I couldn't find it on the site |
| 11:25.07 | acidfu | ok |
| 11:25.09 | FreezeS | only in a file inside the archive |
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| 11:56.24 | maxo22 | hi, i could not believe... using 'sip show registry' will print 'registered' and i could call people, but poeple could not call me |
| 11:56.28 | maxo22 | some ideas? |
| 12:03.00 | acidfu | friend |
| 12:03.25 | acidfu | type=friend |
| 12:04.26 | acidfu | maybe... |
| 12:08.23 | yang | maxo22: your extensions |
| 12:09.14 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
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| 12:12.52 | maxo22 | hi |
| 12:12.53 | maxo22 | k |
| 12:13.02 | maxo22 | i tried it with sipgate |
| 12:13.05 | maxo22 | there it worked |
| 12:13.19 | maxo22 | and then with 1und1 and ther it would not work :( |
| 12:14.06 | maxo22 | register = 4923456789:12345678@sip.1und1.de/001 |
| 12:14.41 | maxo22 | exten = 001,1,Dial(Console/dsp) |
| 12:14.43 | maxo22 | for debug |
| 12:15.39 | maxo22 | and show register gaves me 'sip.1und1.de:5060 4923456789 28248 Registered' |
| 12:20.32 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
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| 12:49.35 | *** mode/#asterisk [+o corydon76-dig] by ChanServ |
| 12:52.54 | jm|home | can you use OR in if conditionals in ael? |
| 12:53.33 | jm|home | ah |
| 12:53.34 | jm|home | || ? |
| 12:53.43 | jm|home | hm. |
| 12:54.12 | *** join/#asterisk ProofPack (n=no@208.255.225.67) |
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| 13:00.28 | *** join/#asterisk Katty (n=Katty@hera.copi-rite.com) |
| 13:01.08 | Katty | mew. |
| 13:01.30 | jm|home | proffers catnip |
| 13:02.22 | *** join/#asterisk MihiNomenEst (n=argh@cerebus.clandestineresearch.com) |
| 13:03.26 | annielou | Has anyone been able to build Asterisk with a Voicetronix OpenLine4 card? |
| 13:03.28 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
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| 13:10.19 | lmadsen | jm|home: should be able to -- it's just a single pipe though |
| 13:10.27 | jm|home | oh |
| 13:10.28 | lmadsen | & is AND, | is OR |
| 13:10.32 | jm|home | cool |
| 13:10.34 | jm|home | fixes |
| 13:10.40 | lmadsen | = is equal to |
| 13:11.02 | jm|home | thanks |
| 13:11.19 | lmadsen | np |
| 13:16.48 | *** join/#asterisk trelane (i=trelane@denali.sosdg.org) |
| 13:17.25 | Katty | i'm awake. |
| 13:17.27 | Katty | nodsnods |
| 13:17.29 | atomspace | is it possible to use an extension on two phones? --- and when that extension rings, have both phones ring? |
| 13:17.44 | atomspace | well, guten tag Katty |
| 13:17.51 | trelane | hey, I've got a trixbox 1.4 connecting to a BTS10200, but I don't think I'm dealing with a trixbox related issue, as it's strictly an RTP problem. I've got a sip trace (see URL), and was wondering if someone would look at it. http://pastebin.com/m76050c0d. It looks like everything works except I'm not getting any RTP data. There are no firewalls involved here, and the connection tears up and down correctly, I STFW'd but didn't see anything on RTP i |
| 13:17.54 | Katty | guten tag! |
| 13:18.01 | Katty | vie heist du (spelling optional) |
| 13:18.19 | Katty | jbot: rtp? |
| 13:18.19 | jbot | it has been said that rtp is The Internet-standard protocol for the transport of real-time data, including audio and video. RTP is used in virtually all voice-over-IP architectures, for videoconferencing, media-on-demand, and other applications. A thin protocol, it supports content identification, timing reconstruction, and detection of lost packets. |
| 13:18.47 | atomspace | du hast mesh ? ... I don't speak German, I just know 'good morning' - my apologies |
| 13:18.56 | atomspace | I hope you don't hate me |
| 13:18.59 | Katty | lol |
| 13:19.02 | Katty | no (= |
| 13:19.05 | trelane | Katty: yep, that's not the problem, is there anything in 1.4 that lets me debug rtp and possibly see why the RTP connection isn't establishing? |
| 13:19.07 | Katty | i know more spanish than german. |
| 13:19.12 | *** join/#asterisk BipBip (n=BipBip@static-b5-253-32.telepac.pt) |
| 13:19.23 | Katty | atomspace: i've tried to get two polycoms to use the same extension, btw. no go. |
| 13:19.34 | atomspace | hmmm |
| 13:19.38 | Katty | atomspace: they spaz out and try to register over themselves every other second. |
| 13:20.01 | Katty | atomspace: what are you trying to accomplish? |
| 13:20.09 | atomspace | I'm using a grandstream gxp-2000 and a analog phone on an ATA ---- yea, thats my issue, they're both registered, but only 1 rings |
| 13:20.19 | Katty | exactly. |
| 13:20.27 | Katty | when the other takes registration, it kicks the other one off. |
| 13:20.28 | atomspace | but I can call out of both of them on that extension |
| 13:20.34 | atomspace | hmmm |
| 13:20.41 | trelane | atomspace: they aren't both registered, they just seem to be, give them different extensions and do a Dial(sip/polycomextension&sip/grandstreamextension) |
| 13:20.43 | Katty | why not Dial(SIP/moo&SIP/moo2) |
| 13:20.44 | trelane | it'll ring both |
| 13:20.50 | trelane | Katty: :) |
| 13:21.13 | atomspace | hmmm |
| 13:21.13 | trelane | jbot: bts |
| 13:21.14 | jbot | bts is probably a bug tracking system, or http://bugs.kde.org |
| 13:21.16 | trelane | jbot: bts10200 |
| 13:21.25 | trelane | guess he doesn't know that one |
| 13:21.52 | *** join/#asterisk DonAlex (n=DonAlex@host86-137-212-179.range86-137.btcentralplus.com) |
| 13:22.04 | trelane | jbot: BTS10200 is a class 5 redundant telephony switch system produced by cisco supporting MGCP and SIP. |
| 13:22.05 | jbot | okay, trelane |
| 13:22.09 | DonAlex | Hello all .. you wonderful people.. :) |
| 13:22.09 | trelane | <3! |
| 13:22.20 | trelane | tosses jbot a botsnack |
| 13:22.29 | Katty | morning don |
| 13:22.29 | trelane | older version or disabled? |
| 13:22.32 | trelane | jbot: botsnack |
| 13:22.32 | jbot | trelane: thanks |
| 13:22.33 | DonAlex | Anyone feel like putting their experience forth on a problem about CLID in the UK? |
| 13:22.46 | atomspace | unfortunately, I'm using *NOW, so I'm not as proficient with the config files as a gui-less * user - but I'm getting there...slowly. Katty, you said: Dial(SIP/moo&SIP/moo2) |
| 13:23.02 | *** join/#asterisk delparnel (n=delparne@balance.wlu.ca) |
| 13:23.03 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.141.193) |
| 13:23.04 | DonAlex | We 've hunted down all the usual suspects and flippedon cidstart-polarity etc etc.. |
| 13:23.24 | atomspace | Katty: "Dial(SIP/moo&SIP/moo2)" --- this would be in extensions.conf, I believe? |
| 13:23.30 | Katty | atomspace: yesh. |
| 13:23.32 | trelane | atomspace: Dial allows you to list things to dial, which can be anything really that's attached to the system. So it'll dial a whole list of extensions, or phone numbers, but will only connect the call to the first extension to give complete progress (the first person to answer) |
| 13:23.38 | DonAlex | but still having no joy on this BT Feature line set that behaves as if you were attached to a PBX (ie dial 9 to to get out) |
| 13:24.10 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 13:24.14 | atomspace | would there be a way for the two extensions to have a single voicemail ---- as if they were 1 extension? |
| 13:24.15 | trelane | aah a centrex line, DonAlex if you get that working someone should give you a medal! |
| 13:24.32 | [TK]D-Fender | atomspace: Yes, point each device to the same box |
| 13:24.45 | trelane | atomspace: absolutely, they're going to have a voicemail number, but just dial(sip/extb) voicemail(extA) |
| 13:24.46 | [TK]D-Fender | atomspace: and stop calling "devices" as "extensions". |
| 13:24.54 | Katty | atomspace: sure, just point them tot he smae voicemail account (= |
| 13:24.58 | Katty | [TK]D-Fender: YOU |
| 13:25.03 | Katty | [TK]D-Fender: why i outta...just... |
| 13:25.09 | Katty | hugs [TK]D-Fender |
| 13:25.12 | [TK]D-Fender | Katty: :) |
| 13:25.15 | [TK]D-Fender | Katty: Mew. |
| 13:25.23 | atomspace | ok, I think I follow. |
| 13:25.24 | [TK]D-Fender | Katty: Having fun with Horde yet? |
| 13:25.27 | DonAlex | trelane: Ahhh methinks there is some tale to be told here.. Pray .. what do ye know of this beast called Centex..? |
| 13:25.33 | Katty | [TK]D-Fender: no, but i bookmarked it. |
| 13:25.40 | Katty | [TK]D-Fender: i want to tinker with Monket too |
| 13:25.51 | [TK]D-Fender | Katty: I run it on my home server & domain. It is pretty... |
| 13:26.02 | Katty | [TK]D-Fender: it certainly looks it (= |
| 13:26.10 | trelane | DonAlex: they figured out how to emulate some basic phone system functions on a POTS line using the CO switch as a PBX. They're finicky and do wierd things |
| 13:26.18 | Katty | [TK]D-Fender: right now i'm thinking about openfire stuff |
| 13:26.32 | trelane | hey, I've got a trixbox 1.4 connecting to a BTS10200, but I don't think I'm dealing with a trixbox related issue, as it's strictly an RTP problem. I've got a sip trace (see URL), and was wondering if someone would look at it. http://pastebin.com/m76050c0d. It looks like everything works except I'm not getting any RTP data. There are no firewalls involved here, and the connection tears up and down correctly, I STFW'd but didn't see anything on RTP i |
| 13:26.47 | Katty | [TK]D-Fender: what did you do for easter sunday? |
| 13:27.00 | *** join/#asterisk Victor_Yure (n=Victor_Y@unaffiliated/victoryure/x-837844) |
| 13:27.12 | trelane | file: muffin |
| 13:27.28 | DonAlex | trelane: Indeed!!! Tell me about it?! I spent entire yesterday with a Xorcom top bod and we still have no joy? Any pointers as to how to vanquish such a challenge? Or plain just get centrex to fck of and die? ;) |
| 13:27.32 | Katty | trelane: he requires orange juice sacrafices too. |
| 13:27.57 | Katty | xorcom? |
| 13:28.04 | Katty | is that the linux with asterisk built in iso? |
| 13:28.05 | DonAlex | CLID is the biggest hurdle so far. . |
| 13:28.34 | glaz | asterisknow is the linux w/ asterisk distro. |
| 13:28.36 | DonAlex | Katty: No no .. tis and embedded box.. the XR1000.. you should have a look. quite neat in it's own way.. |
| 13:28.37 | [TK]D-Fender | Katty: I came into the ofice to complete a server migration, went to a clients place to fix a printer problem, and found out our jam night had been closed... |
| 13:28.39 | trelane | DonAlex: with a PBX, I'd eliminate the centrex stuff, call up your lec and tell 'em you want real pots lines and that you're moving your pbx functions inhouse, it's cheaper usually too (Don't know about BT because they're fscked up, but generally cheaper) |
| 13:28.43 | [TK]D-Fender | Katty: short weekend... |
| 13:28.45 | DonAlex | Katty: Running embedded linux of course |
| 13:29.05 | Katty | [TK]D-Fender: :< |
| 13:29.09 | Katty | [TK]D-Fender: no fluffy bunny chasing? |
| 13:29.13 | Katty | [TK]D-Fender: or kid tickling? |
| 13:29.24 | Katty | [TK]D-Fender: did you get any chocolate?! |
| 13:29.42 | DonAlex | trelane: Hmmm what I get narked off by is that when I plug in a synergy 4500 stright into thte line IT recognises the CLID.. but asterisk is left out in limbo over it |
| 13:30.02 | trelane | DonAlex: what are you using to interface the lines? Digium? Sangoma? |
| 13:30.12 | [TK]D-Fender | Katty: Yeah the GF got a small box of little egg-shaped chocolates. They didn't last long :) |
| 13:30.19 | DonAlex | trelane: Xorcom.. XR1000.. |
| 13:31.11 | DonAlex | Katty: Awww should have said could have sent over my Cadburys Egg I have not touched.. not been in the mood to celebrate with chocolate yet :( |
| 13:31.23 | trelane | DonAlex: I havn't used that product, so I'm not familiar with the hardware in it. This might not be an asterisk issue, but I can't say either way for sure. All I know is I always eliminte centrex lines when I find 'em |
| 13:31.53 | trelane | http://en.wikipedia.org/wiki/Centrex |
| 13:31.57 | DonAlex | trelane: Well that is good to know.. but will the DID number pooling be elimanted too? |
| 13:32.04 | *** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org) |
| 13:32.08 | DonAlex | trelane: Atm they have 3 lines 1 number.. |
| 13:32.32 | *** join/#asterisk nighty^ (n=nighty@p5187-adsau17honb13-acca.tokyo.ocn.ne.jp) |
| 13:33.07 | trelane | no, the telco can set up a hunt group on normal POTS lines |
| 13:33.45 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 13:33.46 | *** join/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 13:33.49 | DonAlex | trelane: Yeah well they could have done that in the first place.. but you do not want to know the horror stories about the installation |
| 13:34.16 | trelane | DonAlex: I live in AT&T land, if it can be fucked up, it has been fucked up, and it will be fucked up again soon. |
| 13:34.23 | trelane | I _HATE_ those people |
| 13:34.27 | Katty | [TK]D-Fender: but did YOU get any? :P |
| 13:34.45 | Katty | DonAlex: how can you /not/ bein the mood for chocolate. you're insane. |
| 13:34.53 | [TK]D-Fender | Katty: Oh, sorry, got ME thoe chocolates.. |
| 13:35.00 | DonAlex | trelane: I hear you brother.. Sooner I get my client moved to VOIP the happier I will be.. |
| 13:35.09 | *** join/#asterisk boblutz (n=stansmit@adsl-68-252-62-180.dsl.wotnoh.ameritech.net) |
| 13:35.11 | trelane | DonAlex: VoIP has it's own issues |
| 13:35.15 | Katty | [TK]D-Fender: oh. right. |
| 13:35.28 | Katty | [TK]D-Fender: i helped a giddy 3 year old find eggs. |
| 13:35.36 | DonAlex | Katty: Insanity seems like a luxary with the maddness that is this stupid installation :( |
| 13:35.57 | DonAlex | trelane: To be sure.. but one of them is not Centrex!!? |
| 13:36.29 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
| 13:36.29 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:37.30 | boblutz | I replaced a tdm400p (1 fxo/1 fxs) with a tdm402b. I made the changes to zapata and zaptel conf files. When I run `ztcfg -vvvvv` I get "ZT_CHANCONFIG failed on channel 1: No such device or address (6)" --- any ideas? |
| 13:38.00 | trelane | boblutz: you loaded the modules right? |
| 13:38.12 | boblutz | trelane: correct, both cards use wctdm |
| 13:38.29 | boblutz | both have the same fxo modules |
| 13:38.36 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 13:38.45 | trelane | lspci -v and see if it's showing up, and that it's not sharing IRQ? |
| 13:39.17 | trelane | call digium and get free support? |
| 13:39.26 | trelane | forgot that those were a warranty setup/support item |
| 13:39.30 | boblutz | trelane: It usually shows up as "tiger" something or another under `lspci -v` right? |
| 13:39.36 | trelane | yes |
| 13:40.32 | boblutz | It is not showing up under `lspci -v`, so most likely an IRQ problem ? |
| 13:41.35 | trelane | irq problem, dead card, card needs to be reseated, you didn't bow 5 times to huntsville, AL before inserting it |
| 13:41.55 | [TK]D-Fender | boblutz: "modprobe wctdm" , "cat /proc/interrupts" , "ls -l /dev/zap", and pastebin your zaptel.conf & zapata.conf |
| 13:42.27 | trelane | [TK]D-Fender: I'll let you take over because I don't even know that it's gonna show up in any of those :/ |
| 13:44.26 | Katty | wonders which project to work on: monkey, horde.org, openfire/spark |
| 13:44.32 | Katty | s/monkey/monket |
| 13:45.14 | boblutz | LOL trelane - your right, i didnt bow 5 times to the east |
| 13:45.39 | boblutz | [TK]D-Fender: Ill PB in a second... I rebooted the server right quick |
| 13:45.42 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 13:46.03 | trelane | boblutz: make sure the card's seated in the PCI slot correctly |
| 13:46.49 | [TK]D-Fender | And you didn't replace your card so much as effective change the modules on one. |
| 13:47.23 | boblutz | trelane: Thats what I just did |
| 13:47.31 | boblutz | [TK]D-Fender: New card + new modules |
| 13:53.35 | polerin | I know this is way way off topic for the channel, but i'm trying to find help for a friend.... if anyone has experience with Avaya CMS Supervisor |
| 13:53.42 | polerin | ... can you drop me a line :P |
| 13:53.53 | boblutz | [TK]D-Fender: http://www.pastebin.ca/955234 |
| 13:53.59 | polerin | got an error that is hard to google for :P |
| 13:55.00 | [TK]D-Fender | boblutz: Where's the modprobe first? Oh yeah, and "ztcfg -vvvv" first, then modprobe, then "cat /proc/interrupts" |
| 13:55.10 | trelane | I've got a trixbox 1.4 connecting to a BTS10200, but I don't think I'm dealing with a trixbox related issue, as it's strictly an RTP problem. I've got a sip trace (see URL), and was wondering if someone would look at it. http://pastebin.com/m76050c0d. It looks like everything works except I'm not getting any RTP data. There are no firewalls involved here, and the connection tears up and down correctly, I STFW'd but didn't see anything on RTP issues |
| 13:59.41 | *** join/#asterisk RobH (n=RobH@cpe-72-184-129-136.tampabay.res.rr.com) |
| 14:01.56 | atomspace | I'm trying to setup two different phones to act as 1. Per Katty's recommendation, I have setup each phone with diff. extensions and put them together in a ring group. The last obstacle is getting these two extensions/devices to share 1 voicemail. How would one go about making two different devices (using two different extensions) use a single mailbox? |
| 14:02.05 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 14:02.51 | boblutz | [TK]D-Fender: http://www.pastebin.ca/955244 |
| 14:02.54 | trelane | oej: greetings :) |
| 14:03.11 | oej | Hello there :-) |
| 14:03.19 | polerin | atomspace: you assign them the same mailbox ;P |
| 14:05.04 | trelane | jbot: tell adomspace about book |
| 14:05.05 | atomspace | polerin: so a mailbox isn't tied to an extension? |
| 14:05.19 | trelane | jbot: tell atomspace about book |
| 14:05.21 | *** join/#asterisk wmaulik (n=wmaulik@158.59.192.218) |
| 14:05.34 | trelane | atomspace: I give you the gift of book, after reading book all will become clear! |
| 14:06.01 | atomspace | unfortunately, I've already read that. I'm guessing something didn't stick |
| 14:06.13 | atomspace | I'll search through it. thanks |
| 14:06.35 | trelane | atomspace: that would be a correct assumption, reread it including the exceptionally useful application reference in the rear |
| 14:06.46 | polerin | atomspace: remember, phone users are NOT extensions, extensions are more like functions |
| 14:07.13 | trelane | polerin: we have given him the book. It is in Leif's hands now. |
| 14:08.14 | Katty | [TK]D-Fender: make me photon bubble ship. |
| 14:08.50 | trelane | [TK]D-Fender: would you mind looking at the sip debug I posted, and comment, or help me figure out how to get some debug for RTP data? |
| 14:09.22 | [TK]D-Fender | trelane: If I had any to say about it, I would ahve... |
| 14:10.05 | [TK]D-Fender | boblutz: You may have to insmod for your card... |
| 14:10.23 | *** join/#asterisk edugas (i=edugas@ninjapants.org) |
| 14:11.14 | trelane | [TK]D-Fender: thanks for looking then |
| 14:12.00 | edugas | heya, I need some software/hardware on top or in Asterisk to filter incoming call by numbers, we'll have 8-5pm week day class support and 24/7/365 class support and will give the customer |
| 14:12.05 | edugas | different number all going to the NOC, we want to see to see what number they're calling so we don't answer if the customer's calling before 8 or after 5pm |
| 14:13.10 | JayTee52 | how can I take a wav file and format it for gsm? |
| 14:13.18 | [TK]D-Fender | edugas: "core show application GotoIfTime", "core show function CALLERID" |
| 14:13.18 | JayTee52 | or ulaw |
| 14:13.28 | [TK]D-Fender | JayTee52: "man sox" |
| 14:13.37 | edugas | [TK]D-Fender: oh, thanks! |
| 14:13.46 | Katty | there sure has been a lot of sox talk lately. |
| 14:13.46 | JayTee52 | [TK]D-Fender, thanks, man! |
| 14:13.56 | DonAlex | trelane: Thanks for you help mate.. Talked to a BT engineer and he explained how the 'Feature' line works.. and you are right surplus to requirements :) |
| 14:14.11 | DonAlex | trelane: I'll be talking to them tomorrow. |
| 14:14.22 | DonAlex | Later peeps.. !! |
| 14:15.36 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 14:15.36 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:16.25 | *** join/#asterisk RobH (n=RobH@cpe-72-184-129-136.tampabay.res.rr.com) |
| 14:24.46 | *** join/#asterisk [gnubie] (n=[gnubie]@cm32.omega116.maxonline.com.sg) |
| 14:25.06 | *** join/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 14:25.38 | grandpapadot | Is there a way with the dial command to specify a destination context? |
| 14:26.44 | *** join/#asterisk delparnel (n=delparne@balance.wlu.ca) |
| 14:27.42 | codefreeze | grandpapadot: depends. If the destination is asterisk, then the configs usually specify which context to associate with the incoming channel. |
| 14:28.24 | codefreeze | But some protocols allow you to specify context/exten |
| 14:28.30 | *** join/#asterisk hfb (n=hfb@pool-71-118-254-245.lsanca.dsl-w.verizon.net) |
| 14:28.46 | grandpapadot | Yea, I got that part. I have two hosts that are back to back, I can transfer calls between them no problem, but I was hoping for a way to dial to a specific context on the other host without having to route from a single destination context or have multiple sip.conf entries.. |
| 14:28.53 | grandpapadot | codefreeze: SIP? |
| 14:28.57 | grandpapadot | (Thanks for the help) |
| 14:29.10 | boblutz | [TK]D-Fender: I just uninstalled zaptel-1.4.8 and compiled zaptel-1.4.9.2, I noticed that when i did `make config`, it says "I thin kthat the zaptel hardware you have on your system is:", on the old card, it was identified |
| 14:29.17 | boblutz | definately IRQ issue, no? |
| 14:29.24 | *** join/#asterisk sacitec (n=tobi@201.144.211.82) |
| 14:31.08 | codefreeze | grandpapadot: you have me there. I remember making a fix to AEL because SIP allowed an extension name in the INVITE. whether you can also specify the context is a good question.... |
| 14:31.56 | grandpapadot | I'll try it and see... Do you know how your format the dial string? i.e., SIP/someext@somepeer, where would I specify the alternate context? I tried the wiki but no help. |
| 14:32.11 | codefreeze | grandpapadot: personally, I would consider allowing incoming callers to spec. the context would be a huge security flaw... |
| 14:32.50 | grandpapadot | It wouldn't be, this is a special consideration and will be hard-coded in the dialplan ... |
| 14:33.26 | grandpapadot | But I see your point ... |
| 14:35.20 | codefreeze | grandpapadot: for instance, lobby phones could be assigned the "untrusted" context... would be bad if some visitor could program the phone to use "trusted", and then called Taiwan or Bangkok on your dime... |
| 14:36.00 | grandpapadot | Right... hrm... Ok, I'll go to plan "B", thanks! |
| 14:36.54 | boblutz | [TK]D-Fender: hmm.. upgrading 1.4.8 to 1.4.9.2 seems to fix it...`ztmonitor 1 -vv` is showing activity on the card. I will have to confirm this as soon as the line is up |
| 14:37.11 | boblutz | [TK]D-Fender: just in case someone else asks you about a tdm402b not working ^^^ |
| 14:38.56 | codefreeze | grandpapadot: besides, I think "context" is more an asterisk concept, than a sip concept... |
| 14:41.22 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177584185.dsl.bell.ca) |
| 14:41.26 | codefreeze | the lurking experts are more than welcome to shoot me out of my boots... |
| 14:42.08 | *** join/#asterisk jpeeler (n=jpeeler@216.207.245.1) |
| 14:43.29 | sacitec | Hello, i'm have two asterisk servers linked with SIP protocol between them. I am able to call from box 'B' to box 'A', but i'm unable to call from box 'A' to box 'B'. When trying to call, CLI displayz |
| 14:43.36 | *** part/#asterisk sacitec (n=tobi@201.144.211.82) |
| 14:43.51 | *** join/#asterisk seanbright (i=seanbrig@65.207.74.18) |
| 14:44.42 | grandpapadot | codefreeze: right... |
| 14:45.11 | JayTee52 | displayz.........thud |
| 14:45.14 | *** join/#asterisk sacitec (n=tobi@201.144.211.82) |
| 14:45.59 | grandpapadot | leet |
| 14:46.05 | grandpapadot | oops, I mean l33t |
| 14:46.49 | sacitec | Hello, i'm have two asterisk servers linked with SIP protocol between them. I am able to call from box 'B' to box 'A', but i'm unable to call from box 'A' to box 'B'. When trying to call, CLI displays a '484 SIP error'. This one is CLI http://pastebin.com/m567cce45 and this CLI SIP Debug http://pastebin.com/m27559c7e |
| 14:51.03 | *** join/#asterisk CVirus (n=GoD@82.201.178.6) |
| 14:51.40 | *** join/#asterisk delparnel (n=delparne@balance.wlu.ca) |
| 14:51.41 | x86 | sacitec: use IAX between asterisk servers |
| 14:51.45 | x86 | it's much better |
| 14:52.20 | boblutz | Has anyone successfully compiled Asterisk on Vista w/ SP1 ? |
| 14:52.22 | *** join/#asterisk m4sk4r4 (n=m4sk4r4@189.13.189.184) |
| 14:52.49 | codefreeze | sacitec: it's saying "Address incomplete"... don't you need to specify a server? |
| 14:53.08 | [TK]D-Fender | sacitec: Go look at your dialplan and show us the RECEIVING end's debug next time. |
| 14:53.54 | sacitec | x86: i've been using IAX for a while, but there has been modifications on my network and ther is no NAT anymore between servers |
| 15:00.01 | *** join/#asterisk ManxPower (n=manxpowe@65.sub-70-220-244.myvzw.com) |
| 15:00.15 | boblutz | [TK]D-Fender: Yes I can confirm that upgrading 1.4.8 to 1.4.9.2 works for tdm402b, should I post this somewhere? |
| 15:00.55 | *** part/#asterisk trelane (i=trelane@denali.sosdg.org) |
| 15:02.26 | Datax | hi quick question... if I have 2 dial patterns that match a number for an outgoing call which pattern "wins" ? |
| 15:03.02 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-71-126-7-60.bflony.east.verizon.net) |
| 15:03.04 | codefreeze | Datax: the most specific of the two. |
| 15:03.12 | Datax | ok excellent ! |
| 15:03.18 | Datax | asterisk rules :) |
| 15:03.33 | ManxPower | Datax: assuming both patterns are in the same context |
| 15:03.42 | Datax | that's really going to simplify my config :) |
| 15:03.47 | Datax | yeah |
| 15:03.51 | codefreeze | Datax: I hope we are on the same frequency! |
| 15:05.20 | lmadsen | _2XX will match over _XXX if the extension is 205 for example |
| 15:05.23 | codefreeze | Datax: If the num to be matched is 321, pattern 3XX is more specific than XXX |
| 15:05.29 | lmadsen | codefreeze: lol |
| 15:05.29 | ixx | I have 4 zombie sip calls showing under sip show channels, but they do not show when you run core show channels. Is there some way to kill them w/o a restart when they are not showing with other SIP calls under core show channels (along with the other info). |
| 15:05.31 | codefreeze | lmadsen: well said |
| 15:05.38 | lmadsen | codefreeze: I was just thinking the same thing |
| 15:05.56 | Datax | codefreeze: that's what I'd understood :) |
| 15:05.59 | codefreeze | wow, we ARE on the same frequency... lol |
| 15:06.03 | lmadsen | ixx: the zombie channels are probably a transfer or something that are related to a current channel |
| 15:06.04 | Datax | ;) |
| 15:06.16 | lmadsen | ixx: they will go away at some point |
| 15:06.29 | ixx | lmadsen: it has been a few days |
| 15:07.06 | *** join/#asterisk ming_zym (n=ming_zym@220.181.54.107) |
| 15:07.35 | ixx | lmadsen: one side of the bridge dropped off immediately.. the other side is for a system that has restarted since the call hung... |
| 15:07.43 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 15:07.50 | lmadsen | shrugs |
| 15:08.00 | ManxPower | Doesn't Asterisk now have sip timeout and rtp timeout settings? |
| 15:08.05 | *** join/#asterisk PepOSX (n=angeldav@190.78.222.225) |
| 15:08.47 | lmadsen | yep |
| 15:08.54 | lmadsen | rtp at least |
| 15:09.59 | *** join/#asterisk Angel-D (n=angeldav@190.72.156.13) |
| 15:10.14 | ixx | so no way to flush these manually? |
| 15:10.53 | Datax | codefreeze: thanks it works great :) |
| 15:11.37 | Datax | next step is figuring out some kind of backup asterisk in case the main server goes down |
| 15:11.46 | Datax | currently I'm running everything on one machine |
| 15:11.47 | Datax | :s |
| 15:12.12 | Katty | mmmm, my office smells MINTY |
| 15:13.56 | boblutz | Katty: that would be me |
| 15:14.04 | Nasra | I am in the procces of setup my Asterisk pbx ...(small one)....and I might need some guidance later on....if you please....if you are experienced and willing to give a hand....you can pm me....thanks in advace.... |
| 15:14.24 | Katty | boblutz: ^_- |
| 15:14.24 | Katty | boblutz: no. i just cleaned. |
| 15:14.24 | boblutz | :-/ |
| 15:14.25 | Katty | boblutz: this greenworks stuff is DREAMY |
| 15:14.31 | Katty | spritzes boblutz |
| 15:14.33 | Nasra | just voicemail. ivr....etc.... |
| 15:14.48 | Katty | boblutz: now you can be dreamy too! |
| 15:14.52 | boblutz | lol |
| 15:14.57 | Datax | is it possible in some way to have failover when 2 asterisk servers aren't on the same network ? I'm thinking of a situation where an asterisk server would fail but a vpn to a branch office (different network) would still be up. Is there any way of getting the phones to register on the other asterisk server? |
| 15:15.29 | Datax | perhaps that is something I'd have to configure on the phone itself |
| 15:15.41 | Datax | tell it to register with another server is the main one is down |
| 15:15.52 | *** join/#asterisk nny_2 (n=Scott_My@64.203.239.83) |
| 15:16.14 | [TK]D-Fender | Datax: the phone would have to have a failover option (Polycoms do) or you'd have to point them to a more guaranteed service (like SER) |
| 15:16.37 | Datax | ok thanks for the tip :) |
| 15:17.08 | nny_2 | is it possible to register to sip clients under the same extension, say for a laptop client and a deskphone? ( I can just set up an alternate extension and ring them both, more of a curiosity on behavior |
| 15:17.50 | Datax | nny_2: like a shared line ? |
| 15:18.01 | nny_2 | Datax: yes |
| 15:18.20 | Datax | I think I read something about shared lines and asterisk |
| 15:18.26 | [TK]D-Fender | nny_2: You do not register to an "extension", and no, you can't register 2 devices to the same sip.conf account entry. |
| 15:18.27 | Datax | can't remember though |
| 15:18.46 | [TK]D-Fender | nny_2: You'd have to have 2 sip.conf entries, and make your Dial commands call both. |
| 15:18.46 | nny_2 | [TK]D-Fender: roger |
| 15:18.58 | nny_2 | [TK]D-Fender: thats what i figured, just asking silly questions |
| 15:19.06 | [TK]D-Fender | nny_2: No worries |
| 15:19.29 | Datax | [TK]D-Fender: what about real shared lines ? like with cisco callmanager ? |
| 15:19.36 | Datax | can't asterisk handle that ? I thought it could |
| 15:19.50 | *** join/#asterisk ZPertee (n=ZPertee@rrcs-24-106-241-121.central.biz.rr.com) |
| 15:20.01 | [gnubie] | hello all.. anyone cares to check my paste at http://www.privatepaste.com/ffowi2doIY ? |
| 15:20.17 | [TK]D-Fender | Datax: No. |
| 15:20.52 | [gnubie] | you can find my call scenario on that url. i'm having a problem with a sip phone calling a telephone through the pots trunk |
| 15:20.58 | Katty | how do i cp an entire directory? |
| 15:21.05 | boblutz | Katty: cp -r |
| 15:21.06 | [gnubie] | cp -a |
| 15:21.11 | boblutz | o.0 |
| 15:21.14 | Katty | mew? |
| 15:21.25 | boblutz | throws a smoke bomb and vanishes |
| 15:21.29 | *** join/#asterisk [hC] (n=hardcore@190.10.10.76) |
| 15:21.34 | Katty | r for recursive |
| 15:21.54 | Katty | i guess that's it |
| 15:22.22 | [TK]D-Fender | [gnubie]: How do you connect to "pots"? |
| 15:23.06 | [gnubie] | [TK]D-Fender: my asterisk box has a digium dev kit and Zap/4 is connected to the pots network |
| 15:24.46 | [TK]D-Fender | [gnubie]: if LAN is local, check your server's firewall |
| 15:25.09 | [gnubie] | [TK]D-Fender: i even tried disabling the firewall but the problem is still there |
| 15:25.57 | *** join/#asterisk coppice (n=chatzill@175.203.17.210.dyn.pacific.net.hk) |
| 15:27.55 | nny_2 | anyone ever setup a doorbell type thing to work with asterisk? researching now, curious as to how it has been done... |
| 15:28.12 | nny_2 | considering an FXS port an something old school, but a sip intercom doorbell thing would rock |
| 15:28.54 | mvanbaak | tiptel has them I think |
| 15:29.39 | [TK]D-Fender | [gnubie]: pastebin "iptables --list" and show us the output of a defective call (COMPLETE) at verbose 10, SIP DEBUG enabled |
| 15:29.41 | [TK]D-Fender | ~pb |
| 15:29.42 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 15:29.43 | [TK]D-Fender | ^^^^ |
| 15:30.14 | [gnubie] | ok |
| 15:30.19 | [gnubie] | please wait.. |
| 15:30.22 | VJFROMGT | kwik question, if a trunk keeps ringing, does asterisk throw an error code after a while? |
| 15:31.09 | [hC] | VJFROMGT: no, theres no error. |
| 15:31.17 | [TK]D-Fender | VJFROMGT: Only if it thinks theres an error |
| 15:31.52 | *** join/#asterisk delparnel (n=delparne@KTNRON06-1168103470.sdsl.bell.ca) |
| 15:32.02 | *** join/#asterisk steliosk (n=Stelios@athedsl-105864.home.otenet.gr) |
| 15:32.27 | VJFROMGT | what if there the time to ring trunk has reached? |
| 15:33.38 | [TK]D-Fender | VJFROMGT: then * stops dialing |
| 15:34.30 | Katty | anyone have any idea why my server thinks it's december 1969? |
| 15:34.35 | Katty | http server |
| 15:34.50 | Katty | the linux machine itself seems to have the correct date |
| 15:36.45 | anonymouz666 | Katty: your server wanna go to woodstock |
| 15:36.47 | atomspace | I'm trying to setup the same voicemail box (702) over two devices. It seems like its half completed, because I've got the Voicemail message waiting indicator ("VMWI") working on both phones. On the phone using extension 702, it 100% working, but the other phone, using ext. 703, is getting the VMWI for 702's voicemail, but when I dial into the voicemail using the phone on x703 (using # as the extension to login), it says "you have no me |
| 15:36.59 | *** join/#asterisk Bonix-BR (i=Bonix-BR@217-lo1.rt2.isimples.com.br) |
| 15:37.02 | Katty | anonymouz666: :< |
| 15:37.24 | anonymouz666 | he wanna see Jimi Hendrix playing again |
| 15:37.50 | atomspace | Oh, perhaps I can't use # as the "mailbox number" at all. I have to type it in manually? |
| 15:37.52 | boblutz | o shoot! 1337791448@10.69.1.126 |
| 15:37.54 | boblutz | im leet! |
| 15:37.57 | Katty | anonymouz666: maybe you can translate a lil somethin somethin for me. |
| 15:38.30 | Katty | anonymouz666: http://pastebin.ca/955335 |
| 15:38.39 | Katty | anonymouz666: i think it's wanting to write to the directory i chmod 777ed |
| 15:38.47 | Katty | anonymouz666: i think apache is saying NO WAY IN HELL |
| 15:38.50 | Katty | anonymouz666: what do you think? |
| 15:39.22 | Katty | anyone else is welcome to comment (= |
| 15:39.54 | coppice | anyone who could get Hendrix playing again could become very rich |
| 15:40.35 | *** join/#asterisk moa_ (n=moa_@216-129-228-165.vnet-inc.com) |
| 15:41.52 | Warp4 | good morning all |
| 15:41.54 | moa_ | quick question, is it possible to set a globalvar using the manager interface? |
| 15:42.12 | anonymouz666 | Katty: yes, you are probably right. |
| 15:42.15 | Warp4 | once i run asterisk in cli mode and get the "Asterisk Ready." message, is there anything else I need to do to get it to start taking calls? |
| 15:42.53 | *** join/#asterisk ZPertee (n=ZPertee@rrcs-24-106-241-121.central.biz.rr.com) |
| 15:43.00 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 15:43.00 | *** mode/#asterisk [+o russellb] by ChanServ |
| 15:43.21 | *** join/#asterisk ming_zym (n=ming_zym@f0-0-tep-rtr1.corp.cnb.yahoo.com) |
| 15:44.44 | JayTee52 | I am getting intermittent music on hold when I make a test call. I made sure that I have wav, gsm and ulaw files in my moh directory but that doesn't seem to make a difference. The music cuts in an out and it's doing it on two different servers. I'm using Grandstream GXP2000 phones. |
| 15:44.45 | x86 | Warp4: what does your documentation say? |
| 15:45.03 | Katty | anonymouz666: do you happen to know what i should be looking for? |
| 15:46.42 | [gnubie] | [TK]D-Fender: kindly check this link => http://www.privatepaste.com/ed1dhcOa8R |
| 15:48.08 | [TK]D-Fender | Warp4: How about actually building your dialplan and configuring your devices... |
| 15:50.49 | [TK]D-Fender | Warp4: look at 327-330. Doesn't look good. And that is full of excess crap |
| 15:51.11 | *** join/#asterisk lowtone (n=lowtone@ip98-165-136-222.ph.ph.cox.net) |
| 15:51.26 | [TK]D-Fender | Warp4: Your sipphone reg is failing too... all sort of issues with your configs |
| 15:51.43 | Warp4 | ??? |
| 15:52.08 | [TK]D-Fender | Warp4: 454 ,<- |
| 15:52.26 | Warp4 | [TK]D-Fender: um i believe that was supposed to be for someone else |
| 15:53.07 | [TK]D-Fender | [gnubie]: You rather :) |
| 15:53.12 | [TK]D-Fender | Warp4: sorry, bad aim :) |
| 15:53.17 | Warp4 | not a prob |
| 15:53.17 | *** part/#asterisk moa_ (n=moa_@216-129-228-165.vnet-inc.com) |
| 15:53.27 | [TK]D-Fender | Warp4: And I did and your question previously... |
| 15:54.17 | *** join/#asterisk Strom_C (n=strom@208.127.172.112) |
| 15:54.26 | [gnubie] | [TK]D-Fender: please ignore it. extension 341 is not involve in this scenario |
| 15:54.32 | [TK]D-Fender | Katty: Look at the privileges on the files in your folder directly |
| 15:55.11 | [gnubie] | [TK]D-Fender: kindly check the extension 102 for the caller and 994567000 for the callee |
| 15:55.12 | [TK]D-Fender | [gnubie]: also you seem to be masking stuff on 102's invite... |
| 15:55.18 | Katty | [TK]D-Fender: i dunno how :< |
| 15:55.30 | Katty | [TK]D-Fender: been doing ls -l /path/to/file |
| 15:55.39 | [TK]D-Fender | Katty: "ls -l" inside your folder... |
| 15:56.14 | Katty | hmmmm |
| 15:56.34 | [gnubie] | [TK]D-Fender: but there's no error for the 102 == 994567000 call session |
| 15:57.02 | Katty | hmmmm |
| 15:57.09 | Katty | there we go |
| 15:57.12 | [TK]D-Fender | [gnubie]: Where is the problem we are trying to debug? |
| 15:57.22 | Katty | [TK]D-Fender: i didn't chmod it recursively |
| 15:57.43 | *** join/#asterisk PepOSX (n=angeldav@190.72.156.13) |
| 15:57.45 | [TK]D-Fender | Katty: You want 660 for files typically, 770 for folders |
| 15:57.45 | Katty | [TK]D-Fender: now how do i tell apache it's not December 1969? |
| 15:57.53 | [TK]D-Fender | Katty: "date" |
| 15:58.07 | Katty | Mon Mar 24 10:57:58 CDT 2008 :< |
| 15:58.11 | [TK]D-Fender | :/ |
| 15:58.43 | [gnubie] | [TK]D-Fender: the scenario i did was sip hardphone (102) calls a mobile phone through the pots trunk of the asterisk box and the callee cannot hear the voice from the caller but caller can hear the callee's voice.. this is a one-way audio problem |
| 15:59.27 | *** join/#asterisk [hC] (n=hardcore@190.10.10.76) |
| 15:59.56 | [TK]D-Fender | [gnubie]: In this pastebin? |
| 16:00.30 | [gnubie] | [TK]D-Fender: yes, at the http://www.privatepaste.com/ed1dhcOa8R |
| 16:02.28 | [TK]D-Fender | [gnubie]: Looks like the call is coming from a phone behind a remote NAT |
| 16:02.34 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:02.38 | [TK]D-Fender | [gnubie]: For which you did not tell * that it was behind NAT |
| 16:02.56 | [gnubie] | [TK]D-Fender: it's not behind nat |
| 16:03.31 | [gnubie] | [TK]D-Fender: the sip hard phone is connected to the asterisk box through a crossover cable |
| 16:03.34 | [TK]D-Fender | [gnubie]: What is this about then? <--- SIP read from 58.71.7.66:14703 ---> Contact: <sip:102@192.168.11.102:5060> |
| 16:03.56 | [TK]D-Fender | [gnubie]: why is the call coming from a WAN IP? |
| 16:04.17 | [gnubie] | [TK]D-Fender: i don't know. |
| 16:05.05 | ZPertee | how do I interface asterisk with an external loudspeaker paging system? |
| 16:05.28 | [TK]D-Fender | ZPertee: Go find hardware you can connect to * somehow. |
| 16:06.11 | [TK]D-Fender | [gnubie]: Go look at your phone setup then. |
| 16:06.54 | [gnubie] | ok |
| 16:07.28 | [gnubie] | [TK]D-Fender: is it possible that the sip.conf can have 2 localnet at the [global] context? |
| 16:07.48 | ZPertee | [TK]D-Fender, would it be better to connect via a TDM card or an 1/8" plug to my soundcard? |
| 16:08.16 | [TK]D-Fender | [gnubie]: Yes |
| 16:09.00 | [TK]D-Fender | ZPertee: depends. I prefer not having to depend on having a soundcard or risking wiring something physically direct to my server (more than ethernet) |
| 16:09.52 | [gnubie] | [TK]D-Fender: how do i do that? separated by comma or space, or shall i have to lines for localnet ? |
| 16:11.33 | [TK]D-Fender | [gnubie]: Multiple lines |
| 16:12.11 | [gnubie] | ok |
| 16:13.33 | *** join/#asterisk [hC] (n=hardcore@190.10.10.76) |
| 16:18.21 | *** join/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net) |
| 16:23.58 | ZPertee | [TK]D-Fender, over the weekend I had asked you about picking up a zap line and flashing the line in order to connect to my paging system. what is the reason that I can't "pickup", "flash", "sendDTMF(70)" "say what I want"? Trying to troubleshoot the problem and don't understand why the Flash() app won't do the trick? |
| 16:24.10 | [gnubie] | hhmmmm.. maybe this phone is buggy |
| 16:24.11 | [gnubie] | :( |
| 16:24.30 | [TK]D-Fender | ZPertee: Just go try stuff. |
| 16:25.00 | ZPertee | [TK]D-Fender, isn't that what the Flash() cmd is for? how else would I use it???? |
| 16:25.10 | [TK]D-Fender | [gnubie]: Its pointing to the WAN IP go check the phone itself |
| 16:25.15 | atomspace | Katty: earlier we discussed setting up the same voicemailbox for two extensions. My configuration is using two extensions 702 and 703. I want both of these to use mailbox # 702. Calling from extension 702, I can hit # to supply the mailbox number at login, but calling from 703 seems to be inputting "703" when I hit #, even though in users.conf under 703's context, I've got "mailbox=702". Is it possible that hitting # to login is impossi |
| 16:26.14 | *** join/#asterisk VaNNi (n=VaNNi@lgb-static-216.70.165.200.mpowercom.net) |
| 16:26.17 | atomspace | I'll add that the VMWI is working for the device on extension 703 |
| 16:26.27 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 16:26.31 | atomspace | ...indicating that 702 has a voicemail |
| 16:26.52 | *** join/#asterisk Chris-NB (n=chris@213162066131.public.t-mobile.at) |
| 16:26.54 | [gnubie] | [TK]D-Fender: as far as i understand the web ui config of this unknown ip phone, it doesn't say for any wan ip configuration especially the 58.71.7.66 ip address.. |
| 16:29.21 | [gnubie] | [TK]D-Fender: this sip hard phone was given to me by a friend |
| 16:29.37 | [gnubie] | [TK]D-Fender: it doesn't have any clue as to what brand and model it is |
| 16:30.45 | [gnubie] | [TK]D-Fender: i asked google and it says it's similar to the YVH1000 of this document => http://skyfon.skynet.cz/docs/yvkx000_usermanual.pdf |
| 16:35.53 | *** join/#asterisk AstPT (n=wewe@213.63.128.1) |
| 16:36.00 | AstPT | hi all |
| 16:36.03 | *** join/#asterisk hohum (n=dcorbe@70.0.187.213) |
| 16:36.04 | *** join/#asterisk ber___ (i=brad@neu.cow.org) |
| 16:36.09 | AstPT | i have a question |
| 16:36.17 | boblutz | AstPT: Oh yea? |
| 16:36.19 | AstPT | can someone help me? |
| 16:36.21 | AstPT | hi |
| 16:36.26 | boblutz | ~ask |
| 16:36.27 | jbot | 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. |
| 16:36.28 | russellb | ~justask |
| 16:36.43 | AstPT | Call failed to go through, reason X |
| 16:36.44 | ZPertee | anyway to get around teliax blocking 800 caller id? |
| 16:37.05 | lmadsen | you could talk to teliax about it |
| 16:37.12 | AstPT | where can i find the documentation to know this reasons |
| 16:37.23 | lmadsen | I don't know how you expect to control CID that an upstream provider is blocking |
| 16:37.33 | *** join/#asterisk BBHoss (n=hoss@c-71-207-220-138.hsd1.al.comcast.net) |
| 16:37.56 | *** join/#asterisk delparnel (n=delparne@KTNRON06-1168103470.sdsl.bell.ca) |
| 16:38.35 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 16:38.41 | AstPT | where can i find the documentation to know the reasons for : (NOTICE[4758]: Call failed to go through, reason 3) |
| 16:38.41 | ZPertee | teliax won't do anything about it they won't allow it, effort to cut costs. seems pretty stupid to have 800 and not be able to have caller id |
| 16:40.36 | AstPT | anyone knows what do these reasons mean |
| 16:40.37 | AstPT | ? |
| 16:41.23 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 16:41.51 | *** part/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 16:44.17 | *** join/#asterisk bsaxon (n=bsaxon@66.0.66.4) |
| 16:44.39 | *** part/#asterisk iamhrh (n=iamhrh@office.amsvans.com) |
| 16:44.54 | AstPT | Can anyone tell me what are the meaning of the reasons for a failed call to go through (NOTICE[3799]: Call failed to go through, reason 3) |
| 16:45.33 | ber___ | maybe it is these: |
| 16:45.36 | ber___ | http://www.voip-info.org/wiki/view/Asterisk+variable+hangupcause |
| 16:45.41 | AstPT | thks |
| 16:46.54 | AstPT | i will take a look on it |
| 16:49.23 | *** join/#asterisk stimpie (n=stimpie@84-104-5-115.cable.quicknet.nl) |
| 16:50.47 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 16:51.26 | lesouvage | I have this message in the cli every one second: "WARNING[135]: chan_sip.c:1775 __sip_xmit: sip_xmit of 0x831428 (len 532) to 192.168.1.117:5060 returned -1: Host is down" Any suggestion for the reason or the cure? |
| 16:51.55 | tzanger | god damn I love my asterisk |
| 16:52.07 | tzanger | callerid is blocked, empty or invalid? straight to voicemail |
| 16:52.07 | *** join/#asterisk hardwire (n=bip@xvm-189-205.ghst.net) |
| 16:52.12 | hardwire | it is i.. hardwire. |
| 16:52.19 | hardwire | waits for the smoke to clear. |
| 16:52.29 | tzanger | call from someone for me or my wife and we're not home? it calls the right cell phone |
| 16:53.08 | lmadsen | tzanger: yep.... asterisk is pretty handy :) |
| 16:53.13 | hardwire | so.. fax detect via exten => fax,n,.... |
| 16:53.16 | hardwire | how does that work for pots? |
| 16:53.20 | hardwire | whats does it detect. |
| 16:53.24 | lmadsen | the fax tone |
| 16:53.25 | tzanger | I'm on the road? I can call or take calls from the laptop |
| 16:53.30 | tzanger | god damn asterisk is awesome |
| 16:54.02 | lmadsen | tzanger: and my cell phone has wifi in it, so I can get asterisk to call my cell phone if ChanIsAvail() via wifi instead of via the PSTN |
| 16:54.21 | lmadsen | it's the new hawtness |
| 16:54.22 | hardwire | lmadsen: so the fax tone is recieved before call routing is started? |
| 16:54.25 | lmadsen | I think this asterisk think might catch on |
| 16:54.31 | lmadsen | hardwire: something like that |
| 16:54.38 | lmadsen | I haven't done faxing |
| 16:54.52 | lmadsen | I think I answer via 's' first |
| 16:54.58 | hardwire | I'm hoping to use our dedicated fax lines as inbound via a hunt group if our PRI is down |
| 16:55.04 | lmadsen | then it jumps to fax when it gets the tone... or something |
| 16:55.08 | lmadsen | you could always just try it :) |
| 16:55.12 | hardwire | lmadsen: hmm.. I'll have to play |
| 16:55.19 | hardwire | lmadsen: its easy to try on test gear |
| 16:55.21 | hardwire | I don't have any |
| 16:55.24 | hardwire | I have live gear |
| 16:55.25 | hardwire | :) |
| 16:57.14 | *** join/#asterisk adjohn (n=adjohn@i222-151-36-16.s05.a013.ap.plala.or.jp) |
| 16:58.12 | [hC] | This is kinda strange. In an asterisk installation I have in costa rica, asterisk recognizes DTMF keypresses on an IVR, but as soon as i call DISA() the tones arent picked up any longer. |
| 16:58.23 | [hC] | maybe i just need to upgrade. |
| 16:59.20 | tzanger | asterisk decouples my phones from the network |
| 16:59.31 | tzanger | mythtv decouples my media from the network |
| 16:59.47 | tzanger | what else... x10 I guess but there's something better for home control |
| 17:00.57 | lmadsen | hardwire: heh... time to get some test gear. Way too many people "play" on their live equipment |
| 17:00.57 | hardwire | robots. |
| 17:01.07 | hardwire | robots are awesome for home control. |
| 17:01.17 | hardwire | lmadsen: easier said than done :) |
| 17:01.31 | Katty | [TK]D-Fender: http://pastebin.ca/955438 :< |
| 17:01.35 | tzanger | lmadsen: that's where you get the best feedback! |
| 17:02.01 | lmadsen | not really... I'm an independent consultant, and I have 1 laptop that I work on, and 1 server with VMware that I do all my testing on |
| 17:02.15 | lmadsen | there is never an excuse not to have a test rig |
| 17:02.41 | lmadsen | working on a production server is just stupid |
| 17:02.55 | lmadsen | but if that's how people think it should be done, then who am I to question them |
| 17:03.02 | lmadsen | I've gotten work because of stuff like that :) |
| 17:04.07 | *** join/#asterisk Poizonb0x (n=assdasd@ip247-10.ct.co.cr) |
| 17:04.17 | MihiNomenEst | the only time I "test" in the production environment is when I demo features for the office manager. |
| 17:05.07 | *** join/#asterisk ToTo (n=ToTo@host206-211-dynamic.2-87-r.retail.telecomitalia.it) |
| 17:05.41 | *** join/#asterisk ToTo (n=ToTo@host206-211-dynamic.2-87-r.retail.telecomitalia.it) |
| 17:06.56 | [TK]D-Fender | Katty: date is clearly getting passed to the page via GET |
| 17:07.21 | Katty | [TK]D-Fender: okay. |
| 17:07.40 | Katty | [TK]D-Fender: should i be getting the date some other way? |
| 17:08.02 | *** join/#asterisk Olobola (i=Olobola@15.sub-75-211-66.myvzw.com) |
| 17:08.03 | [TK]D-Fender | Katty: that isn't your code is it? |
| 17:09.23 | Katty | [TK]D-Fender: no |
| 17:09.27 | Katty | [TK]D-Fender: i odn't know a whole lot of php |
| 17:09.29 | tzanger | Katty: you have a server named dustbunny? haha |
| 17:09.34 | Katty | tzanger: yes (= |
| 17:09.37 | [TK]D-Fender | Katty: Go look at the config of the doc calling it |
| 17:09.48 | Katty | i did |
| 17:09.51 | Katty | that's the snippet at the top |
| 17:09.56 | Katty | the index.php is posting it |
| 17:10.16 | [TK]D-Fender | Katty: The date is a programatic choice somewhere I might think... |
| 17:11.30 | *** join/#asterisk delparnel (n=delparne@CPE001839f2889c-CM00137185dfb6.cpe.net.cable.rogers.com) |
| 17:12.58 | Katty | [TK]D-Fender: erm k |
| 17:14.35 | *** join/#asterisk Juggie (i=Juggie@CPE001601df17fb-CM001a6682d7b6.cpe.net.cable.rogers.com) |
| 17:18.31 | Olobola | does anyone know of an open source sip client I can drop into a vb6 app? |
| 17:20.20 | [gnubie] | waves to all.. |
| 17:20.23 | [gnubie] | gtg now.. |
| 17:20.36 | [gnubie] | thanks [TK]D-Fender for helping me find the problem.. |
| 17:20.45 | *** part/#asterisk [gnubie] (n=[gnubie]@cm32.omega116.maxonline.com.sg) |
| 17:20.47 | [TK]D-Fender | [gnubie]: You're welcome |
| 17:25.21 | *** part/#asterisk annielou (n=anne@c-76-119-139-36.hsd1.ma.comcast.net) |
| 17:29.22 | *** join/#asterisk pa (n=paolo@unaffiliated/pa) |
| 17:29.28 | *** join/#asterisk rpminihan (n=ryanpm@66-241-70-231.reverse.projecta.com) |
| 17:30.21 | rpminihan | Hey guys. Can someone tell me what would cause this error: "chan_sip.c:14758 handle_request_register: Registration from '<sip:170@192.168.1.131>' failed for '192.168.1.170' - Device does not match ACL" |
| 17:35.26 | AstPT | rpminihan: I think is the allow and deny from |
| 17:35.47 | AstPT | in sip.conf or users.conf |
| 17:35.58 | rpminihan | Ok. I don't think I have one set. Is there a default? |
| 17:37.51 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 17:38.14 | [TK]D-Fender | rpminihan: Stop thinking, start looking... |
| 17:39.25 | rpminihan | I just looked again. None set. I've added this line: permit=192.168.1.0/255.255.255.0 to my sip.conf file and issued a sip reload. |
| 17:39.27 | rpminihan | No change. |
| 17:41.25 | [TK]D-Fender | rpminihan: Perhaps you should show us your full setup, and the full CLI output w/ sip debug of your attempt.... |
| 17:43.46 | *** join/#asterisk tripps (n=sean@72.20.150.196) |
| 17:45.03 | Rico29 | somebody who knows OPAL ? (http://www.opalvoip.org/) |
| 17:51.24 | *** join/#asterisk Rico29 (n=Rico@ARennes-358-1-60-3.w90-49.abo.wanadoo.fr) |
| 17:55.35 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
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| 18:01.53 | *** join/#asterisk NinjaJon (n=Administ@liv-lxp-001.northenden.ninja.org.uk) |
| 18:02.28 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 18:06.48 | rpminihan | I figured it out. Wasn't specifically an asterisk problem. |
| 18:07.15 | rpminihan | Misconfiguration in my phone configs. |
| 18:07.58 | NinjaJon | Hi guys. I have just got a T1 card and Zhone channel bank from eBay, but am struggling to configure both ends successfully. I have got as far as bringing up the link between the two (both ends show OK) but Asterisk currently reports the PRI span is provisioned & down. This is my first time with T1/E1 and I'm not sure where to go next.. Any tips? |
| 18:08.21 | NinjaJon | (zttool reports the T1 card is in state OK) |
| 18:09.04 | [TK]D-Fender | NinjaJon: You don't use PRI signalling with channel banks normally. |
| 18:09.20 | [TK]D-Fender | NinjaJon: You'd use CAS. pastebin your zaptel.conf & zapata.conf |
| 18:09.26 | *** join/#asterisk wordzilla (n=me@d58-106-139-71.sbr4.nsw.optusnet.com.au) |
| 18:10.04 | [TK]D-Fender | ~pb |
| 18:10.05 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 18:10.06 | [TK]D-Fender | ^^^^^^^ |
| 18:11.22 | NinjaJon | D-Fender: Ahhh, OK, that would be a good starting point! Thanks.. Pastebin is here: http://pastebin.ca/955513 |
| 18:12.38 | MooingLemur | anyone ever try to get linksys/sipura phones to provision over HTTPS? (the web server gives a message about an unrecognized CA, and I'm not sure whether the web server is complaining about the authenticity of the client cert (which it shouldn't do based on config), or the phone doesn't recognize the server CA and is cutting off the connection. |
| 18:13.26 | NinjaJon | D-Fender: My current config is probably a mismash of the info I have found via google; as you can see there are a few commented out lines & I have tried a number of different methods.. |
| 18:15.08 | boblutz | NinjaJon: Wilson? |
| 18:15.26 | NinjaJon | MooingLemur: If your web server is producing the error message (is it apache? in error.log?) then I would think it will be complaining about the client cert. Surely if the phone is not recognising the server CA, it wouldn't log this to the server but via syslog/wherever. What method have you used to ignore the client cert? |
| 18:15.36 | [TK]D-Fender | NinjaJon: http://pastebin.ca/955519 |
| 18:15.38 | NinjaJon | boblutz: nope? sorry |
| 18:16.10 | NinjaJon | D-Fender: thanks, will try this now! |
| 18:18.33 | MooingLemur | NinjaJon: apache error is [Mon Mar 24 11:09:00 2008] [info] SSL Library Error: 336151576 error:14094418:SSL routines:SSL3_READ_BYTES:tlsv1 alert unknown ca |
| 18:18.33 | MooingLemur | NinjaJon: phone syslog output is http://pastebin.ca/955523 |
| 18:19.41 | *** join/#asterisk vicram (n=vr@cm136.delta11.maxonline.com.sg) |
| 18:20.46 | NinjaJon | D-Fender: Thanks for the config - it works OK in so far as Asterisk loads without errors; "zap show channels" shows all the channels correctly. No dialtone on my test telephone handset though; is there any debugging I can run on the Asterisk/zap side to track down whether the problem is on the channel bank or within Asterisk? I had previously gone down the "pri debug span 1" route, but if as you say I shouldn't be running PRI... |
| 18:21.23 | MooingLemur | NinjaJon: the config option is in the SSL vserver: SSLVerifyClient optional_no_ca |
| 18:23.33 | NinjaJon | MooingLemur: It does look like the web server is refusing the connection based on it not recognising the client CA. Have you tried "SSLVerifyClient none" ? I must admit that my SPAs use straight HTTP (no HTTPS yet...) |
| 18:24.30 | glaz | why does it keep asking for my mailbox number when I dial the Voicemail extension |
| 18:24.51 | glaz | I have extension *98 configure as this: exten => *98,2,VoiceMailMain(s${VM_CALLERID}) |
| 18:25.03 | [TK]D-Fender | glaz: Because you are not telling it which box to use |
| 18:25.17 | [TK]D-Fender | glaz: And where is ${VM_CALLERID} set? |
| 18:25.29 | MooingLemur | NinjaJon: as far as I know, it's the default. However, since SSLVerify client also has "optional", and "optional_no_ca", I would assume "none" would bail out if it sees a client cert, as odd as that is. |
| 18:25.44 | glaz | I tought VM_CALLERID was a pre-define variable |
| 18:27.05 | NinjaJon | MooingLemur: Not 100% sure of the process flow behind SSLVerifyClient; but I have found another post with a similar Apache error message, it does look like the server doesn't like the client cert: |
| 18:27.05 | NinjaJon | http://www.omniorb-support.com/pipermail/omniorb-list/2004-November/026154.html |
| 18:27.42 | NinjaJon | MooingLemur: we just need to find the public CA for linksys/SPA.. |
| 18:27.46 | *** join/#asterisk r0d3nt (n=astrutt@backup.realprohosting.com) |
| 18:27.57 | glaz | [TK]D-Fender: I have VM_CALLERID in voicemail.conf too |
| 18:29.41 | [TK]D-Fender | glaz: Doesn't work like you think it does. |
| 18:30.13 | [TK]D-Fender | glaz: Go use a real var. |
| 18:30.21 | [TK]D-Fender | glaz: or real CALLERID |
| 18:30.23 | NinjaJon | D-Fender: Do you know what I should be looking for in terms of config on the channel bank? Will I need to configure the line type (currently esf) and line code (currently b8zs) on the DS1 (T1) line - or are these not relevant in my case (no PRI signalling)? "ds1-mode" is currently set to "csu". Sorry for the newbie questions..! |
| 18:31.54 | [TK]D-Fender | NinjaJon: Check between CSU/DSU, and don't forget you'll have to stop * and redo "ztcfg -vvvv" and restart * |
| 18:33.16 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 18:33.39 | glaz | [TK]D-Fender: on an other setup we have here, and it works, I have this: exten => *98,1,VoiceMailMain(${CALLERIDNUM}) |
| 18:33.46 | glaz | same version everything |
| 18:41.37 | NinjaJon | D-Fender: thanks, I have restarted * and re-run ztcfg. Still not sure if I am looking at the right thing as far as my channel bank is concerned: http://pastebin.ca/955554 |
| 18:41.38 | *** join/#asterisk l2trace99 (n=asd@static-71-251-65-16.tampfl.fios.verizon.net) |
| 18:43.35 | l2trace99 | has anyone delt with DTMF issues with * 1.4 ? |
| 18:43.36 | *** join/#asterisk shido6 (n=shido6@204.126.120.132) |
| 18:44.50 | NinjaJon | l2trace99: what sort of DTMF issues are you having? SIP/ZAP? Detection / passthrough? |
| 18:45.26 | Datax | Hi again, I'm working at trying to get my moh to stream off my local icecast relay what is the best way of getting it to work because whatever I try it won't work. I followed this without success : http://nerdvittles.com/index.php?p=92 |
| 18:45.31 | MooingLemur | NinjaJon: thanks.. I've considered that :) but it seemed easier to make apache ignore it.. I'm not sure if it's not ignoring the client cert, or if something else is going on. |
| 18:46.05 | l2trace99 | SIP detection for meetme |
| 18:46.48 | l2trace99 | i have 2 * boxen ( 1.2 and 1.4 ) hanging off a CM cluster |
| 18:47.06 | l2trace99 | 1.2 works for the most part |
| 18:47.22 | l2trace99 | 1.4 has alot if issues |
| 18:48.00 | l2trace99 | debug rtp doesn't see the dtmf when it fails |
| 18:49.02 | [TK]D-Fender | NinjaJon: I think the "ks" should be "ls" (loop-start). |
| 18:49.13 | *** join/#asterisk droops (n=droops@74.193.237.138) |
| 18:49.33 | [TK]D-Fender | l2trace99: What mode? What exact version? |
| 18:50.17 | [TK]D-Fender | glaz: well that 2nd one is at least valid for 1.2-. It was deprecated in 1.2, but at least still present. |
| 18:50.29 | [TK]D-Fender | glaz: the CALLERID function replaced it in 1.2+ |
| 18:51.55 | [TK]D-Fender | NinjaJon: I'm sorry I think you did need that span line back... : span=1,0,0,b8zs,esf |
| 18:52.13 | [TK]D-Fender | NinjaJon: note : 1,0,0 <- * provides timing. |
| 18:52.36 | *** join/#asterisk tripps (n=sean@72.20.150.196) |
| 18:52.37 | glaz | [TK]D-Fender: thanks |
| 18:52.42 | [TK]D-Fender | NinjaJon: And you are using a cross-over T! cable, right? |
| 18:52.45 | [TK]D-Fender | T1* |
| 18:52.57 | l2trace99 | [TK]D-Fender: * 1.4.16 rfc2833 |
| 18:53.20 | l2trace99 | i tried with other modes also |
| 18:53.20 | [TK]D-Fender | l2trace99: And that matches your expectation for CCM? |
| 18:53.44 | glaz | [TK]D-Fender: even with CALLERID I get : [Mar 24 14:52:11] WARNING[25369]: app_voicemail.c:6304 vm_authenticate: Couldn't read username |
| 18:53.54 | *** join/#asterisk tainted_ (n=tainted@adsl-75-34-145-25.dsl.irvnca.sbcglobal.net) |
| 18:53.59 | NinjaJon | D-Fender: Thanks.. Yup, I found out the 'hard way' that I needed a crossover cable :-) Have now manufactured one and both ends are happy. I've now added the span line back & will try restarting asterisk/zaptel |
| 18:54.26 | *** join/#asterisk BCS-Satori (n=somewher@75.148.21.113) |
| 18:54.35 | l2trace99 | there isn't a way that I have found to specify in CM. But it works with * 1.2.5 and with 1.4.16 coming from local phones |
| 18:55.22 | BCS-Satori | I am trying to register vonage business with asterisk, with the following command in the sip.conf register => 14105551234:password@sphone1.vonage.net:5061/DID_trunk_1 currently when i reload asterisk i see Format for authentication entry is user[:secret]@realm at line 53 |
| 18:56.48 | *** join/#asterisk generalhan (n=asd@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 18:56.50 | l2trace99 | also tried with relaxdtmf |
| 18:56.54 | generalhan | hey all ! |
| 18:57.53 | generalhan | i just switch my asterisk boxes out, and now they have a new IP ... but for some reason, some of the remote phones are still hitting the old server... even though all my port forwarding is set to the new address |
| 18:57.59 | generalhan | anyone ever seen something like this before ? |
| 18:58.33 | NinjaJon | generalhan: are you behind a NATting firewall/proxy? Could it maintain a state table and still be sending packets relating to previously established sessions to the old destination? |
| 18:59.33 | generalhan | NinjaJon: hmm possibly, i use an IPCop box ... but some of my remote users are working just fine, with no other config changes other than changing the port forwarding on port 5060 to the new IP |
| 19:00.45 | NinjaJon | D-Fender: Many thanks for your help! I now have dialtone on the first port on my channel bank. All I need to do now is configure the other 23 ports to match... :) Much appreciated. |
| 19:00.52 | Datax | are there any known problems between asterisk 1.4.18 and mpg123 v 0.61 ? |
| 19:04.29 | NinjaJon | generalhan: can you check /proc/net/ip_conntrack for any established sessions relating to the broken clients? |
| 19:06.09 | generalhan | NinjaJon: no, no reference to that remote user's ip |
| 19:07.16 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 19:07.35 | BCS-Satori | when programing auth=md5 for a trunk what is the correct register syntax for the sip.conf |
| 19:08.42 | *** join/#asterisk [intra]lanman (n=lanman@va-76-6-212-48.dhcp.embarqhsd.net) |
| 19:12.45 | *** join/#asterisk CrashHD (n=CrashHD@76-14-102-131.rk.wavecable.com) |
| 19:15.05 | *** join/#asterisk sosoriosv (n=salvador@200.31.160.7) |
| 19:15.19 | *** part/#asterisk sosoriosv (n=salvador@200.31.160.7) |
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| 19:21.26 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 19:22.47 | BCS-Satori | hmmm does anyone know if it is possible to mask callerid(num) on vonage? i can mask the name but cant seem to get the number |
| 19:23.05 | ZPertee | is the best way to get asterisk to pay file over a soundcard to use System app? |
| 19:24.50 | *** join/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net) |
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| 19:29.53 | supjigator | BCS-Satori: Huh? |
| 19:30.02 | supjigator | BCS-Satori: Outbound masking? |
| 19:30.20 | ZPertee | BCS-Satori, I would say not, although haven't tried it to be sure |
| 19:30.56 | supjigator | You can't mask either on a outbound. |
| 19:31.08 | supjigator | You can do whatever you want on inbound. |
| 19:31.18 | weazahl | im having some serious problems with my sangoma a102d. i have connected the spans together with a t1 crossover, used a loopback plug and i have only had a green condition once. i unplugged the xover and plugged it back in, RED. have not got green again. anyone have ideas on what to do next? |
| 19:31.28 | atomspace | I'm attemping to setup up the same voicemailbox for two extensions. My configuration is using two extensions 702 and 703. I want both of these to use mailbox # 702. Calling from extension 702, I can hit # to supply the mailbox number at login, but calling from 703 seems to be inputting "703" when I hit #, even though in users.conf under 703's context, I've got "mailbox=702". Is it possible that hitting # to login is impossible for 703 ? |
| 19:32.00 | *** join/#asterisk Greek-Boy (n=email@41.221.58.13) |
| 19:33.51 | BCS-Satori | when using an md5 auth on a trunk, how do you use the register command in the sip.cfg i am currently recieving WARNING[2780]: chan_sip.c:16192 add_realm_authentication: Format for authentication entry is user[:secret]@realm at line 54 |
| 19:37.50 | [TK]D-Fender | NinjaJon: You're welcome, though you DID have 24 all functioanl from your config IIRC. |
| 19:38.17 | [TK]D-Fender | glaz: and I DID tell you that ${calleridnum} was removed in 1.4 |
| 19:38.29 | [TK]D-Fender | glaz: please use the CALLERID function. |
| 19:39.24 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 19:40.43 | *** join/#asterisk kannan (n=kannan@121.241.156.83) |
| 19:43.41 | kannan | hello all, I have a Digum 120P E-1 card, with which I want to connect to a Windows 2003 server with a Dialogic E-1 card, to route all cal originating from the dialogic thru a SIP service provider. I am not able to get OK zap status, the alrm is red(patlooptest is fine), I have set the signalling as pri_net , switchtype as euroisdn. When i see the sapn details in the cli, it is showing that * is on CPE side and not the network side |
| 19:45.03 | [TK]D-Fender | kannan: Hopefull you set the jumper on the TE120 to E1.... |
| 19:45.11 | flush | yo |
| 19:45.19 | [TK]D-Fender | kannan: And it'd be a good idea for you to pastebin your zaptel.conf & zapata.conf... |
| 19:45.27 | [TK]D-Fender | ~pb |
| 19:45.27 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 19:45.29 | flush | i sometimes get home and i ahve like 2 blank mesasges on the voicemail |
| 19:45.39 | flush | what does it mean.. its like totally on mute and asterisk says its a message |
| 19:45.47 | generalhan | has anyone had any issues with MixMonitor deciding to just stop recording in 1.4.18 ... all calls stop after they reach the same size of 2,284 bytes |
| 19:45.51 | flush | why this is happening? how can i make sure i wont have any more "empty" voicemail messages again |
| 19:46.03 | kannan | ~pb |
| 19:46.03 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 19:46.06 | [TK]D-Fender | flush: taht either inbound sound is not working, or they hung up instead of completing a real message |
| 19:46.32 | flush | and asterisk continues recording even when they hang up before the voicemail starts recording ? |
| 19:47.20 | kannan | [TK]D-Fender , Is it compulsory to use PRI only with a telco, (whicjh is working fine), or can we use it internally, between 2 own devices too? |
| 19:47.29 | [TK]D-Fender | flush: it stops once * has detected the hangup (an analog issue) |
| 19:47.48 | flush | i see |
| 19:48.00 | [TK]D-Fender | kannan: you can use whatever signally you want based on the capabilities of the other side. |
| 19:48.40 | *** join/#asterisk Yourname` (n=chatzill@unaffiliated/yourname/x-837320) |
| 19:49.06 | Yourname` | [Mar 24 15:46:41] NOTICE[29359]: rtp.c:1287 ast_rtp_read: Unknown RTP codec 126 received from '74.xx.xx194' -> There's 5-10 SIP peers behind this IP. How can I find out who is the rogue packet sender her? |
| 19:49.08 | Yourname` | here* |
| 19:51.55 | *** join/#asterisk KaiK (n=KaiK@dslb-088-077-198-166.pools.arcor-ip.net) |
| 19:52.31 | kannan | [TK]D-Fender > thanks , i have pasted at http://pastebin.ca/955657 |
| 19:53.05 | KaiK | hey guys, i am searching for a way to output (ie in CLI) the number of current active phone-calls/lines... but i dont find a way... |
| 19:53.43 | weazahl | ok, i can loop my T1 spans untill zaptel loads. why does zaptel break my link? signaling? also should echo cancel be off in zapata.conf since it has hardware ec? |
| 19:53.53 | [TK]D-Fender | kannan: Looks legit for E1, if zaptel comes up, make sure you are using a proper E1 cross-over cable |
| 19:54.06 | generalhan | KaiK: sip show channels, iax2 show channels ? |
| 19:54.12 | [TK]D-Fender | KaiK: "core show channels" |
| 19:54.14 | generalhan | ect. |
| 19:55.54 | boblutz | ~background |
| 19:55.55 | jbot | rumour has it, background is shell job control lets you start processes backgrounded. "command &". To run a command and exit a session: "command & exit;" |
| 19:56.02 | KaiK | cool thx |
| 19:56.05 | CCFL_Man2 | i need an 8th of an inch spacer so this rack tail will fit |
| 19:56.25 | CCFL_Man2 | tail = rail |
| 19:57.20 | CCFL_Man2 | i hate to use my ruler stick |
| 19:57.20 | *** join/#asterisk slima (i=slima@unaffiliated/slima) |
| 19:57.33 | kannan | [TK]D-Fender > thanks , i have run patlooptest , it rns fine, but the card is now coonectd to a dialigic card in a Windows server, the alrm is red |
| 19:57.46 | kannan | also, ri show shows as cpe , not net |
| 19:57.59 | [TK]D-Fender | kannan: also make sure of who is supposed to be running timing.... |
| 19:58.04 | kannan | on reload zap , it says ignoring signalling |
| 19:58.09 | [TK]D-Fender | kannan: And who should be net/cpe |
| 19:58.13 | kannan | ohh ok |
| 19:58.28 | kannan | both the achines are to be configured by me only |
| 19:58.49 | kannan | the dialog is on a call center server to make outbond cals, |
| 19:59.00 | kannan | asterisk has to route it to a SIP termination rovder |
| 19:59.05 | kannan | provider |
| 20:01.02 | kannan | i will check and get back tomorrow |
| 20:01.02 | tripps | anyone have difficulty in general in successfully receiving a fax from an ip fax service/system when running nvfax? i get analog faxes perfectly, but faxes from other IP faxes don't seem to work very well |
| 20:01.02 | glaz | [TK]D-Fender: even with exten => *98,1,VoiceMailMain(CALLERID) it does not work, I tried ${CALLERID} also, same thing. |
| 20:01.05 | kannan | tansk a lot |
| 20:01.11 | kannan | exit |
| 20:01.14 | *** part/#asterisk kannan (n=kannan@121.241.156.83) |
| 20:01.58 | *** join/#asterisk edwin_quijada (n=m@25.116.88.200.m.sta.codetel.net.do) |
| 20:02.00 | edwin_quijada | Hi! |
| 20:02.13 | x86 | glaz: ${CALLERID(num)} |
| 20:02.16 | *** join/#asterisk slima (i=slima@unaffiliated/slima) |
| 20:02.34 | edwin_quijada | I am making a IVR with 24 lines, T1, and I 'd like to know what is the best hardware for this |
| 20:03.04 | edwin_quijada | I think in a DualCOre machine, 2GBRam and 200Gb disk HD |
| 20:03.09 | [TK]D-Fender | glaz: I said use the CALLERID function. it should be referenced as ${CALLERID(number)} . Since you did not know this you should go read over 1.4's dialplan basics again |
| 20:03.22 | *** join/#asterisk tobias (n=tobias@user-0ce2hpk.cable.mindspring.com) |
| 20:03.55 | [TK]D-Fender | edwin_quijada: By itself very little. You'll have to be more specific about what else * will be doing. You could get away with a P3 for this so far... |
| 20:03.58 | glaz | [TK]D-Fender: Yeah, I am new to 1.4. I will do. Also going to Asterisk bootcamps on may 5th-9th heh. |
| 20:04.25 | glaz | but ${CALLERID(num)} works, thanks. |
| 20:05.31 | [intra]lanman | everyone's probably already seen this... but i feel compelled to post it anyway http://digg.com/software/FreeSWITCH_1_0_RC1_Released/ |
| 20:05.46 | *** join/#asterisk [hC] (n=hardcore@190.10.10.76) |
| 20:06.15 | Qwell | "it took me 2 years" |
| 20:06.20 | Qwell | O.o |
| 20:07.07 | edwin_quijada | [TK]D-Fender: Just that! I need to do a IVRS for payments and this T1 is just for that |
| 20:07.17 | Nasra | hi, what do I use to open file with .exe in ubuntu? |
| 20:07.25 | edwin_quijada | No calls, nothing else for that T1 |
| 20:07.29 | Nasra | wrong channel |
| 20:07.54 | [TK]D-Fender | edwin_quijada: Well you need to be specific about what processing this box will do? intense DB? Transcoding? Only internal processing? etc. |
| 20:08.05 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 20:08.06 | [TK]D-Fender | Nasra: vi :) |
| 20:08.28 | boblutz | Nasra: DONT DO IT! |
| 20:09.50 | edwin_quijada | [TK]D-Fender: Seek in database, using Festival to TTS |
| 20:10.11 | boblutz | edwin_quijada: |
| 20:10.46 | *** join/#asterisk jmesquita (n=jmesquit@201.7.117.114) |
| 20:11.09 | [TK]D-Fender | edwin_quijada: Get a pretty normal P4 + 1GIG ought to be fine |
| 20:11.42 | boblutz | edwin_quijada: sorry - I am pushing a Xeon + 2 gigs o' ram - X11 and it runs great! |
| 20:11.55 | Qwell | X11 on a server is...kinda dumb |
| 20:11.58 | boblutz | (overkill in my opinion...especially for < 4 lines |
| 20:12.02 | boblutz | ) |
| 20:12.03 | atomspace | I agree |
| 20:12.22 | boblutz | i said " - X11 " as in minus X11 |
| 20:12.23 | atomspace | I have 3 lines running fine on a p3 1ghz with 512mb ram |
| 20:12.38 | edwin_quijada | boblutz: No X |
| 20:12.46 | boblutz | IM NOT RUNNING X |
| 20:13.01 | ZPertee | [TK]D-Fender, I figured out the solution to my external paging problem. I started looking at all of the equipment in the server/phone room and I found that there is a radio connected to the system by rca. so i'll just end up using my sound card and bypass my older pbx...Now if I can just find a decent command line audio player and I'll have it! |
| 20:13.02 | hardwire | capslock? |
| 20:13.05 | boblutz | yea, sorry |
| 20:13.38 | edwin_quijada | atomspace:for Ivr or everything? |
| 20:23.17 | atomspace | 10-4 |
| 20:23.26 | atomspace | edwin: yes |
| 20:23.39 | atomspace | sorry, runnin back and forth atm |
| 20:30.04 | weazahl | figured it out... the span numbers on the T1 ports where what wancfg_zaptel configured +1. changed that and it worked. of course * isnt answering calls now |
| 20:39.32 | Greek-Boy | when jumping from one extension to another using Gotoif, ${EXTEN} is empty and does not contain the extension that was originally called. Is this normal behaviour? |
| 20:41.34 | lesouvage | Is somebody running Asterisk on an iPhone without the problem of 100 % cpu load all the time. |
| 20:44.35 | Kobaz | haha, a roaming pbx |
| 20:44.38 | Kobaz | iphone, heh |
| 20:45.52 | lesouvage | Kobaz: It's more for fun then for serious use but Asterisk is runing pretty well on small devices. I had Asterisk running on a nokia 770 and it could handle the average soho load of calls. |
| 20:46.03 | Kobaz | nifty |
| 20:46.29 | lesouvage | Kobaz: ? |
| 20:46.36 | Kobaz | nifty? |
| 20:47.03 | lesouvage | Yes, sorry I don't know the word. |
| 20:47.06 | Kobaz | oh |
| 20:47.22 | Kobaz | good/nice/cool |
| 20:47.46 | Kobaz | http://dictionary.reference.com/browse/nifty |
| 20:48.02 | Kobaz | #2 |
| 20:48.12 | *** join/#asterisk my64 (n=my64@cro38-1-82-236-151-12.fbx.proxad.net) |
| 20:52.24 | lesouvage | Kobaz: problem with the Iphone is that it runs on 100% cpu. It changes the iPhone in a handy device to keep your hands warm. |
| 20:57.00 | boblutz | iFone r0x! |
| 20:59.38 | Yourname` | [Mar 24 15:46:41] NOTICE[29359]: rtp.c:1287 ast_rtp_read: Unknown RTP codec 126 received from '74.xx.xx194' -> There's 5-10 SIP peers behind this IP. How can I find out who is the rogue packet sender here? |
| 21:01.24 | Greek-Boy | lol |
| 21:01.54 | anonymouz666 | Yourname`: SIP debug |
| 21:05.27 | Yourname` | anonymouz666: done |
| 21:07.20 | *** join/#asterisk ice_croft (n=nolan@213.132.86.246) |
| 21:10.38 | JayTee52 | I've tried everything I can find and I still get choppy MOH. *headdesks* |
| 21:12.10 | boblutz | Hi, is there a fast/simple/easy way to figure out the exact zaptel driver needed for your card other than modprobing your way to insanity? |
| 21:13.07 | Greek-Boy | i'm surprised that asterisk doesn't have the option for customised ringback tones |
| 21:20.35 | *** join/#asterisk implicit (n=implicit@74.95.210.211) |
| 21:20.39 | lesouvage | greek-boy: you can do that with moh. Define a class, add the ringtone you want the caller to hear and use the m(class) option to pick the proper ring. |
| 21:25.38 | lesouvage | greek-boy: something like Dial(SIP/5000, 30, m(nice_ringback_tone)) |
| 21:28.19 | Greek-Boy | lesouvage: i tried that but the problem is that with every call it starts the song where it left off, doesn't start from beginning |
| 21:32.22 | *** join/#asterisk pfn (n=pfnguyen@hanhuy.com) |
| 21:33.34 | lesouvage | Greek-boy: You cuold use the M(<macro>) option, write a macro and play the sound in the background while the called party is called. Haven't tried this but it should work. Then you can pick any sound. |
| 21:33.51 | *** join/#asterisk radulovity (n=root@vs2055029.vserver.de) |
| 21:33.54 | radulovity | Hi! |
| 21:34.17 | radulovity | I have problem with configuring sip.conf |
| 21:34.49 | radulovity | i have make 2 accounts in this file, but i can log in only with one of them, and on another PC with Ekiga, can't connect |
| 21:35.18 | Greek-Boy | lesouvage thanks man |
| 21:35.19 | *** join/#asterisk BipBip (n=BipBip@bl8-135-168.dsl.telepac.pt) |
| 21:41.22 | lesouvage | radulivity: can you paste some cli output to pastebin. |
| 21:41.54 | lesouvage | ~pastebin |
| 21:41.54 | jbot | [~pastebin] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 21:41.56 | radulovity | lesouvage: sure ... |
| 21:42.02 | radulovity | wait moment :) |
| 21:44.10 | radulovity | http://pastebin.com/m3fbe78da |
| 21:45.49 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 21:46.51 | Katty | does asterisk do voice recognition yet? |
| 21:47.14 | _ShrikE | It does with lumenvox |
| 21:47.21 | *** join/#asterisk SomethingISODD (n=wno@S010600a0d1757bfb.cg.shawcable.net) |
| 21:47.29 | Katty | jbot: lumenvox |
| 21:47.40 | SomethingISODD | hello all question i am trying to set a internet radio station for my music onhold and i keep getting this error [Mar 24 17:44:13] NOTICE[28650]: res_musiconhold.c:531 monmp3thread: Request to schedule in the past?!?! |
| 21:47.41 | Katty | :< |
| 21:47.46 | Katty | _ShrikE: wiki? |
| 21:48.17 | radulovity | lesouvage: i can't make sip call between my Laptop and desktop |
| 21:48.29 | _ShrikE | Their site is www.lumenvox.com |
| 21:49.39 | radulovity | how can i test is asterisk able to handle with my ISDN card? |
| 21:50.01 | radulovity | If i call me from my mobile, to asterisk, nothing happes |
| 21:50.30 | lesouvage | radulovity: can you do "sip debug off" , do "set verbose 20" try to register both softphones then do "sip show peers" and paste the output to pastebin. Maybe that brings some light. |
| 21:50.49 | ber___ | somethingisodd, make sure to have your clock ntp synced |
| 21:51.02 | radulovity | lesouvage: sure, tnx for you time, i try this now ... |
| 21:51.35 | *** join/#asterisk Strom_C (n=strom@208.127.172.112) |
| 21:51.50 | Katty | _ShrikE: neat. |
| 21:53.29 | SomethingISODD | ber___ ok thanks |
| 21:55.09 | radulovity | lesouvage: hmmm, i don't become now any debuging information ... |
| 21:55.48 | radulovity | lesouvage: the asterisk is installed on my desktop, and i try to connect with ekiga from desktop to the asterisk on localhost |
| 21:56.10 | radulovity | but, no debuging information, and only message from ekiga "Registration Faild" |
| 21:56.21 | lesouvage | radulovity: there is no output on the cli while registering th softphones? |
| 21:56.36 | Rico29 | looks like a routing problem |
| 21:56.44 | Rico29 | if no message in CLI |
| 21:57.06 | radulovity | lesouvage: correct ... |
| 21:57.37 | *** join/#asterisk Dovid (n=Dovid@bzq-79-182-161-187.red.bezeqint.net) |
| 21:57.38 | radulovity | Rico29: hmmm, i beleive, my Ekiga are able to route to 127.0.0.1 :) |
| 21:57.48 | Rico29 | mmhm |
| 21:57.50 | Rico29 | yes |
| 21:57.52 | Rico29 | :D |
| 21:57.54 | radulovity | :) |
| 21:57.55 | Rico29 | firewall ? |
| 21:58.06 | radulovity | no, i am in DMZ ... |
| 21:58.48 | lesouvage | You have to open a terminal and type asterisk -r, then you enter the character line interface where Asterisk spit out everything happening on the asterisk server with "set verbose 20" |
| 21:58.53 | radulovity | i will to test one call with sip between laptop and desktop |
| 21:59.40 | radulovity | /usr/sbin/asterisk -vvvvvv -g -dddddd -c |
| 21:59.43 | radulovity | with this ... |
| 22:00.49 | lesouvage | radulovity: /usr/sbin/asterisk -vvvvvvvvr will do |
| 22:02.28 | radulovity | moment |
| 22:03.44 | radulovity | Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
| 22:03.50 | radulovity | but this file don't exist ... |
| 22:03.52 | radulovity | hmmm .... |
| 22:03.54 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 22:04.20 | lesouvage | raulovity: try /usr/sbin/asterisk I think your Asterisk server isn't running yet |
| 22:05.25 | radulovity | true ... |
| 22:06.42 | radulovity | http://pastebin.com/d580f73eb |
| 22:06.49 | lesouvage | radulovity: if ou now do (don't forget the "r" after all the v's) /usr/sbin/asterisk -vvvvvvvvr then you enter the cli |
| 22:07.46 | *** join/#asterisk craigk (n=craigk@58.174.150.119) |
| 22:08.01 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 22:08.02 | radulovity | i beleive that i am in cli |
| 22:08.12 | radulovity | with level 20 debugign |
| 22:08.18 | radulovity | please, look url up |
| 22:11.04 | lesouvage | Could you try to register testsip and paste the cli output (restarting the softphone would force a attempt to register) |
| 22:11.39 | radulovity | yes, i try ... |
| 22:11.58 | lesouvage | And could you paste sip.conf also (without passwords, public IP numbers and other dangerous info) |
| 22:12.39 | radulovity | You will not be able to receive incoming SIP calls. Please check that no other program is already running on the port used by Ekiga. |
| 22:12.48 | radulovity | hmmm, what of he port use Ekiga? |
| 22:13.19 | *** join/#asterisk Maliuta_ (n=nikolai@kiev.lusan.id.au) |
| 22:14.01 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 22:14.28 | *** join/#asterisk jameswf (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 22:15.09 | lesouvage | radulovity: I noticed that Ekiga uses port 5062 for the signaling while 5060 is the default SIP port for Asterisk. |
| 22:15.48 | radulovity | can i change port on ekiga? |
| 22:16.05 | *** join/#asterisk SteveTotaro (n=root@pool-71-179-116-45.bltmmd.east.verizon.net) |
| 22:17.31 | *** join/#asterisk BipBip (n=BipBip@bl8-135-168.dsl.telepac.pt) |
| 22:18.51 | lesouvage | radulovity: I have no idea but I guess you can. I really think it is a good idea if you first do some reading and studying. If you google for Ekiga asterisk I'm sure the info you need will pop up. You also had an isdn question and google will be your best friend for this question too. |
| 22:20.04 | radulovity | lesouvage: ok, understand, thank you for your time ... :\ |
| 22:21.26 | lesouvage | radulovity: see http://wiki.ekiga.org/index.php/Ekiga_as_an_Asterisk_client . Succes with Asterisk. As soon as you understand the basics the sky is the limit. See You ;-) |
| 22:23.05 | radulovity | Nice doc, i read it, but my problem with ISDN is not to simple, but hope that is google enought big ;D |
| 22:23.07 | *** join/#asterisk adrift (n=anon@71.143.169.172) |
| 22:26.04 | *** join/#asterisk sacitec (n=tobi@201.144.211.82) |
| 22:28.10 | *** join/#asterisk emist_ (n=emist@unaffiliated/emist) |
| 22:28.26 | lesouvage | radulovity: what card do you have |
| 22:31.25 | *** join/#asterisk Katty (n=The@adsl-68-92-250-115.dsl.stlsmo.swbell.net) |
| 22:33.05 | radulovity | moment ... |
| 22:33.36 | Katty | i has fajitas! |
| 22:33.42 | radulovity | looking now with lsof |
| 22:33.44 | radulovity | ekiga |
| 22:35.19 | radulovity | hmmm, ekiga use only port 1721 for H.323, but no information about any other used port .... |
| 22:35.34 | radulovity | i use Cologne Chip Designs GmbH ISDN network controller [HFC-PCI] (rev 02) |
| 22:35.38 | radulovity | this tell me lspci |
| 22:36.42 | lesouvage | radulovity: check http://users.pandora.be/Asterisk-PBX/InstallWildcard.htm . I think it will hep you. |
| 22:36.45 | Katty | why so quiet? |
| 22:37.01 | Yourname` | Can an agent stay logged into two queues? |
| 22:37.01 | Katty | pokes Qwell |
| 22:37.06 | radulovity | tnx :) |
| 22:37.09 | Yourname` | And get calls on both these queues? |
| 22:37.14 | Katty | Yourname`: i believe so, yes |
| 22:37.30 | Katty | Yourname`: tho i've not personally tried. |
| 22:38.45 | Yourname` | Thanks Katty |
| 22:38.54 | lesouvage | Yourname: yes, this should be possible, but use a recent version of asterisk, there has been serious problems with queus and agents with earlier 1.4 versions. |
| 22:38.57 | Katty | Yourname`: lemme know what happens. i'd be interested in finding out as well. |
| 22:40.02 | Katty | jbot: asterisk 1.8? |
| 22:40.08 | Katty | jbot: version 1.8 |
| 22:40.11 | Katty | jbot: 1.8? |
| 22:40.11 | jbot | hmm... 1.8 is doign a little better here, becuse it also scales tanks, but it used a realistic size - which would maek them very tiny on zoom 3, so I added a minimum pixel size |
| 22:40.22 | Katty | jbot: 2? |
| 22:40.23 | jbot | 2 is a number, silly |
| 22:40.28 | Katty | jbot: asterisk 2? |
| 22:40.33 | Katty | jbot: you infernal bot. |
| 22:43.17 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 22:48.03 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177582556.dsl.bell.ca) |
| 22:52.53 | *** join/#asterisk bsaxon (n=bryantsa@71-8-14-108.dhcp.leds.al.charter.com) |
| 22:53.58 | Yourname` | jbot: <reply> to "you infernal bot" "sure" |
| 22:55.31 | Yourname` | jbot, forget 2 |
| 22:55.31 | jbot | Yourname`: i forgot 2 |
| 22:55.59 | Yourname` | jbot: tell Katty she's cute |
| 22:56.38 | Yourname` | jbot: tell Katty about sipnat |
| 22:57.27 | Yourname` | jbot: 2? |
| 22:57.34 | draygon-w | jbot: tell draygon hes hot. |
| 22:57.49 | draygon-w | 6:57] <jbot> No, draygon-w, I won't. (target invalid?) |
| 22:57.50 | draygon-w | heh |
| 22:58.15 | Yourname` | jbot: 1.4? |
| 22:58.17 | *** join/#asterisk ManxPower (n=manxpowe@115.sub-75-202-11.myvzw.com) |
| 22:58.48 | Yourname` | jbot: 1.4.18.1 is <reply> Asterisk's killer new version! |
| 22:58.48 | jbot | okay, Yourname` |
| 22:59.10 | draygon-w | jbot: 1.4? |
| 23:02.54 | *** join/#asterisk cjk (n=cjk@d90-129-46-224.cust.tele2.lu) |
| 23:03.08 | mvanbaak | jbot: 1.4.18.1 |
| 23:03.09 | jbot | Asterisk's killer new version! |
| 23:04.05 | Yourname` | jbot: 1.4.18.1 |
| 23:04.06 | jbot | Asterisk's killer latest version! |
| 23:07.13 | *** join/#asterisk quigon (n=matias@190.3.121.15) |
| 23:09.39 | *** join/#asterisk axisys (n=axisys@ip70-174-179-120.dc.dc.cox.net) |
| 23:12.36 | *** join/#asterisk [hC] (n=hardcore@190.10.10.76) |
| 23:14.04 | atomspace | anyone here use a sipura 3000 ? |
| 23:15.28 | atomspace | With my ATA, my FXO port's "name" is getting passed along with the incoming call's number. So, instead of my name and my cellphone showing on the CID for those receiving the call, it says "sipura1FXO" and then my phone number. I'm not sure what "switch" i have flipped the wrong way... any suggestions? |
| 23:16.02 | *** join/#asterisk RoyK (n=roy@ip-105-51-149-91.dialup.ice.no) |
| 23:16.37 | ManxPower | maybe you should not set callerid= for that port in sip.conf |
| 23:17.44 | atomspace | ManxPower: http://pastebin.ca/955975 |
| 23:18.58 | atomspace | how would I go about "not setting" that ? Can you tell me from that context what to change? |
| 23:19.01 | ManxPower | atomspace: looks good to me, I guess you will have to pour thru the farious pages on the net that talk about that device and Asterisk |
| 23:19.14 | atomspace | booya, thats what I was hoping to hear |
| 23:19.18 | *** join/#asterisk Robba (n=rob@203.56.181.15) |
| 23:19.24 | atomspace | Manx: thanks anyway |
| 23:19.29 | Robba | Hi Guys |
| 23:19.42 | Robba | Hope everyone had a great Easter |
| 23:20.58 | Robba | Does anyone know why when dialling out it for the CLI displays the main number of our 100 number range instead of the DID Number? |
| 23:22.01 | Robba | i'm starting to think i broke something as it did work originally then just stopped one day |
| 23:22.57 | eric2 | is confused, bailing out |
| 23:31.43 | *** join/#asterisk davevg-btwtech (n=davevg-b@nj-67-76-177-147.sta.embarqhsd.net) |
| 23:33.16 | *** join/#asterisk obnauticus (n=obnautic@c-67-160-183-109.hsd1.wa.comcast.net) |
| 23:35.13 | *** join/#asterisk Mavvie (n=edwin@ppp121-44-58-231.lns10.syd7.internode.on.net) |
| 23:39.54 | thudd | anyone here have much experience with the manager API? |
| 23:41.08 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 23:43.42 | *** join/#asterisk radulovity (n=emil@212.200.221.174) |
| 23:44.17 | radulovity | googel known to much ... |
| 23:44.53 | *** join/#asterisk trnzmeta (n=bleh@iinet.guard.com.au) |
| 23:45.58 | *** join/#asterisk beasty (n=beasty@about/apple/macbook/beasty) |
| 23:46.01 | beasty | morning |
| 23:46.42 | radulovity | beasty: nigh :) |
| 23:48.33 | radulovity | have somebody try to use destar? |
| 23:53.56 | beasty | ok |
| 23:54.01 | beasty | my turn |
| 23:54.08 | beasty | anyone ever used gotoiftime ? |
| 23:56.29 | *** join/#asterisk coppice (n=chatzill@175.203.17.210.dyn.pacific.net.hk) |
| 23:58.27 | puzzled | beasty: yes |
| 23:59.48 | beasty | puzzled: can you show me how you solved it ? |
| 23:59.55 | beasty | cause i have a incomming iax2 trunk |