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| 00:09.44 | variable_office | it appears that externnotify is not called when you set pollmailboxes=yes; has anyone else encountered this? I want to make sure it is not just me |
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| 00:26.07 | ottoshmidt | OK, a simple question: Can I record landline phone calls through pci modem with asterisk?? |
| 00:27.10 | coppice | not unless your modem uses an intel 536 chip |
| 00:29.05 | ottoshmidt | coppice, i see... and probably not 'cause mine is an old genius 56kb modem |
| 00:29.41 | ottoshmidt | but is such thing available for linux? |
| 00:30.02 | ottoshmidt | any hints would be appreciated |
| 00:36.10 | usam | ottoshmidt, buy yourself a x100p card and use asterisk to record calls |
| 00:37.04 | ottoshmidt | x100? hmm |
| 00:37.22 | usam | you can find it cheap on ebay |
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| 00:40.45 | pabelanger-lap | ottoshmidt: no, modem is not a voicecard |
| 00:43.29 | coppice | a winmodem modem *is* a voice card. most just lack dadhi drivers |
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| 01:07.30 | boodu | hello |
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| 01:21.15 | DogBoy | ~book |
| 01:21.16 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
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| 01:24.38 | DogBoy | can't get this ring back free google voice thing working |
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| 05:14.27 | FILLVAIO3 | Guys! Need help! I have a g729 codec and using dtmfmode=inband, but g729 not supported inband only rfc2833. Is possible to use rfc2833 only if selected codec g729? |
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| 05:16.26 | jdoe | FILLVAIO3: inband won't work with g729, |
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| 05:18.54 | FILLVAIO3 | it is my quastion. I can use any of codecs right? is there possible to write any rule where will be if(g729)use rfc2833, elseif(alaw)use inband etc.... |
| 05:18.58 | FILLVAIO3 | ? |
| 05:19.37 | [TK]D-Fender | FILLVAIO3: The codec * will choose will ALWAYS be the same between 2 peers. Each may support 10 codecs, but the accepted on will always be the same so there isn't much point in allowing multiple or in leaving the DTMF moe in doubt. You should never want to use inband anyway. |
| 05:23.54 | FILLVAIO3 | [TK]D-Fender: Many thanx, I will try now. But if i have incoming call from analog line? What to do than? |
| 05:24.41 | [TK]D-Fender | FILLVAIO3: And how is that "analog line" arriving to *? |
| 05:25.56 | jdoe | ... something that just occurred to me, how does mixmonitor work with sip reinvites? |
| 05:26.51 | [TK]D-Fender | jdoe: It prevents them |
| 05:26.55 | [TK]D-Fender | jdoe: Entirely |
| 05:27.50 | jdoe | ah, I figured it would do something like that. |
| 05:27.50 | drclue | OK , I'm back again trying to sort out this Asterisk issue with playing DTMF tones. My research thus far only narrows things down to an entry in the dtmf log which reads (channel.c: Thread -1227400336 Blocking 'SIP/4096-00000000', already blocked by thread -1227646096 in procedure ast_waitfor_nandfds) .While it's nice to have a dtmf log message , I have no idea what to do about it |
| 05:31.17 | [TK]D-Fender | ok, checkout time, later all |
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| 06:12.10 | a_m_y | hi wanted to know if asterisk is able to run, under production, using xen |
| 06:12.14 | a_m_y | tnx in advance |
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| 06:36.39 | jblack | a_m_y: Yes, it can, but it leaves open the possibility of difficult to diagnose problems. I also haven't done hardware w/ asterisk under xen |
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| 07:05.53 | ranjan | Hi all |
| 07:06.21 | ranjan | what is the main purpose of using an IAXModem with asterisk???? |
| 07:10.01 | Corydon76-dig | Isolation of the fax signal to an external process |
| 07:10.10 | *** join/#asterisk Milad (~apple@unaffiliated/slackark) |
| 07:10.30 | Milad | anyone has a clue about this warning ? The device state of this queue member, SIP/216, is still 'Not in Use' when it probably should not be! Please check UPGRADE.txt for correct configuration settings. |
| 07:10.39 | *** join/#asterisk waschtl (~waschtl@3ed8a58a.d.d9tcloud.de) |
| 07:10.49 | Milad | I checked out UPGRADE.txt and do limitpeer things but my problem did not sloved |
| 07:11.01 | Milad | do I need to restart asterisk after limitpeer ? |
| 07:11.15 | Corydon76-dig | You need to reload the sip driver |
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| 07:12.45 | ranjan | Corydon76-dig, Hi, please can you give me the use of an IAXModem? |
| 07:12.49 | Milad | you mean module reload chan_sip.so ? |
| 07:13.17 | Corydon76-dig | Milad: correct |
| 07:13.27 | ranjan | Milad, Hi, please can you give me the use of an IAXModem? |
| 07:13.38 | Kyosh | ranjan, ask jesus |
| 07:13.53 | Milad | ranjan: what do you want exactly ? |
| 07:14.00 | Corydon76-dig | ranjan: I already told you what it's for |
| 07:14.27 | Corydon76-dig | I'm not going to configure it for you. |
| 07:14.46 | ranjan | Corydon76-dig, ok but let me ask one doubt...can that be used to send fax over a broadband connection?? |
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| 07:14.59 | Corydon76-dig | Nope |
| 07:15.18 | Corydon76-dig | Fax over IP is best done with T.38 |
| 07:15.21 | Milad | Corydon76-dig: and if after reload module problem still exist ? ( I reloaded module ) |
| 07:15.35 | Corydon76-dig | Milad: in all likelihood, you have a typo somewhere |
| 07:15.57 | Corydon76-dig | call-limit=999 is sufficient |
| 07:16.08 | ranjan | Corydon76-dig, that means with T.38 we can send FAX without a FAX Modem?? |
| 07:16.24 | Corydon76-dig | Makes utterly no difference what the limit is; it only matters that you have one |
| 07:16.39 | Kyosh | does askterisk support T.38 right out of the box or does T.38 have to be installed? |
| 07:16.51 | Milad | yes I have call-limit=1 by default for my all extension |
| 07:16.57 | Corydon76-dig | Kyosh: has to be installed |
| 07:17.02 | Kyosh | k |
| 07:17.20 | Kyosh | i'll go look for that info |
| 07:17.27 | Corydon76-dig | either res_fax_spandsp or the commercial fax module |
| 07:17.53 | Corydon76-dig | 1.6.2 and later only. |
| 07:18.36 | Kyosh | coryon: was that for me |
| 07:18.39 | Kyosh | ? |
| 07:18.48 | Corydon76-dig | You can't just add a magic module; there needs to be support in the core for fax |
| 07:18.56 | Corydon76-dig | Kyosh: yes |
| 07:19.17 | Kyosh | 1.6.2 and later on..what? it has to be compiled in? |
| 07:19.30 | Corydon76-dig | As in, not 1.4 |
| 07:19.43 | Kyosh | so T.38 will not work on 1.4? |
| 07:20.14 | Corydon76-dig | http://www.digium.com/en/docs/FAX/faa-download.php |
| 07:21.12 | Corydon76-dig | http://www.digium.com/en/docs/FAX/fax_faq.php question 7 |
| 07:22.30 | Kyosh | so ast 1.4 T.38 is not available. thats what i understood |
| 07:22.48 | Corydon76-dig | Correct |
| 07:23.38 | Kyosh | http://www.digium.com/en/docs/FAX/fax_faq.php question 2 |
| 07:23.51 | Kyosh | that appears to be a passthrough capability for T.38 |
| 07:24.17 | Kyosh | so if endpoint 1 is a T.38 fax device and endpoint 2 is a provider, it should work fine.. from what i understand |
| 07:25.07 | Corydon76-dig | Assuming you're using 1.6.2, yes |
| 07:25.11 | ranjan | Kyosh, can we send FAX without a fax modem that is using DSL connections with T.38 |
| 07:25.38 | Kyosh | corydon: thats not how i read it |
| 07:26.03 | Kyosh | "Asterisk 1.4 provides the capability to allow one T.38 enabled endpoint to setup a call to another T.38 enabled endpoint for the purposes of fax transmission. Asterisk 1.4 is not, though, capable of terminating those T.38 calls into Asterisk; they can only be passed through. " |
| 07:26.35 | Kyosh | which means that asterisk acts as a proxy for the call for the 2 endpoints to speak via T.38 |
| 07:26.36 | Corydon76-dig | Assuming that both endpoints are speaking T.38 over IP, then yes |
| 07:27.03 | Kyosh | and i assume there is a handshake that is checked for before initiating T.38? |
| 07:27.22 | Corydon76-dig | That is beyond my scope of knowledge |
| 07:27.53 | Kyosh | http://www.digium.com/en/docs/FAX/faa-download.php only shows 1.4.22 as the lowest version. what if i am using 1.4.21? might it still work? |
| 07:28.33 | Corydon76-dig | If you have problems, the first thing support will tell you to do is to upgrade |
| 07:28.41 | Kyosh | gawd |
| 07:29.08 | Kyosh | i got this 1.4.21 screaming! i mean it runs so smooth for over a year. havent rebooted it once! would hate to have to do it all over |
| 07:29.13 | Corydon76-dig | You are certainly welcome to try whatever you like on your own machines |
| 07:29.29 | Kyosh | yea i'd have to put up a test vm |
| 07:29.54 | Kyosh | would never take down a prod machine for this |
| 07:30.10 | Kyosh | i would just love to be able to show my boss that we have T.38 now |
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| 08:28.26 | Godfather_ | o/ |
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| 08:55.55 | Tweak-- | hello |
| 08:57.04 | E-bola | hi |
| 08:57.15 | Tweak-- | someone here that knows something about a linksys SPA941 ? |
| 08:58.12 | Tweak-- | [TK]D-Fender told me sla is not possible with asterisk... but now i see there is something else in the phone you can configure for SLA. |
| 08:58.12 | E-bola | easier if u ask the real question :) |
| 08:58.44 | Tweak-- | so maybe that will work, without the use of asterisk (im really a noob with this) |
| 08:58.55 | Tweak-- | http://linksys.custhelp.com/cgi-bin/linksys.cfg/php/enduser/std_adp.php?p_faqid=5234&p_created=1168838750&p_sid=HERqo36j&p_accessibility=0&p_lva=&p_sp=cF9zcmNoPTEmcF9zb3J0X2J5PSZwX2dyaWRzb3J0PSZwX3Jvd19jbnQ9MjQmcF9wcm9kcz0wJnBfY2F0cz0xNzIxLDE3MjMmcF9wdj0mcF9jdj0yLjE3MjMmcF9zY2ZfbGFuZz0xJnBfcGFnZT0xJnBfc2VhcmNoX3RleHQ9c3BhOTQx&p_li=&p_topview=1 |
| 08:59.09 | Milad | Corydon76-dig: in case of problem of state in queue still exist, I want to submit a ticket in bug tracker, what do you think I should include into ticket ? to get better answer ? any log ? or dump or ... ? |
| 09:00.35 | elliot98 | in the controlplayback application, how do I set it to use the ABCD dtmfs? |
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| 09:08.44 | Tweak-- | E-bola, you dont know if its possible? |
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| 09:14.46 | sawgood | Is there a 'quick' way to transfer a call straight to a VM box without 'ringing' the extension? |
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| 09:21.21 | E-bola | Tweak--: no idea |
| 09:36.48 | mathslinux | does anybody know what's the meaning of autofallthrough in extensions.conf? |
| 09:38.20 | Milad | Corydon76-dig: I update asterisk to 1.4.30 and my problem solved :) |
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| 11:10.35 | jasim | Hi, I'm using Asterisk in an SS7 network to make Out bound calls only. Should I worry about echo cancellation here? |
| 11:10.48 | jasim | There is no IVR - so no DTMF is required. |
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| 11:21.52 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 11:24.23 | jasim | Last question went unanswered.. I guess the qn was stupid.. However I've another question. |
| 11:25.01 | jasim | Is there any way to detect the required codec for a particular network and convert all voice files into the same format early enough so as to avoid transcoding? |
| 11:27.10 | *** part/#asterisk mifadir (29891eb6@gateway/web/freenode/ip.41.137.30.182) |
| 11:29.29 | leifmadsen | jasim: or no one knew the answer |
| 11:30.02 | jasim | Thats a nice way to think of it. Thanks :) |
| 11:30.40 | leifmadsen | jasim: if you just installed all audio files prior to the call setting up in as many formats as possible, Asterisk will always try and use the voice file of least cost (in regards to transcoding) |
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| 11:31.30 | jasim | Thats a great nugget of info. We've been doing out bound dialling on more than 20 E1s across different servers for quite some time and did not know about it. |
| 11:31.34 | jasim | Thanks!!! |
| 11:32.33 | jasim | Leif Madsen, I was wondering whether you've this kind of info in your book .. I haven't yet read it.. |
| 11:32.55 | fauxalliance | p00t |
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| 11:33.51 | leifmadsen | yes, should be in the first couple of chapters, or perhaps chapter 7 |
| 11:34.18 | jasim | Okie.. Thanks. |
| 11:34.52 | *** part/#asterisk neurosys (~neurosys@c-71-196-20-208.hsd1.fl.comcast.net) |
| 11:35.50 | jasim | I'm wondering how we can know whether Asterisk is doing transcoding or not.. Will there be any message shown in the CLI ? |
| 11:39.30 | redax | how the hell can I get rid of all vm-* messages to be played at all in VoiceMail app "? |
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| 11:40.56 | redax | deleting vm-* files in sounds/ :D |
| 11:42.36 | fauxalliance | good morning redax... i'd read further on the topic.... |
| 11:43.12 | redax | reading the source of app_voicemail :) |
| 11:43.21 | redax | morning fauxalliance |
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| 11:47.14 | redax | fauxalliance: ok. got it s<voicemail> goes silent |
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| 12:18.39 | *** join/#asterisk t_dot_zilla (~chatzilla@firewall-a.buf.ny.i-evolve.net) |
| 12:22.00 | evilbit | morning |
| 12:23.59 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 12:26.34 | rymkus | hi all! |
| 12:27.17 | rymkus | can someone direct me to which things should I debug for catch strange hangups |
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| 12:45.54 | *** join/#asterisk Sipster_ (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
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| 12:47.43 | xEBIx | I have got an Sipgate Account when it is called the phone 201 gets ringing, but why is he searching for a vm with the Sipgate ID, an not just the 201 one? |
| 12:58.17 | *** join/#asterisk grinder13 (~grinder@146.176.165.57) |
| 13:04.04 | evilbit | what's the difference btwn a macro and a subroutine in *? |
| 13:04.42 | pabelanger | ~collectdebug |
| 13:04.43 | infobot | rumour has it, collectdebug is a method of collecting logs allowing others help troubleshoot an issue. Refer to doc/HOWTO_collect_debug_information.txt or http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt |
| 13:04.45 | pabelanger | rymkus: ^^ |
| 13:05.27 | rymkus | pabelanger: thanks =) |
| 13:07.38 | leifmadsen | evilbit: macro will use a large amount of stack space and subroutines will not. Also, Macro() will implicitly return and GoSub() needs to be called with Return() |
| 13:07.53 | pabelanger | evilbit: core show application Gosub and core show application Macro |
| 13:08.02 | *** join/#asterisk n3hxs (~HAMming@69.171.176.98) |
| 13:08.09 | leifmadsen | basically, don't have a Macro() within a Macro() which calls func_odbc or Asterisk will run out of stack space and crash |
| 13:08.22 | leifmadsen | GoSub() is better when you want to next (much faster) |
| 13:12.03 | grinder13 | hello guys! there is a problem with a "482 loop detected" error. I 've setup a SIP trunk between two servers. when I 'm trying to call an extension on the second server either by using the originate CLI command or SIP communicator client I get this 482 error. When I am using Ekiga or sflphone the call is connected and the SIP trunk works without any hassle. I am using Asterisk SVN r271300. any thoughts? |
| 13:12.56 | *** join/#asterisk UQlev (~yuriy@212.50.100.76) |
| 13:14.58 | *** join/#asterisk n3hxs (~HAMming@iad-69-171-176-132.evdo.leapwireless.net) |
| 13:16.00 | [TK]D-Fender | grinder13: I think you should take that message for what it says.. the call is going circular and maybe you should really go look at it |
| 13:16.35 | grinder13 | yes, ok. but why with ekiga/sfphone clients the calls are passing through? |
| 13:17.57 | [TK]D-Fender | grinder13: Nothing "passes through. And you aren't showing anything, and we're not psychic |
| 13:18.08 | evilbit | thanks guys |
| 13:18.46 | tuxx- | hey guys, is it possible to set a hint status via AMI? we dont have devstate, and cant find another command to set a hint state.. |
| 13:19.11 | [TK]D-Fender | tuxx-: what version are you using? |
| 13:19.15 | tuxx- | 1.4.31 |
| 13:19.30 | [TK]D-Fender | tuxx-: then go install it |
| 13:19.38 | tuxx- | hehe |
| 13:19.47 | tuxx- | was afraid u were gonna say that ;-) |
| 13:20.35 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 13:20.37 | *** join/#asterisk coppice (~chatzilla@210.17.254.185) |
| 13:26.52 | grinder13 | here are my confs. Server A: http://pastebin.com/L0pacyML , Server B: http://pastebin.com/HzS739yE |
| 13:33.11 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-50.hsd1.nj.comcast.net) |
| 13:34.31 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:34.31 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:36.07 | *** join/#asterisk eliel (~eliels@201.234.94.226) |
| 13:36.30 | *** join/#asterisk cesar_CR (~cesar@201.192.86.30) |
| 13:37.12 | evilbit | I have my sip.conf define users by username... most examples I see for using exten are in the format of ${EXTEN:0:4} but that shows blank in terms of sip users. what should I be using? |
| 13:37.35 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 13:38.47 | [TK]D-Fender | evilbit: pardon? Using EXTEN? It holds what is DIALED. |
| 13:39.11 | evilbit | ah, then I misunderstood... how do I get the dialer? |
| 13:39.12 | [TK]D-Fender | evilbit: What is that supposed to tell us? All we can see it it taking the first four digits. |
| 13:39.31 | [TK]D-Fender | evilbit: Perhaps their CALLERID? <- |
| 13:40.02 | evilbit | ah, right... duh :-/ |
| 13:45.34 | leifmadsen | ${EXTEN:offset:length} |
| 13:46.50 | *** join/#asterisk minaguib (~mina@modemcable109.56-20-96.mc.videotron.ca) |
| 13:47.22 | [TK]D-Fender | ${ORANYVARIABLEORFUNCTION:offset:length} :) |
| 13:48.27 | evilbit | yeah, I was wrong about what was happening... I'm trying to setup a dynamic queue login but what I'm passing into the macro doesn't seem to be right... |
| 13:48.34 | evilbit | http://pastebin.com/n0AeRSKA |
| 13:52.08 | [TK]D-Fender | evilbit: eXCELLENT JOB HIDING OFF IMPORT PARTS OF THE DEBUG. |
| 13:52.15 | [TK]D-Fender | important* |
| 13:52.52 | evilbit | sorry |
| 13:53.06 | evilbit | I thought I grabbed all of the relavent stuffs |
| 13:53.35 | evilbit | what should I post? |
| 13:54.32 | [TK]D-Fender | evilbit: We don't see your READ being called, etc |
| 13:54.40 | [TK]D-Fender | evilbit: We don't see context headers... |
| 13:55.17 | evilbit | hmm, maybe verbosity isn't turned up enough? |
| 13:55.20 | evilbit | one sec |
| 13:55.23 | evilbit | let me re-run |
| 13:55.29 | *** join/#asterisk Carlos_PHX (~Carlos@ip68-99-199-10.ph.ph.cox.net) |
| 13:56.53 | pepselap | carlos_phx: i'm in phx :D |
| 13:57.38 | evilbit | how's this? http://pastebin.com/1gczqpp0 |
| 14:01.32 | Katty | hello my asterisk does nto work at all how to fix pls?????? |
| 14:02.03 | [TK]D-Fender | Katty: rm -rffffffukkit |
| 14:02.13 | evilbit | haha |
| 14:02.17 | Katty | [TK]D-Fender: yeshhh |
| 14:02.19 | Katty | [TK]D-Fender: also |
| 14:02.22 | Katty | [TK]D-Fender: i moved! |
| 14:02.26 | Katty | infobot: crittercam |
| 14:02.27 | infobot | extra, extra, read all about it, crittercam is Katty's live broadcast of The Nut House at http://ustre.am/8H5d |
| 14:02.27 | tzafrir_laptop | Katty, I have no idea what 'pls' is and how it is related to Asterisk |
| 14:02.43 | Katty | so crittercam is looking a bit different these days |
| 14:02.52 | [TK]D-Fender | evilbit: -- User disconnected <--- pardon? Where do we see that you entered anything? |
| 14:03.02 | Katty | no visitors yet, but they will find it soon enough, me thinks |
| 14:03.11 | [TK]D-Fender | Katty: Weren't you in a house before? Downsized? |
| 14:03.27 | Katty | yep i'm in a 2bedroom apt now |
| 14:03.34 | Katty | that's my balcon you're looking at |
| 14:03.38 | Katty | balcony |
| 14:03.57 | Katty | the 9000 pro doesn't do as well as the c900 |
| 14:04.01 | Katty | erm 6c00 |
| 14:04.06 | Katty | will have to swap cameras around tonight |
| 14:06.50 | [TK]D-Fender | Katty: Take it the other place was too big a budget hit? |
| 14:07.06 | *** join/#asterisk jpeeler (~jpeeler@asterisk/digium-software-dev/jpeeler) |
| 14:07.14 | Katty | [TK]D-Fender: not at all |
| 14:07.21 | evilbit | [TK]D-Fender: hmm, right... I guess I'm just bumbling with this. I'll go back and relook at it. I'd taken a example from the intarwebs |
| 14:07.22 | Katty | [TK]D-Fender: i got sick of dealing with ryan |
| 14:07.28 | Katty | [TK]D-Fender: decided i'd rather be single again |
| 14:07.41 | [TK]D-Fender | Katty: z0mg |
| 14:07.47 | Katty | [TK]D-Fender: so i took my doggy, and moved out |
| 14:08.26 | *** join/#asterisk dmz (~dmz@64.203.207.101.dyn-cm-pool-54.hargray.net) |
| 14:08.34 | [TK]D-Fender | Katty: Must have missed that update |
| 14:09.22 | Katty | [TK]D-Fender: well it happened pretty quickly |
| 14:09.26 | Katty | [TK]D-Fender: last sunday |
| 14:09.41 | Katty | [TK]D-Fender: been busy nearly every night packing and moving and all that fun stuff |
| 14:10.21 | Katty | [TK]D-Fender: my tinnitus is nearly gone |
| 14:11.21 | [TK]D-Fender | \o/ |
| 14:13.53 | evilbit | [TK]D-Fender: ok, got the queues working... thanks for pointing out the things I didn't realize |
| 14:16.46 | *** join/#asterisk ccesario (~ccesario@189-29-60-113-ac.cpe.vivax.com.br) |
| 14:18.01 | Katty | infobot: seen eppigy |
| 14:18.04 | infobot | eppigy <~eppigy@c-69-180-16-188.hsd1.ga.comcast.net> was last seen on IRC in channel #asterisk, 78d 29m 3s ago, saying: 'lol'. |
| 14:18.09 | Katty | whatttt |
| 14:18.12 | Katty | 78 days?! |
| 14:18.16 | Katty | goes eppigy-hunting |
| 14:19.56 | fenrus | =) |
| 14:20.07 | Katty | hey fenrus |
| 14:20.10 | Katty | long time no hug |
| 14:20.12 | Katty | hugs fenrus |
| 14:20.15 | fenrus | hey there katty |
| 14:20.17 | fenrus | *hugs* |
| 14:20.19 | Katty | what's the good word |
| 14:22.56 | [TK]D-Fender | Katty: Haven't you heard? |
| 14:23.28 | [TK]D-Fender | THE BIRD BIRD BIRD THE BIRD IS THE WORD!!! THE BIRD BIRD BIRD THE BIRD IS THE WORD!!! THE BIRD BIRD BIRD THE BIRD IS THE WORD!!! THE BIRD BIRD BIRD THE BIRD IS THE WORD!!! THE BIRD BIRD BIRD THE BIRD IS THE WORD!!! THE BIRD BIRD BIRD THE BIRD IS THE WORD!!! THE BIRD BIRD BIRD THE BIRD IS THE WORD!!! THE BIRD BIRD BIRD THE BIRD IS THE WORD!!! |
| 14:27.24 | Katty | infobot: bird is the word is http://www.youtube.com/watch?v=2WNrx2jq184 |
| 14:27.25 | infobot | ...but bird is already something else... |
| 14:27.32 | Katty | infobot: bird? |
| 14:27.33 | infobot | bird is, like, the word. |
| 14:27.47 | Katty | well. |
| 14:27.51 | Katty | i suppose that will work. |
| 14:31.27 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 14:31.35 | Katty | hi ariel_ |
| 14:31.45 | *** join/#asterisk n3hxs (~HAMming@69.171.176.49) |
| 14:32.29 | ariel_ | hi Katty |
| 14:32.32 | ariel_ | Morning folks |
| 14:34.49 | *** join/#asterisk Yoda_1204 (~chatzilla@12.111.169.98) |
| 14:34.58 | *** join/#asterisk nettie (~nettie@stewie.freax.it) |
| 14:37.23 | nettie | Hi guys, I'm buying a Digium TE121 Digital E1/T1 telephony card. I'm planinng using it with voip endpoints. I was wondering if the extra hardware echo cancellation douaghtercard is a must or not? Considering I'm staying all digital and voip endpoints will be deployed locally at first sight I don't see the need for it. Anyone know more on the subject please? Thanx in advance. |
| 14:38.21 | *** join/#asterisk hugorebelo (~hugo@200-171-132-124.completo.com.br) |
| 14:39.11 | WIMPy | You might get away without, but when bridging to voip you have a good chance of generation annoying echos. |
| 14:39.55 | [TK]D-Fender | nettie: the issue is the people you are calling over the TE aren't digital <- |
| 14:40.02 | [TK]D-Fender | nettie: this has little to do with your phones |
| 14:40.12 | [TK]D-Fender | nettie: And yes you want it |
| 14:40.25 | [TK]D-Fender | (HWEC that is) |
| 14:40.36 | [TK]D-Fender | nettie: Mind younot the card I'd use, but whatever |
| 14:41.22 | *** join/#asterisk auro (~auro@74-47-231-2.dr01.atmr.al.frontiernet.net) |
| 14:41.40 | *** join/#asterisk timeshell (~timeshell@gw.lusi.on.ca) |
| 14:41.59 | *** join/#asterisk Joe_CoT (~joecot@pdpc/supporter/active/joe-cot) |
| 14:47.50 | *** join/#asterisk auro (~auro@74-47-231-2.dr01.atmr.al.frontiernet.net) |
| 14:48.25 | Katty | hugs ariel_ |
| 14:49.52 | coppice | nettie: what is it in that setup which makes you think you don't need EC? |
| 14:50.21 | *** join/#asterisk viraptor (~viraptor@212.11.65.66) |
| 14:50.52 | viraptor | could someone tell me if I'm right here, or not? 1) if I want to connect a phone to my tdm card, I need FXS interfaces (on the card); 2) if I want to connect my tdm card to the phone line, I need FXO interfaces (on the card) |
| 14:51.34 | ariel_ | Katty: t/y, I needed that today. After fixing my car's 2 flat tires. (boy do fix a flat can's work well). |
| 14:52.35 | *** join/#asterisk auro (~auro@unaffiliated/auro) |
| 14:54.13 | [TK]D-Fender | viraptor: Yes |
| 14:54.31 | viraptor | [TK]D-Fender: thanks |
| 14:54.37 | [TK]D-Fender | viraptor: Thought for analog phone handsets I really recommend ATA's / mass gateways |
| 15:02.11 | rocksfrow | what advantages does asterisk 1.6 have other 1.4? other than supporting t.38 faxing? |
| 15:03.04 | Chainsaw | rocksfrow: Active development. |
| 15:03.48 | rocksfrow | Chainsaw, well, my current servers run on asterisknow, which only supported 1.4 when i installed them, but now i see asterisknow 1.7 allows the installation of 1.6! |
| 15:03.56 | [TK]D-Fender | rocksfrow: So did 1.5 |
| 15:04.07 | rocksfrow | [TK]D-Fender, are you sure? |
| 15:04.19 | rocksfrow | i had no option at instalation time |
| 15:04.23 | [TK]D-Fender | rocksfrow: Do you even read the channel topic in THEIR channel? |
| 15:04.48 | rocksfrow | what the hell |
| 15:04.54 | [TK]D-Fender | rocksfrow: Stop equating "comes with" vs "possible" and even "easy" |
| 15:05.01 | nettie | [TK]D-Fender I see thanx for the infos.. what you'll buy instead sangoma? |
| 15:05.04 | rocksfrow | [TK]D-Fender, suck it |
| 15:05.07 | Chainsaw | rocksfrow: 1.4 is now security-only, so even if you find a bug, manage to produce a reproducable test case and a patch... Digium will ignore you. So 1.6 is probably a worthy disruption to your workflow. Go for 1.6.2 |
| 15:05.26 | rocksfrow | awesome, thanks Chainsaw |
| 15:05.27 | Qwell | Chainsaw: No it isn't. |
| 15:05.29 | [TK]D-Fender | 1.6.0 & 1.6.1 aer security only as well now |
| 15:06.20 | Chainsaw | rocksfrow: But apparently Qwell disagrees, so perhaps you'll want to check with him what the official word is. |
| 15:06.32 | [TK]D-Fender | rocksfrow: 1.6 adds T.38, alternatives to meetme not requiring DAHDI, native devstate stuff and ton of other things we've taken for granted for years |
| 15:06.52 | rocksfrow | [TK]D-Fender, yeah, it's that t.38 i'm really worried about it |
| 15:07.20 | *** join/#asterisk brookshire (mbrooks@hijacked.us) |
| 15:07.33 | [TK]D-Fender | rocksfrow: http://forums.digium.com/viewtopic.php?p=128174 |
| 15:07.52 | [TK]D-Fender | rocsOver a year ago and is in #asterisknow 's channel topic and has been for a long long time |
| 15:08.15 | Chainsaw | rocksfrow: As always in an Asterisk upgrade I'd recommend getting a separate box that is identical to your production platform so you can migrate your configs and test without disrupting your day to day operation. Whether you're big or small... upset customers will ruin your day if you approach this wrong. |
| 15:08.40 | rocksfrow | Eh, i'm doing a fresh install |
| 15:08.42 | rocksfrow | i do have two servers |
| 15:08.44 | rocksfrow | its in failover |
| 15:08.52 | rocksfrow | so i'm just going to disable failover, install 1.6 on the backup server |
| 15:08.55 | rocksfrow | make the backup server the new primary |
| 15:09.00 | rocksfrow | and then setup the primary as the backup, with 1.6 |
| 15:09.02 | Qwell | Chainsaw: http://www.asterisk.org/asterisk-versions |
| 15:09.16 | Chainsaw | rocksfrow: *nod* As long as you don't expect a smooth, flawless transition you will be fine. |
| 15:09.27 | rocksfrow | Chainsaw, heh..yeah |
| 15:09.34 | *** join/#asterisk ruied (~ruied@89.180.246.185) |
| 15:09.39 | rocksfrow | i'm also considering getting an analog for the fax |
| 15:09.45 | rocksfrow | to delay the process some |
| 15:09.46 | rocksfrow | heh. |
| 15:09.58 | Chainsaw | rocksfrow: The TDM400 works well for me. The TDM410 should also be a good option. |
| 15:10.19 | Chainsaw | rocksfrow: Avoid the X100 please. It is just a rebadged PCI modem and it will perform as such. |
| 15:10.37 | rocksfrow | chainsaw...so what are you using the TDM for? |
| 15:10.41 | rocksfrow | can you elaborate |
| 15:10.50 | ruied | Hi, I have chan_lcr.so loading and working properly, but I don't know how can I change the language coming from lcr to 'es' for instance, instead of en |
| 15:10.51 | [TK]D-Fender | My guess would be lines & phones... |
| 15:10.52 | Chainsaw | rocksfrow: For fax service, it is connected to an analog BT line. |
| 15:11.08 | rocksfrow | oh, so you still run it all through the pbx though |
| 15:11.17 | rocksfrow | hrm |
| 15:11.18 | Chainsaw | rocksfrow: That analog line is separate from all other services. |
| 15:11.28 | rocksfrow | right, a separate trunk? |
| 15:11.35 | rocksfrow | im using PRI |
| 15:11.38 | rocksfrow | for everything |
| 15:11.52 | Chainsaw | rocksfrow: I have it presented on RJ11, completely separate, completely analog. |
| 15:11.54 | rocksfrow | what's frustrating is faxing was working perfectly fine |
| 15:11.55 | rocksfrow | before updating |
| 15:13.21 | ruied | is there a way to change the asterisk voice prompts coming from chan_lcr to "es" instead of "en", I have that for sip.conf, but do not know if it's possible with chan_lcr |
| 15:14.01 | Chainsaw | rocksfrow: What's going wrong now? |
| 15:14.14 | Chainsaw | rocksfrow: Is it answering the line but hanging up without responding appropriately? |
| 15:14.20 | rocksfrow | Chainsaw, it's weird, calling or faxing fax machiens will give a busy s ignal |
| 15:14.22 | Chainsaw | rocksfrow: Because that's a familiar failure mode. |
| 15:14.38 | rocksfrow | Chainsaw, yeah and says, 'other hangup' |
| 15:14.55 | rocksfrow | i was thinking maybe its the far end forcing t.38? |
| 15:15.16 | Chainsaw | rocksfrow: I suspect it's trying to hand over to a fax extension that doesn't exist or isn't configured correctly. Hard for me to be specific without the trace of a failed attempt though. |
| 15:15.54 | WIMPy | ruied: Not sure. Wanted to find out about that myself, but didn;t get round to doing so, yet. You could always set the variable yourself in the dialplan. |
| 15:18.19 | [TK]D-Fender | ruied: language=es |
| 15:19.27 | Trixboxer | Hi, how to avoid a X sip acc from getting registered at more than 1 location |
| 15:19.39 | *** join/#asterisk brycebaril (~bbaril@sea02-v612-nat.marchex.com) |
| 15:20.34 | ChannelZ | tell them not to do it |
| 15:21.00 | ChannelZ | but anyway the last person to register wins |
| 15:21.15 | Trixboxer | ChannelZ, any way to avoid it |
| 15:21.31 | Trixboxer | like giving some error that already registered or something ? |
| 15:21.51 | ruied | [TK]D-Fender, I have that for sip.conf but I need for chan_lcr... don't have any kind of lcr.conf |
| 15:22.13 | [TK]D-Fender | ruHow does the call arrive to *? |
| 15:22.23 | [TK]D-Fender | ruied: How does the call arrive to *? |
| 15:22.31 | *** join/#asterisk spiceycurry (~mcurry@proxy.hostopia.com) |
| 15:22.38 | ChannelZ | To what end? Someone has to drive back to work to unregister their phone? |
| 15:23.40 | [TK]D-Fender | Trixboxer: restrict the IP's they can auth from. |
| 15:24.44 | fauxalliance | yep. tell [TK]D-Fender what you need... and he'll tell you how to get along without it... |
| 15:25.16 | spiceycurry | Have a question. If I am receiving a call from 4161231234, is there a way to mask that number to another? (Like an alias?) What we are trying to do, is map 1-800 numbers to our lines, but want them to be masked to the local numbers via some type of dlal plan |
| 15:25.50 | [TK]D-Fender | spiceycurry: the call arrives on whatever exten it arrives on. Do whatever you want after. thats what the dialplan is for |
| 15:26.04 | fauxalliance | spiceycurry like call forward... or CID based routing.. or blacklisting. |
| 15:26.23 | xheliox | step away from the pbx.. |
| 15:26.32 | spiceycurry | lol |
| 15:26.47 | spiceycurry | I touched nothing! |
| 15:26.54 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 15:27.20 | Trixboxer | [TK]D-Fender, extensions are behind NAT so on server it shows a Single IP for all |
| 15:28.10 | [TK]D-Fender | Trixboxer: Ok. And? You said not from multiple IP's. So restrict where they can reg from. |
| 15:28.21 | [TK]D-Fender | Trixboxer: What does proximity to other phones have to do with this? |
| 15:28.26 | *** join/#asterisk raden_work (~jon@69-179-99-17.stat.centurytel.net) |
| 15:29.19 | Trixboxer | Server at X public IP and extensions are behind Y public IP .. |
| 15:29.36 | *** part/#asterisk viraptor (~viraptor@212.11.65.66) |
| 15:29.41 | [TK]D-Fender | Trixboxer: Yes, restrict it to THAT IP <- |
| 15:29.46 | ruied | [TK]D-Fender, from a context [misdn] but, it's coming from chan_lcr.so |
| 15:29.55 | Trixboxer | oh wow |
| 15:30.41 | Trixboxer | so if its a LAN IP say 10.10.10.X behind any public IP , can I put that IP in allowed list in asterisk ?? |
| 15:30.42 | [TK]D-Fender | ruied: Go set it int he dialplan if you don't have the option at the channel driver level |
| 15:30.58 | [TK]D-Fender | Trixboxer: LAN IP = MEANINGLESS. its the WAN |
| 15:31.00 | *** join/#asterisk FredericoLeao (~fred@201.90.113.74) |
| 15:31.00 | Katty | BIRD |
| 15:31.09 | [TK]D-Fender | TriIt reg's listing the PUBLIC, not the PRIVATE |
| 15:31.19 | ruied | [TK]D-Fender, I can us that in extensions.conf? goingo to try... |
| 15:32.12 | Katty | yay. |
| 15:32.17 | Katty | my first visitor is a cardinal |
| 15:33.38 | Trixboxer | extensions are getting registered from multiple LAN ips using a single WAN IP |
| 15:38.10 | ruied | [TK]D-Fender, I've tried to add language=pt_m at [misdn] context in extensions.conf and didn't work... |
| 15:38.33 | [TK]D-Fender | ruied: I don't see how you attempted to set it. |
| 15:38.51 | [TK]D-Fender | Trixboxer: so restrict to that WAN IP |
| 15:39.25 | Trixboxer | [TK]D-Fender, thats not possible |
| 15:39.33 | [TK]D-Fender | Trixboxer: yes, it is |
| 15:39.39 | Trixboxer | [TK]D-Fender, I need to allow that IP |
| 15:39.50 | [TK]D-Fender | [11:38]<[TK]D-Fender>Trixboxer: so restrict to that WAN IP <----------- |
| 15:39.55 | *** part/#asterisk spiceycurry (~mcurry@proxy.hostopia.com) |
| 15:39.58 | Trixboxer | [TK]D-Fender, its my other office.. i cant restrict |
| 15:39.59 | *** join/#asterisk matagou_ (57f8bf49@gateway/web/freenode/ip.87.248.191.73) |
| 15:40.16 | [TK]D-Fender | Trixboxer: Restrict it ***TO*** that IP. |
| 15:40.26 | [TK]D-Fender | reaches for his ClueBat (tm) |
| 15:40.32 | matagou_ | hello, asterisk users |
| 15:40.33 | evilbit | for dynamic meetme do the rooms need to be setup in meetme.conf [rooms] beforehand or does the app take care of that? |
| 15:40.50 | matagou_ | please tell me your opinion about https://issues.asterisk.org/view.php?id=17575 |
| 15:40.51 | [TK]D-Fender | evilbit: the app.. if you had to define all of them then they wouldn't be DYNAMIC |
| 15:41.05 | evilbit | ok, thx |
| 15:41.43 | *** join/#asterisk darkskiez (~dz@62-50-207-43.client.stsn.net) |
| 15:42.32 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 15:43.00 | Trixboxer | [TK]D-Fender, If i restrict the WAN ip all extensions will get blocked , blocking their local IP 10.10.10.x wont help as the request will come from WAN IP |
| 15:43.03 | *** part/#asterisk rossand (~aross@CPE485b390978ce-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 15:43.40 | Trixboxer | I just want that one person uses only one extension at his phone only |
| 15:43.54 | [TK]D-Fender | ...... |
| 15:44.16 | *** join/#asterisk spiceycurry (~mcurry@proxy.hostopia.com) |
| 15:44.17 | [TK]D-Fender | Trixboxer: I didn't say BLOCK that IP I said ONLY ACCEPT CONNECTIONS FROM THAT ONE FUCKING IP <--------------- |
| 15:44.18 | n3hxs | Maybe ALLOW only that IP |
| 15:44.45 | [TK]D-Fender | restrict it *********TO********* |
| 15:45.48 | Trixboxer | I guess I failed to explain.... hmm rewriting with exact IP's |
| 15:46.17 | [TK]D-Fender | Trixboxer: You trying to say that he has multiple devices behind there all trying to reg to the same peer? |
| 15:46.26 | Trixboxer | yes |
| 15:46.50 | *** join/#asterisk Naikrovek (~jjohnson@unaffiliated/naikrovek) |
| 15:46.54 | [TK]D-Fender | Shouldn't have taken this to squeeze it out. |
| 15:47.01 | Trixboxer | :) |
| 15:47.02 | *** join/#asterisk retentiveboy (~pdugas@69.169.199.82) |
| 15:47.03 | *** join/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell) |
| 15:47.03 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 15:47.08 | evilbit | hmm, in dynamic meetme how do you ensure that the user doesn't choose a conference number that corresponds to a existing extension? |
| 15:47.36 | [TK]D-Fender | Trixboxer: No real way to stop the reg from taking. You CAN restrict calls manually in the dialplan, but that's it. |
| 15:48.07 | [TK]D-Fender | evilbit: WTF does a "conference number" have to do with an "extension number"? |
| 15:48.39 | Trixboxer | [TK]D-Fender, thats why I was stuck.. I guess i need to put some change in asterisk code .. ( need to learn a lot ) |
| 15:48.43 | evilbit | I don't know... by your answer I guess nothing, lol |
| 15:49.00 | [TK]D-Fender | Trixboxer: tell them not to be idiots |
| 15:49.20 | [TK]D-Fender | Trixboxer: or run a SIP proxy in front and restrict the UA, etc |
| 15:49.51 | Trixboxer | [TK]D-Fender, world would be so safe place if ppl do as said :) |
| 15:49.56 | Trixboxer | anyways thanks for your time |
| 15:50.06 | [TK]D-Fender | Trixboxer: then again a proxy could allow multiple phones to simultaneously reg to the same peer. |
| 15:51.07 | Trixboxer | give me you bat and I'll beat them if they do it again... |
| 15:52.36 | *** join/#asterisk n3hxs (~HAMming@63.68.135.4) |
| 15:55.59 | spiceycurry | Am I correct in saying... Dial(...., 1, 2); 1= source, 2 = destination ? |
| 15:56.38 | spiceycurry | 1= local channel, 2= what to dial |
| 15:56.41 | [TK]D-Fender | spiceycurry: No. |
| 15:56.43 | Naikrovek | no |
| 15:56.51 | [TK]D-Fender | spi"core show application dial"<- |
| 15:56.58 | [TK]D-Fender | spiceycurry: "core show application dial"<- |
| 15:57.00 | spiceycurry | ah, ok, thanks |
| 15:57.14 | *** join/#asterisk minaguib (~mina@modemcable109.56-20-96.mc.videotron.ca) |
| 15:57.15 | [TK]D-Fender | spiceycurry: the current call is the source. |
| 15:57.50 | spiceycurry | Ok, I'll have to re-read this section |
| 15:58.50 | spiceycurry | thats great, I didn't know about that help |
| 15:58.51 | spiceycurry | thanks |
| 15:58.52 | *** join/#asterisk outtolunc (~me@adsl-66-218-53-172.dslextreme.com) |
| 16:00.29 | xheliox | ~book |
| 16:00.30 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 16:00.37 | *** join/#asterisk ruied (~ruied@89.180.248.186) |
| 16:00.53 | spiceycurry | I have the book, reading page 131 now- building an interactive dialplan |
| 16:01.35 | Trixboxer | [TK]D-Fender, while searching came contactpermit this might help me as I;m getting their local IPs in it :) |
| 16:02.03 | [TK]D-Fender | Trixboxer: * shouldn't be listening to that. |
| 16:02.50 | spiceycurry | trying to figure out how to forward the current call to an extension |
| 16:03.19 | Trixboxer | u sure ? will try tomo .. |
| 16:03.34 | [TK]D-Fender | spiceycurry: No such thing as "forwarding". And extension = line in extensions.conf |
| 16:03.57 | Trixboxer | I hope it work |
| 16:04.23 | spiceycurry | So if the current call comes in on a particular dial pla, could I then Dial(... to move it ? |
| 16:04.29 | spiceycurry | *dial plan |
| 16:05.09 | spiceycurry | i.e: Dial(${exten}) |
| 16:06.24 | *** join/#asterisk iscario (~quassel@LMontsouris-156-24-36-77.w217-128.abo.wanadoo.fr) |
| 16:07.20 | *** join/#asterisk ruied (~ruied@89.180.253.82) |
| 16:07.54 | *** join/#asterisk iscario (~quassel@laureades.davout02.net1.nerim.net) |
| 16:08.14 | *** join/#asterisk florz (nobody@2001:1a50:503c::1) |
| 16:11.32 | *** join/#asterisk florz (nobody@2001:1a50:503c::1) |
| 16:15.20 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-50.hsd1.nj.comcast.net) |
| 16:17.46 | evilbit | if I want auth'd users to be able to create conferences, and un-auth'd users just to join with conf num and pin is it best to create two different extensions so that un-auth'd can't create a conf? |
| 16:23.28 | *** join/#asterisk wcselby (~wcselby@216.110.88.194) |
| 16:23.30 | wcselby | o/ |
| 16:26.35 | spiceycurry | how di I override the caller id in a dial plan? |
| 16:26.37 | spiceycurry | *do |
| 16:26.55 | spiceycurry | Set(??) |
| 16:27.11 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 16:27.35 | spiceycurry | xxxx, n,Set(CALLERID(text)=555-1212) ? |
| 16:28.36 | spiceycurry | I think I found it... http://www.voip-info.org/wiki/view/Asterisk+cmd+SetCallerID |
| 16:29.14 | spiceycurry | Woops: http://www.voip-info.org/wiki/view/Setting+Callerid |
| 16:31.47 | [TK]D-Fender | spiceycurry: core show function CALLERID |
| 16:33.02 | spiceycurry | [TK]D-Fender: would the variables ${CALLERIDNAME} and ${CALLERID} still work with the new command? |
| 16:33.27 | spiceycurry | Hmm, I see there is new getters/setters... |
| 16:33.34 | spiceycurry | I;ll test |
| 16:33.35 | wcselby | spiceycurry - I think that's for older versions (1.2) |
| 16:34.06 | wcselby | spiceycurry - look at the options you have for the function CALLERID |
| 16:34.16 | wcselby | i.e "Core show function CALLERID" from the CLI |
| 16:34.16 | spiceycurry | ah I see |
| 16:34.35 | spiceycurry | I take it they do not have to be ${} wrapped |
| 16:34.56 | spiceycurry | Ok, I have it :D |
| 16:35.31 | wcselby | all functions (that I know of) need to be wrapped in ${} |
| 16:35.48 | spiceycurry | it looks like they are variables, not funcitons |
| 16:36.19 | [TK]D-Fender | spiceycurry: do NOT attempt to use the old vars |
| 16:36.24 | spiceycurry | I think I do- Set(CALLERID(num)=1324) |
| 16:36.44 | spiceycurry | and for name Set(CALLERID(name)="Mike") |
| 16:37.38 | spiceycurry | cool- Set(CALLERID(all)= Testing (555-1212)) |
| 16:37.40 | spiceycurry | that works |
| 16:44.28 | p3nguin | It shouldn't. |
| 16:45.05 | spiceycurry | Set(CALLERID(number)=9052981234) works |
| 16:45.08 | nettie | [TK]D-Fender hi again, just wondering if you're suggesting sangoma instead of Digium for a PRI card. I'm asking because afaik Sangoma was always a step ahead of Digium but now the games seems rather filled specwise. |
| 16:45.17 | p3nguin | The format for CALLERID(all) should be "Name <number>" |
| 16:45.25 | p3nguin | Notice the angled brackets. |
| 16:45.26 | spiceycurry | that example there had wrong parenthesis |
| 16:45.29 | spiceycurry | yea |
| 16:45.34 | wcselby | p3nguin - no quotes, though. |
| 16:45.50 | spiceycurry | I decided to leave the name as-is, and just overwrite the # |
| 16:46.06 | spiceycurry | no sense in killing a perfectly good name |
| 16:46.07 | spiceycurry | :D |
| 16:46.35 | [TK]D-Fender | spiceycurry: Politicians do it all the time. |
| 16:46.39 | spiceycurry | I was going to change the CID name on every caller id to "T.38 WHAT?" just to piss my boss off |
| 16:46.47 | spiceycurry | haha |
| 16:47.09 | *** join/#asterisk cesar_CR (~cesar@201.192.86.30) |
| 16:49.09 | *** join/#asterisk [T]ank (~T]ank@206.71.78.158) |
| 16:50.10 | [T]ank | I have installed pwlib pwlib-dev opal and opal-devel but still asterisk make menuselect shows that I have not met the dependencies for enabling h323. Can anyone please help? |
| 16:50.27 | wcselby | [T]ank - have you re-run ./configure? |
| 16:50.48 | [T]ank | yes, i did a make clean ; ./configure ; make menuselect |
| 16:51.03 | wcselby | do you have the right versions of each of those pre-reqs? |
| 16:51.05 | *** join/#asterisk bran (~info@unaffiliated/bran) |
| 16:51.23 | *** join/#asterisk sgimeno (~chatzilla@95.122.8.54) |
| 16:51.48 | [T]ank | that I do not know.... I have the current repo versions provided by CentOS5.5 |
| 16:52.01 | [T]ank | did a yum install of all of those packages. |
| 16:53.32 | *** join/#asterisk rustyclarkson (~rusty@u53.sutus.com) |
| 16:54.35 | wcselby | it could be that yum installed a newer version than what the h323 is looking for, or it could be that yum installed them somewhere asterisk doesn't know to look....lots of things. i haven't installed h323 myself, so I'm not sure. I do remember some back and forth on the asterisk-users list recently about getting h323 to work with current asterisk versions, you may want to search there. |
| 16:54.57 | rrb3942 | have you tried doing a make clean or make distclean before re-runing ./configure? |
| 16:55.11 | [T]ank | I have done both make clean and make distclean |
| 16:57.00 | wcselby | does the polycom ip550 take an expansion module, or does that only work on the 6xx models? |
| 16:58.43 | [TK]D-Fender | wcselby: 6XX |
| 17:00.38 | *** join/#asterisk Lantizia (~Lantizia@erebus.seaquake.net) |
| 17:01.13 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 17:02.16 | wcselby | [TK]D-Fender - thanks |
| 17:05.46 | *** join/#asterisk srbaker (~srbaker@S0106c03f0e8ab2c2.cc.shawcable.net) |
| 17:13.45 | minaguib | Hey folks. I'm having trouble with the "forward" feature of some phones here. When forwarding to a number is activated and the phone is called, it rings briefely, but then in the asterisk console I see forwarding initiated to a wrong number. It seems "909" is prepended to the number the user set in the phone. Any ideas ? |
| 17:14.09 | minaguib | I can't find "909" mentioned in any of the config files |
| 17:14.32 | [TK]D-Fender | minaguib: * prepends nothing. Look at your phones. |
| 17:14.36 | spiceycurry | check the phones? |
| 17:17.38 | *** join/#asterisk jtodd (ubx7jwdyw9@asterisk/community-director-and-tie-dye-shirt-lover/jtodd) |
| 17:17.38 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 17:18.52 | [T]ank | following a couple of tutorials, i am doing a make opt on openh323 give me this output and error: http://www.pastebin.com/JDHw0dDY, i am not finding anything helpful on google. Any ideas? |
| 17:18.52 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-147.cablep.bezeqint.net) |
| 17:22.23 | *** join/#asterisk sarah_p (~sarah_p@71.117.223.226) |
| 17:25.02 | *** join/#asterisk retentiveboy (~pdugas@69.169.199.82) |
| 17:25.48 | rrb3942 | do you have your kernel headers installed? |
| 17:26.37 | *** join/#asterisk sarah_p_ (~sarah_p@108.1.169.2) |
| 17:26.56 | [T]ank | yes.... however, i am just wondering... after kernel headers are install, i probably need to reboot. rebooting to test |
| 17:29.42 | auro | reboot for kernel HEADERS? |
| 17:30.11 | xheliox | sighs |
| 17:30.36 | xheliox | auro: Yes, rebooting 3 times fixes everything. |
| 17:31.03 | [T]ank | *** sensing sarcasm *** |
| 17:31.21 | [T]ank | im currently re-running the make opt... we will see what the outcome is. |
| 17:34.22 | [T]ank | ok, sooooo... rebooting did not change the result. I have installed kernel and kernel-devel. That should install the kernel headers, correct? |
| 17:36.07 | Katty | ATTENTION |
| 17:36.10 | Katty | I HAS SPAHGETTIS |
| 17:36.12 | Katty | that is all. |
| 17:36.24 | rrb3942 | what is the output of 'ls /usr/src/kernels/' and 'uname -rm'? |
| 17:37.05 | *** join/#asterisk RobH (~robh@wikimedia/RobH) |
| 17:38.19 | [T]ank | rrb3942:[root@sahara ~]# ls /usr/src/kernels/ |
| 17:38.19 | [T]ank | 2.6.18-164.el5-x86_64 2.6.18-194.8.1.el5-x86_64 |
| 17:38.19 | [T]ank | [root@sahara ~]# uname -rm |
| 17:38.19 | [T]ank | 2.6.18-194.8.1.el5 x86_64 |
| 17:38.19 | [T]ank | [root@sahara ~]# |
| 17:39.03 | rrb3942 | yep looks like they are installed |
| 17:39.11 | [T]ank | ok. |
| 17:44.52 | tzafrir_laptop | [T]ank, what's the output of: ls -l /lib/modules/`uname -r`/build |
| 17:45.29 | [T]ank | ls -l /lib/modules/`uname -r`/build |
| 17:45.29 | [T]ank | lrwxrwxrwx 1 root root 50 Jul 2 11:01 /lib/modules/2.6.18-194.8.1.el5/build -> ../../../usr/src/kernels/2.6.18-194.8.1.el5-x86_64 |
| 17:45.29 | [T]ank | [root@sahara ~]# |
| 17:47.27 | [TK]D-Fender | [T]ank: No, headers are in another package. |
| 17:47.41 | [T]ank | ? |
| 17:48.55 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.167.9.dsl.dyn.forthnet.gr) |
| 17:49.01 | [T]ank | kernel-devel is not the package that provides headers? |
| 17:49.52 | carrar | Looks like centos 5.5 |
| 17:49.59 | [T]ank | corect |
| 17:50.02 | [T]ank | correct |
| 17:50.16 | rrb3942 | yeah he is right, there is also the kernel-headers package |
| 17:50.33 | [T]ank | *** Checking |
| 17:51.24 | [T]ank | Package kernel-headers-2.6.18-194.8.1.el5.x86_64 already installed and latest version |
| 17:53.38 | *** join/#asterisk JamesHarrison (~jharr@hometree.mmmetrics.co.uk) |
| 17:53.41 | [TK]D-Fender | [T]ank: trash your extracted folder and start from scratch |
| 17:54.16 | [T]ank | which extracted folder? openh323? |
| 17:55.51 | [TK]D-Fender | [T]ank: * addons |
| 17:56.44 | JamesHarrison | I'm having issues making Asterisk respond to SIP from a PSTN-SIP trunk; I just get a "Ignoring this INVITE request" message if I enable SIP debugging (1.6.2.10-rc1). I'm new to Asterisk and looking for some ideas on where to work out how to debug this further, ie working out why it's ignoring it |
| 17:57.16 | [T]ank | [TK]D-Fender: doesnt asterisk get installed before addons? Im still trying to get asterisk installed. |
| 17:57.38 | *** join/#asterisk aidinb (~Aidin@wsip-98-190-45-225.oc.oc.cox.net) |
| 17:57.59 | [T]ank | if I am wrong that that is the whole issue, i sincerely apologize for wasting everyones time. |
| 17:58.06 | *** join/#asterisk JB_DC (~JBrule@206.205.228.222) |
| 17:58.47 | *** part/#asterisk JB_DC (~JBrule@206.205.228.222) |
| 17:59.33 | [TK]D-Fender | [T]ank: Then trash * and redo the folder |
| 18:04.38 | *** join/#asterisk theHub (~theHub@69.177.93.21) |
| 18:07.45 | wcselby | [T]ank - http://lists.digium.com/pipermail/asterisk-users/2010-June/249920.html |
| 18:07.58 | wcselby | read that thread - doesn't look good |
| 18:08.55 | Naikrovek | vSphere 4 Enterprise licensed per-cpu? |
| 18:09.10 | *** join/#asterisk phidelta (~philipp@mk092248121166.a1.net) |
| 18:09.17 | *** join/#asterisk retentiveboy (~pdugas@69.169.199.82) |
| 18:09.24 | Naikrovek | whoops wrong chan |
| 18:09.30 | phidelta | Hi everyone |
| 18:09.42 | wcselby | lol Naikrovek |
| 18:10.24 | phidelta | I have a rather strange problem with CDRs, where my records are prematurely written so I always get billsec=1 on 1.6.2.9 |
| 18:10.45 | phidelta | The call is established via the AMI between 2 Local/ channels |
| 18:11.13 | phidelta | does anyone have any idea where I could start looking for answers (google, etc have been tried ;) ) |
| 18:15.20 | *** join/#asterisk gadams999 (~gadams999@173-165-184-25-atlanta.hfc.comcastbusiness.net) |
| 18:26.16 | [T]ank | haha.... GOT IT!!!! |
| 18:26.43 | [T]ank | copied compiler.h from the kernel headers into the directory it was complaining about. |
| 18:26.45 | [T]ank | works |
| 18:26.47 | [T]ank | thanks all |
| 18:32.15 | *** join/#asterisk SiNGLer (~singler@88.118.8.230) |
| 18:36.03 | *** join/#asterisk Nugget (nugget@carrera.macnugget.org) |
| 18:40.36 | gloin | Weirdish problem: I have a * box set up to a SIP trunk, and from SIP extensions within * I can dial out fine, but when I try to dial out using AMI, I get Failed to authenticate on INVITE to '"asterisk" <sip:asterisk@ipaddress> |
| 18:40.55 | evilbit | does this seem like a "sane" way to setup authenticated and pin-only based conferences? |
| 18:41.02 | gloin | is there a second level of auth I need to do from manager, or a permission I need to set in manager.conf? |
| 18:43.20 | *** join/#asterisk bmg505 (~leon@196-209-7-27.dynamic.isadsl.co.za) |
| 18:45.06 | *** join/#asterisk [sr] (~Unknowned@pal-213-228-140-150.netvisao.pt) |
| 18:45.12 | [sr] | 41ºC today :| |
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| 18:57.23 | jaxyeh | Im trying to initiate a simple SIP call without extension (direct IP Address dialing), how can I setup dial plan to process this approach? |
| 19:03.15 | xheliox | ~book |
| 19:03.16 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
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| 19:23.43 | [TK]D-Fender | jaxyeh: Dial(SIP/something@somewhere.tld) |
| 19:24.37 | Joe_CoT | Meetme options - t: Set talk only mode. (Talk only, no listening). Is there a real world case where anyone would ever want to do this? |
| 19:24.52 | [TK]D-Fender | Joe_CoT: Obviously |
| 19:24.53 | Qwell | Joe_CoT: broadcast messages |
| 19:25.10 | jaxyeh | [TK]D-Fender: I figured it out by using the "s" extension⦠thanks |
| 19:25.43 | Joe_CoT | but then why wouldn't you just use listen-only for the other participants? Was it just for completeness i guess? |
| 19:27.18 | *** join/#asterisk stix (~stix@exchange2003.corporate.billetkontoret.dk) |
| 19:27.37 | *** part/#asterisk jaxyeh (~jaxyeh@c-69-250-52-161.hsd1.md.comcast.net) |
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| 19:27.44 | [TK]D-Fender | joeYou are now talking about 2 DIFFERENT people |
| 19:28.10 | [TK]D-Fender | Joe_CoT: TALK only for the announcer, LISTEN only for the listeners |
| 19:28.19 | *** part/#asterisk jaxyeh (~jaxyeh@c-69-250-52-161.hsd1.md.comcast.net) |
| 19:28.22 | *** join/#asterisk jaxyeh (~jaxyeh@c-69-250-52-161.hsd1.md.comcast.net) |
| 19:29.14 | Joe_CoT | [TK]D-Fender, right, but what does it matter if I already have listen only for the listeners? It's effectively talk only for the announcer. whereas if I had talk only for the announcer and nothing special for the listeners, it's just the announcer talking while everyone else makes fun of him. |
| 19:29.50 | *** part/#asterisk jaxyeh (~jaxyeh@c-69-250-52-161.hsd1.md.comcast.net) |
| 19:29.51 | *** join/#asterisk jaxyeh (~jaxyeh@c-69-250-52-161.hsd1.md.comcast.net) |
| 19:29.57 | [TK]D-Fender | Joe_CoT: Yes, and maybe you DON'T want your listeners to fuck with each others ability to LISTEN to your annoucer <- |
| 19:31.01 | Joe_CoT | yes, and that's what listen-only is for. But in a world where listen-only already exists, what point does talk-only have? |
| 19:31.57 | *** join/#asterisk QubeZ (~nkasu@64.128.254.34) |
| 19:32.00 | QubeZ | hello all |
| 19:32.06 | *** part/#asterisk jaxyeh (~jaxyeh@c-69-250-52-161.hsd1.md.comcast.net) |
| 19:32.08 | *** join/#asterisk jaxyeh (~jaxyeh@c-69-250-52-161.hsd1.md.comcast.net) |
| 19:32.19 | Qwell | Joe_CoT: that would require that all people in the conference are set to listen-only |
| 19:32.50 | [TK]D-Fender | Joe_CoT: When the talker is a public speaker who who talking AT the audience, not WITH them. |
| 19:32.56 | QubeZ | I'm having a strange issue where dtmf is not working on my Gigabit ports Cisco 6509. Same configuration (mode access, access vlan 10) as my fastethernet. When I switch to gigabit, all tcp/udp traffic works except the sip dtmf. When I switch back, everything works again. |
| 19:32.59 | *** join/#asterisk Yoe (~wouter@samba.grep.be) |
| 19:33.00 | QubeZ | Any ideas? |
| 19:33.12 | [TK]D-Fender | Joe_CoT: Called "I'm here to tell you something and not give ashit what you have to say. That's why I'm the speaker" |
| 19:33.22 | [TK]D-Fender | Joe_CoT: You seem to lack imagination |
| 19:33.29 | Yoe | hi -- I've migrated an asterisk setup from one host to another host |
| 19:33.35 | QubeZ | scowered google but no help in troubleshooting this issue |
| 19:33.49 | [TK]D-Fender | Joe_CoT: No heckling from the audience, either to the speaker or to the other listeners. |
| 19:34.00 | Joe_CoT | [TK]D-Fender, yes, but in what scenario would you want to tell them something and not listen to them, but still want them to talk to each other (ie talk-only without them listen-only)? |
| 19:34.14 | Yoe | the old host was a 32bit Debian installation, the new one is an x86_64 one of the same Debian release; I've reinstalled, but copied the files over |
| 19:35.06 | Yoe | now inbound SIP dialing no longer works. When I say 'sip set debug peer <peer>', it looks like it fails with '481 Call leg/transaction does not exist' |
| 19:35.13 | [TK]D-Fender | Joe_CoT: When you want them to listen to a fixed source without feedback, extra distractions, etc |
| 19:35.31 | Yoe | when a call comes in from a Linksys ATA on the same context, that does work |
| 19:35.35 | Yoe | any ideas? |
| 19:43.31 | *** join/#asterisk sezuan (~sezuan@mobil5.vp.ip6.scheff32.de) |
| 19:45.31 | sezuan | what are the requirements for a packet2packet bridge? I've a case where I expect a bridge, but it doesn't happen. Both ends use alaw as codec with 20ms packetization. |
| 19:46.31 | *** join/#asterisk jsidhu (~js@173-8-149-45-SFBA.hfc.comcastbusiness.net) |
| 19:50.16 | jsidhu | im trying to setup a sip trunk.. i'm setting the context to from-pstn which is defined as shown in http://pastebin.com/k0YYcZnN. When i dial my DID, I see a bunch of messages generated (sip debug enabled). My extension 601 doesn't ring... Can someone tell me what im doing wrong? |
| 19:50.56 | Yoe | ah, hrm |
| 19:51.03 | *** join/#asterisk mountainm2k (~mountainm@gw.booyahnetworks.com) |
| 19:51.24 | Yoe | the registration bit says "Contact: asterisk@<ip>" instead of "Contact: s@<ip>" |
| 19:51.54 | Yoe | how would I fix that? |
| 19:52.18 | *** join/#asterisk Micc (~quassel@c-98-225-57-96.hsd1.wa.comcast.net) |
| 19:52.59 | mountainm2k | Question: In a macro, how would I do something like: if ${ARG3} Playback(${ARG3} |
| 19:53.01 | Micc | Anyone have experience with audiocodes adapters? I've got one that keeps trying to register a user 00000003 |
| 19:53.05 | [TK]D-Fender | jsidhu: I see no peer definition there and no call with SIP DEBUG proving that it is matching a peer or showing where it is going |
| 19:54.40 | wcselby | mountainm2k - s,n,GoToIf($[${ARG3}]?playback3) s,n(playback3),Playback(${ARG3}) |
| 19:54.43 | wcselby | or something there abouts |
| 19:56.07 | SiNGLer | Micc: check authentication section, or general settings what type of autentication you selected, if "per endpoint" you should define username for each port |
| 19:56.10 | mountainm2k | wcselby: there's no way to do it in one line? Seems like there outta be... |
| 19:56.31 | *** join/#asterisk evil_gordita (~evilgordi@ip70-188-50-186.rn.hr.cox.net) |
| 19:56.48 | wcselby | mountainm2k - you can use the ${IF()} function to do it in one line, I'm sure. |
| 19:58.13 | [TK]D-Fender | mountainm2k: core show application execif |
| 19:59.07 | jsidhu | [TK]D-Fender: http://pastebin.com/UCNw5Eij updated with sip peer definition (sip.conf) |
| 19:59.10 | *** join/#asterisk sarah_p (~sarah_p@71.117.223.226) |
| 19:59.37 | [TK]D-Fender | jsidhu: I do not see the FAILED CALL |
| 19:59.50 | jsidhu | oh sorry, let me paste the sip debug output |
| 20:02.09 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 20:02.37 | jsidhu | [TK]D-Fender: http://pastebin.com/mdsUQb3J |
| 20:03.45 | [TK]D-Fender | jsidhu: Is that EVERYTHING? |
| 20:04.28 | jsidhu | yup.. thats everything i see on the console |
| 20:04.39 | jsidhu | core set verbose 99 & sip set debug |
| 20:05.22 | [TK]D-Fender | jsidhu: Add "insecure=port,invite" to you trunk def, apply, and call again |
| 20:05.33 | *** join/#asterisk bent_screwdriver (~socain00@74.255.249.66) |
| 20:05.34 | jsidhu | [TK]D-Fender: i've tried that already |
| 20:05.41 | [TK]D-Fender | jsidhu: DO IT |
| 20:05.53 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 20:05.55 | jsidhu | shit.. i had a typo |
| 20:05.56 | jsidhu | insecture |
| 20:06.18 | bent_screwdriver | if i need to upgrade dahdi does asterisk need to be reconfigured recompiled? |
| 20:06.49 | [TK]D-Fender | bent_screwdriver: Generally no |
| 20:07.01 | [TK]D-Fender | bent_screwdriver: if its a lower fractional upgrade |
| 20:07.23 | bent_screwdriver | going from 2.2.1.1 to 2.3.0.1 |
| 20:07.34 | [TK]D-Fender | bent_screwdriver: I doubt it, but maybe |
| 20:07.40 | bent_screwdriver | thx TK |
| 20:08.16 | jsidhu | [TK]D-Fender: chan_sip.c:14703 handle_request_invite: Call from '77e22898' to extension '4055551213' rejected because extension not found. |
| 20:08.24 | jsidhu | thats better i guess |
| 20:08.37 | jsidhu | thanks man, didnt catch that typo |
| 20:09.39 | JamesHarrison | I'm getting this error when trying to handle incoming calls from a PSTN-connected SIP trunk: username mismatch, have <siptrunk>, digest has <6357a278c3> Failed to authenticate device "somepstnnumber"<sip:somepstnnumber@88.208.249.40>;tag=as68c0fcff |
| 20:11.12 | [TK]D-Fender | jsidhu: Ned, exten => s,n,Set(cNum = ${SIP_HEADER(TO):6:10}) <- this line is BROKEN. The extra spaces are bad. |
| 20:11.15 | [TK]D-Fender | Next* |
| 20:11.43 | [TK]D-Fender | jsidhu: And well.... you're now showing us context's FreePBX own. You should not be screwing around in there. Go make a DID for this |
| 20:12.35 | jsidhu | >? |
| 20:12.39 | jsidhu | removed the spaces |
| 20:12.47 | *** join/#asterisk neurosys (~neurosys@173-9-159-177-miami.txt.hfc.comcastbusiness.net) |
| 20:13.09 | [TK]D-Fender | jsidhu: Now go set it up properly |
| 20:13.24 | jsidhu | uhhh.. ok |
| 20:14.44 | *** join/#asterisk cjk (~cjk@vodsl-8936.vo.lu) |
| 20:15.32 | cjk | hi, i have troubles with my xorcom fxs bank and dahdi. i do not see the spans in /proc/dahdi but the firmware seems to be loaded. any idea? |
| 20:15.52 | *** join/#asterisk jesselang (~jesse@c-24-131-130-197.hsd1.mn.comcast.net) |
| 20:17.00 | jsidhu | [TK]D-Fender: can yuo give me a hint? |
| 20:17.11 | [TK]D-Fender | [16:11]<[TK]D-Fender>jsidhu: And well.... you're now showing us context's FreePBX own. You should not be screwing around in there. Go make a DID for this |
| 20:17.38 | jsidhu | i do have DID's.. and which context am i making up? from-pstn? |
| 20:17.43 | paulc | Load testing: Let's say I have an IVR type system that uses CURL to talk to a back end, passing in user digits, getting back what to do next, and I want to load test it. What tool can I use? "sipp" seems to manage sending the calls in and can play audio (so I receive audio back - asterisk's RTP "one in one out" thing right?).. I don't necessarily need to press digits, I can just randomise my back end piece |
| 20:17.50 | paulc | would "sipp" be the right tool for the job? |
| 20:18.08 | *** join/#asterisk eliel (~eliels@201.234.94.226) |
| 20:18.57 | [TK]D-Fender | jsidhu: As of now you stop screwing with CONFIG FILES and use FreePBX to set up the DID properly <- |
| 20:19.25 | jsidhu | [TK]D-Fender: i dont have freepbx. I am configuring on a OpenWRT system |
| 20:19.39 | [TK]D-Fender | jsidhu: Odd your using their contexts and otehr conventions. |
| 20:19.52 | [TK]D-Fender | jsidhu: then fix your exten. "s" will not match that NUMEBR <- |
| 20:19.55 | [TK]D-Fender | NUMBER |
| 20:19.57 | jsidhu | ah |
| 20:20.02 | jsidhu | thanks, will look |
| 20:25.08 | *** part/#asterisk mountainm2k (~mountainm@gw.booyahnetworks.com) |
| 20:27.40 | [TK]D-Fender | checkout time, later all |
| 20:32.08 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 20:32.53 | jesselang | Do you have to have a dchan per PRI? Can you set up a primary/secondary dchan to be used across a 4 span card? |
| 20:33.32 | WIMPy | yes and yes |
| 20:33.44 | tzafrir_laptop | cjk, still having problems? |
| 20:34.04 | tzafrir_laptop | what's the output of dahdi_hardware -v #? |
| 20:34.10 | jesselang | WIMPy, could you expound? Those seem like exclusive statements to me. |
| 20:34.46 | jesselang | WIMPy, can you point me to any resources about how to configure Zaptel in such a way? |
| 20:35.27 | WIMPy | No. You definitely have one D channel per pri, but depending on configuration only one might be used for multiple interfaces. In that case fail-over would be available. |
| 20:35.29 | *** join/#asterisk fcolista (~quassel@93-63-253-26.ip30.fastwebnet.it) |
| 20:36.01 | *** join/#asterisk bn-7bc (bjarne@sttun-1-pt.tunnel.tserv24.sto1.ipv6.he.net) |
| 20:36.01 | WIMPy | There isn't anything more than the example configuration, I think. |
| 20:37.24 | *** join/#asterisk markfeatherston_ (~mark.feat@71.4.41.98.ptr.us.xo.net) |
| 20:37.58 | jesselang | Maybe what I need to ask is "do you have to have a dchan per span?" Can you use 2 dchan for 4 spans? |
| 20:39.02 | jesselang | Is it as simple as specifying which channels you want to be dchan? Right now, the last channel of each span is a dchan. |
| 20:39.20 | WIMPy | No, you can have multiple PRIs in one span. |
| 20:39.30 | *** join/#asterisk iscario (~quassel@laureades.davout02.net1.nerim.net) |
| 20:40.27 | WIMPy | Oh, T1, well in that case you could do without D channel on additional line, giving you one more B channel. |
| 20:42.24 | jesselang | WIMPy, I'm looking to have two D channels serve 4 circuits, giving 94 available channels, with a primary and secondary data stream for those channels. |
| 20:42.55 | jesselang | All the examples I see in zaptel.conf show 1 D channel for every 24 channel circuit. |
| 20:44.16 | Katty | hello my asterisk does nto work at all how to fix pls |
| 20:44.39 | WIMPy | spanmap is the keyword, IIRC. |
| 20:45.10 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 20:45.16 | jesselang | WIMPy, I'll check it out, thanks. |
| 20:45.16 | Katty | fenderbender |
| 20:47.24 | [TK]D-Fender | KutyKittyKatty |
| 20:47.30 | *** part/#asterisk bn-7bc (bjarne@sttun-1-pt.tunnel.tserv24.sto1.ipv6.he.net) |
| 20:48.33 | *** join/#asterisk bn-7bc (bjarne@raskerepc.holmedal.net) |
| 20:49.40 | boodu | hello |
| 20:54.51 | Naikrovek | jeepers apache ticks me off sometimes |
| 20:55.19 | markfeatherston_ | how so? |
| 20:57.54 | [TK]D-Fender | mod_checkbox |
| 20:58.15 | *** join/#asterisk Benwa (~Benwa@86.114-200-80.adsl-dyn.isp.belgacom.be) |
| 20:58.57 | Naikrovek | trying to set up a reverse proxy |
| 20:59.21 | Naikrovek | have a virtual host i'm trying to add a reverse proxy to |
| 20:59.36 | Naikrovek | this server must serve requests from outside the intarnets |
| 20:59.56 | Naikrovek | but it needs to fwd to an internal-only app server to do so |
| 21:00.08 | Naikrovek | and it's barfing on standard config items |
| 21:00.33 | Naikrovek | so i run apache2 -M to see what modules are loaded and that gives me some BS about an environment variable that isn't set (but it is set) |
| 21:00.49 | Naikrovek | whole thing is making me want to murder countless innocent apache developers |
| 21:01.24 | jesselang | [TK]D-Fender, lol |
| 21:02.02 | *** join/#asterisk fofware (fabian@host122.190-31-226.telecom.net.ar) |
| 21:03.40 | *** join/#asterisk ruied (~ruied@bl7-218-14.dsl.telepac.pt) |
| 21:05.37 | *** join/#asterisk jnicola (~jnicola@186.124.123.2) |
| 21:05.42 | jnicola | hello |
| 21:06.12 | jnicola | i get res_fax.c:685 set_fax_t38_caps: channel 'SIP/XXX-00000002' is in an unsupported T.38 negotiation state, cannot continue. |
| 21:06.29 | jnicola | i can't receive fax |
| 21:06.54 | Katty | fenderbender |
| 21:06.55 | Katty | hello my asterisk does nto work at all how to fix pls |
| 21:07.09 | beek | hugs Katty |
| 21:07.18 | jnicola | any idea about res_fax ? |
| 21:07.21 | beek | waves to the rest of the group |
| 21:10.56 | jsidhu | ive got one way audio.. The caller cant hear me... I dont even hear the ring.. how do i go about troubleshooting this? The trunk provider says RTP is on ports 10000-20000 and i've got a rule that allows those ports in... but no audio |
| 21:12.04 | [TK]D-Fender | ~sipnat |
| 21:12.05 | infobot | i heard sipnat is Quick guide on configuring Asterisk + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 21:12.09 | [TK]D-Fender | ^^^ |
| 21:12.19 | carrar | ~slipknot |
| 21:12.20 | infobot | slipknot is, like, a very shitty band |
| 21:12.25 | jsidhu | [TK]D-Fender: this isn't really behind a nat.. my router is my asterisk box |
| 21:12.26 | carrar | haha |
| 21:12.58 | [TK]D-Fender | jsidhu: other important bits in there <- |
| 21:14.19 | *** join/#asterisk ivanv (6d6af47e@gateway/web/freenode/ip.109.106.244.126) |
| 21:15.17 | ivanv | How can I force * peers to register on port 5060? |
| 21:16.38 | [TK]D-Fender | ivanv: Source port is irrelevent |
| 21:16.59 | [TK]D-Fender | ivanv: pers will reg to whatever port you point them to |
| 21:17.06 | [TK]D-Fender | ivanv: Which will be default be 5060 |
| 21:17.24 | ivanv | but I want to force peers to 5060 |
| 21:17.56 | ivanv | I tried to put port=5060 in sip.conf in peer parameters |
| 21:18.39 | [TK]D-Fender | ivanv: * doesn't tell them where to contact *. tis the opther way aeround |
| 21:18.58 | [TK]D-Fender | ivanv: And if * gts anything from them at all then its already going where it is supposed to. |
| 21:20.43 | ivanv | you mean * accept the port from peer? Does it really mater at all? |
| 21:23.32 | [TK]D-Fender | ivanv: The source doesn't matter <- |
| 21:25.08 | ivanv | ok, soory, just one more question, port=5060 in sip.conf in peer parameters is prefered peer listening port or what? |
| 21:25.26 | Katty | hugs beek |
| 21:25.35 | Katty | i need a new trolley statement. |
| 21:29.40 | markfeatherston_ | ivanv: port 5060 is what the server listens on for remote connections. The sip clients will connect to 5060 on whatever source port they want, but the destination will always be what you set for "port=" |
| 21:32.28 | [TK]D-Fender | ivanv: Where to send calls to that peer |
| 21:36.51 | ivanv | another issue, how can I hide mysql user/pass visible in /etc/asterisk/cdr_mysql.conf |
| 21:37.50 | markfeatherston_ | as long as it's only readable by asterisk it should be fine |
| 21:38.22 | markfeatherston_ | if you're concerned about security, you should probably read this though |
| 21:38.23 | markfeatherston_ | http://svn.digium.com/svn/asterisk/trunk/README-SERIOUSLY.bestpractices.txt |
| 21:40.11 | *** join/#asterisk drclue (~drclue@ip-65-49-163-54.wireless.dyn.beamspeed.net) |
| 21:40.33 | *** join/#asterisk sahafeez (~sahafeez@65-119-47-100.dia.static.qwest.net) |
| 21:43.48 | drclue | OK, I'm going to try this question another way. Let's say I have a soft-phone running , but I want to use AMI to originate the call and use AMI to do things like navigate IVR, using the soft-phone as just a place for the headphone and microphone to interact. What would be the best way to do that? |
| 21:43.59 | *** join/#asterisk Blackvel (~blackvel@dslb-088-065-099-102.pools.arcor-ip.net) |
| 21:44.56 | Blackvel | what was the name of the application to let the caller record his name and callingreason and then let the caller party decide by pressing 1/2 to accept or deny the call (playing the recorded name) |
| 21:44.57 | markfeatherston_ | drclue: I don't think there's going to be a very elegant way, but you should be able to just originate a call and then use dtmf to control the ivr by timing it |
| 21:44.59 | jsidhu | ...still trying to solve my audio problem. i've got rtp debug enabled, and I only see statements such as "Got RTP packet from ...." and I dont see anything resembling "Sent RTP packet to..." i should see a mix of got RTP packets and SENT RTP packets right? what should i look at? my asterisk box is not behind nat, and am trying to get a SIP Trunk to work |
| 21:45.14 | Blackvel | and then let the called party... |
| 21:45.20 | [TK]D-Fender | drclue: AMI doesn't "navigate" anything |
| 21:45.41 | [TK]D-Fender | drclue: AMI is not for controlling the full flow of a call. It if for general PBX manipulations |
| 21:46.17 | [TK]D-Fender | Blackvel: There is no application. YOU implement this is a macro called from Dial |
| 21:46.27 | [TK]D-Fender | Blackvel: "core show application dial" |
| 21:46.41 | Blackvel | i thought there was...nevermind |
| 21:47.07 | [TK]D-Fender | jsidhu: "canreinvite=no" <----- supremely important |
| 21:47.12 | markfeatherston_ | Blackvel: there are examples of a privacy manager out there. I think freepbx has it built in |
| 21:47.54 | [TK]D-Fender | This is less than 10 lines of dialplan...... |
| 21:49.04 | drclue | @markfeatherston_,@fender : Well , I want to try and figure a way through this. I've tried playDTMF on the active channel of the call , but it makes for a blocking error in the dtmf log, and but a couple 100ms of silence in the audio stream. |
| 21:50.30 | Blackvel | markfeatherston_: i implemented my own privacy manager for entering the callerid already :) but now this is different usecase..ok will have to check how to connect the two channels by hand and not by dial automatically |
| 21:50.41 | [TK]D-Fender | drclue: AMI does not sit like a nanny to supervise the call. |
| 21:50.51 | [TK]D-Fender | drclue: thats what your DIALPLAN is for |
| 21:51.03 | Blackvel | [TK]D-Fender will probably tell me too look for the correct dial options, don't you? :) |
| 21:51.23 | *** join/#asterisk Benwa (~Benwa@86.114-200-80.adsl-dyn.isp.belgacom.be) |
| 21:51.27 | [TK]D-Fender | [17:46]<[TK]D-Fender>Blackvel: "core show application dial" <- RTFM :) |
| 21:51.40 | Blackvel | right right...i'll go my way |
| 21:51.59 | Blackvel | thanks for brainstorming or giving me assurance to have to go the "manual way" |
| 21:53.26 | *** join/#asterisk arpu (~arpu@chello062178159144.10.14.univie.teleweb.at) |
| 21:54.39 | drclue | @fender: OK , then maybe I'm mis-understanding the purpose of the AMI PlayDTMF command. What is that command actually for? |
| 21:55.32 | [TK]D-Fender | drvYES that command CAN inject DTMF to a call but you have it in your head the you are going to use AMI to hand-hold an ENTIRE CALL |
| 21:55.37 | [TK]D-Fender | drclue: ^ |
| 21:55.57 | [TK]D-Fender | drclue: THAT is the mistake. Just use the DILPLAN you point it to to do that and trigger whatever else you need |
| 21:56.11 | markfeatherston_ | How would I send out a sip notify when record-on-demand is activated on a channel? |
| 21:57.05 | [TK]D-Fender | is off for a bit |
| 22:00.20 | markfeatherston_ | I swear, every time I ask that question it immediately gets quiet in the channel |
| 22:00.51 | drclue | @fender , This is a HTML5 WebSockets interface to AMI, and would have no advance knowledge of any particular dialplan in play on the active channels communicating with an IVR at a remote location , possibly on another machine |
| 22:02.40 | *** join/#asterisk RobH (~robh@wikimedia/RobH) |
| 22:04.07 | *** join/#asterisk MuskyMalamute (vintar@unaffiliated/coil) |
| 22:04.14 | MuskyMalamute | how do i spoof my phone number |
| 22:04.16 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 22:04.29 | *** join/#asterisk citywok (~chatzilla@67-134-194-33.dia.static.qwest.net) |
| 22:06.53 | Qwell | MuskyMalamute: short answer; you don't. |
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| 22:07.36 | drclue | @fender Now since say that PlayDTMF *can* inject a DTMF into an active channel, that is indeed what I want to do in this instance, so I need to clear up the details of making that happen |
| 22:08.13 | markfeatherston_ | drclue: PlayDTMF should work. Are you certain DTMF works in other cases with this system? |
| 22:08.45 | drclue | Nope , I am not sure of that , and some underlying issue with configuration may indeed be the issue. |
| 22:09.12 | markfeatherston_ | Test that first |
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| 22:10.57 | drclue | I get a couple 100ms of silence in the audio when I PlayDTMF from AMI and a notation in the dtmf log of ([Jul 5 07:45:27] DEBUG[3048] channel.c: Thread -1227400336 Blocking 'SIP/4096-00000000', already blocked by thread -1227646096 in procedure ast_waitfor_nandfds |
| 22:10.58 | drclue | ) |
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| 22:13.47 | markfeatherston_ | drclue: are you sure the call is active at the point you're sending dtmf? |
| 22:14.32 | *** join/#asterisk simplydrew (~simplydre@pool-173-69-4-91.prvdri.fios.verizon.net) |
| 22:14.55 | drclue | Well, at the time I try and send the DTMF to the active channel , I'm listening to the call with the soft-phone associated with that channel |
| 22:15.45 | markfeatherston_ | http://www.voip-info.org/wiki/view/Asterisk+DTMF |
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| 22:16.03 | markfeatherston_ | test if it's working and see if it matches any issues there |
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| 22:31.35 | bmoraca_work | lindsey lohan's going to jale |
| 22:31.46 | Qwell | SHOCKING |
| 22:31.53 | bmoraca_work | wow |
| 22:31.57 | bmoraca_work | me be gud sperring |
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| 22:43.40 | drclue | @markfeatherston OK , I've tested DTMF via the dialplan and no tones are audible. I do see the SendDTMF command being executed in CLI, and the Soft-Phone's text monitor displays DTMF messages, but , no sounds |
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| 22:57.56 | Sheeplet | lo all |
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| 22:59.19 | bvector | hey guys, I finally convinced my boss to migrate to Trixbox CE to save some money over Trixbox call center edition, and the only gripe he has is that Trixbox Pro allows you to set alerts when you're utilizing more than a set number of lines |
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| 22:59.41 | bvector | anyone know how to do that in CE or can point me to the right place? I've looked all over and I must be wording my searches wrong |
| 23:00.48 | WIMPy | ~trixbox |
| 23:00.49 | infobot | somebody said trixbox was SH1TB0X. Basically a CRAPPY, closed-source distro. STAY AWAY! |
| 23:01.02 | markfeatherston_ | hahaha |
| 23:01.17 | bvector | Trixbox CE is opensource |
| 23:01.29 | bvector | but still kinda sh1tb0xy |
| 23:01.39 | bvector | but babysteps haha |
| 23:02.29 | drclue | I always love distro wars |
| 23:02.45 | bvector | yeah me neither |
| 23:03.09 | bvector | but I'm really interested to see if there is an easy way to monitor line usage, we only have 8 analog lines and 12 employees |
| 23:04.02 | drclue | I just use asterisk by itself, along with an HTML5 tool I've been developing called AsterClick |
| 23:05.41 | drclue | I can already see all the events Aster kicks out as XML and it's pretty painless to format that as HTML |
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| 23:17.24 | drclue | @markfeatherston OK, I've looked over the various items at http://www.voip-info.org/wiki/view/Asterisk+DTMF , but I've still not been able to make any tones via SendDTMF in the dialplan or PlayDTMF, in AMI so I'm sorta stumped. |
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| 23:24.55 | ChannelZ | drclue: The actual DTMF sounds would probably only be heard depending on the flavors of channels involved. For instance if everyone is rfc2833 it's essentially just metadata being transmitted. |
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| 23:27.09 | drclue | @ChannelZ OK, that sorta makes sense as I do see DTMF messages in the soft-phone. |
| 23:27.09 | drclue | So the trick that I'm trying to do is to get AMI to inject a DTMF into an active channel that the IVR will actually perceive |
| 23:28.39 | ChannelZ | as if the user pressed it on their channel -- that's maybe a trick no matter what |
| 23:29.54 | ChannelZ | (I'm assuming by 'the IVR' you're talking about Asterisk, the same server you're trying to do things via AMI on and not a remote server?) |
| 23:31.20 | drclue | @ChannelZ I know it's a trick, but it would be sorta useful. What I'm doing is that I've written a really cool HTML5 WebSockets middleware that allows allows a web page to manipulate AMI through perpetually connected sockets (Far cooler than polling) and one of the applications of this would be sorta a HUD kind of thing that works alongside a soft-phone or other phone |
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| 23:33.49 | drclue | If your curious as to what it's about , I've tossed up a web page about it at http://asterclick.drclue.net/ |
| 23:35.42 | drclue | In the current development situation the IVR is indeed on the same server as the web application, but that would not always be the case |
| 23:37.29 | ChannelZ | Yah - I think you are in 'hack the source' territory, I don't think there's any built-in means to inject DTMF on the 'incoming' side of a channel. |
| 23:39.57 | drclue | Well, the call is originated by the AMI interface if that makes any difference. Would there perhaps be a way to rig another channel I could initiate DTMF on and perhaps bridge or something? |
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| 23:43.38 | drclue | It would not be practical to "hack-the-source" as whatever happens would need to be something that my middle-ware could cause to happen based upon the stock asterisk interfaces whereevr it was installed |
| 23:53.28 | paulc | Load testing: Let's say I have an IVR type system that uses CURL to talk to a back end, passing in user digits, getting back what to do next, and I want to load test it. What tool can I use? "sipp" seems to manage sending the calls in and can play audio (so I receive audio back - asterisk's RTP "one in one out" thing right?).. I don't necessarily need to press digits, I can just randomise my back end piece |
| 23:53.35 | paulc | would "sipp" be the right tool for the job? |
| 23:58.16 | pabelanger-lap | paulc: no sipp does not handle RTP. |
| 23:58.59 | pabelanger-lap | paulc: http://svn.digium.com/svn/testsuite/asterisk/trunk/ |
| 23:59.16 | pabelanger-lap | some examples for automated tests |
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