| 00:06.57 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
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| 00:27.29 | *** join/#asterisk sequencer (~something@196.218.255.29) |
| 00:27.33 | sequencer | hi all\ |
| 00:28.21 | sequencer | i want to get call information ( at least caller ID ) from asterisk using PHP, any where to start from that any one know s of ? |
| 00:30.37 | *** join/#asterisk Cesare (~Adium@creati59.lnk.telstra.net) |
| 00:35.06 | russellb | use the asterisk manager interface |
| 00:35.24 | sequencer | any help on where is this located? |
| 00:35.24 | russellb | there is a chapter that provides an intro to what the AMI is in this book: |
| 00:35.27 | russellb | ~thebook |
| 00:35.27 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342. Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/, or ~buybook |
| 00:35.45 | sequencer | thanks! |
| 00:35.47 | russellb | np |
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| 01:55.09 | sequencer | hi again |
| 01:55.40 | sequencer | am having a hard time with scripting , i am trying to "catch" a caller id whenever a call comes in to an ext. |
| 01:56.16 | sequencer | am using fpbx and i believe what am trying to do is already done on flash operator panel |
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| 02:38.11 | ruben23 | hi guys anyone have idea how to configure 00h323.conf--add extension/trunk on it.. |
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| 02:45.59 | *** mode/#asterisk [+o russellb] by ChanServ |
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| 02:46.30 | *** mode/#asterisk [+o russellb] by ChanServ |
| 02:48.03 | carrar | h4X0r |
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| 02:50.29 | *** mode/#asterisk [+o russellb] by ChanServ |
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| 02:51.16 | *** mode/#asterisk [+o russellb] by ChanServ |
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| 03:09.57 | ruben23 | hi |
| 03:10.50 | *** join/#asterisk WiretapWork (~Wiretap@unaffiliated/wiretap) |
| 03:11.38 | WiretapWork | quick question, installing asterisk to a specific directory so I can easily zip all the files up and turn them into a package, how do I do it? |
| 03:11.47 | WiretapWork | (working with 1.8.3.3 |
| 03:11.48 | WiretapWork | ) |
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| 03:36.18 | carrar | ./configure --help |
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| 03:43.38 | *** part/#asterisk squidly (~squidly@HoodLUG/member/squidly) |
| 03:46.09 | *** join/#asterisk WiretapWork_ (~Wiretap@unaffiliated/wiretap) |
| 03:46.27 | WiretapWork_ | carrar, that doesn't give me any way to supply a 'virtual root' path for the make install command |
| 03:46.52 | *** join/#asterisk joobie (~joobie@unaffiliated/moo0o0ooo00o0o0o) |
| 03:47.12 | joobie | hey guys.. anyone know of a good 12 port POE switch? cheap preferable :P |
| 03:52.09 | carrar | capture it |
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| 03:53.29 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 03:54.54 | WiretapWork_ | carrar, even easier it turned out to be.... use 'checkinstall' |
| 03:55.01 | WiretapWork_ | kills two birds with one stone |
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| 03:57.23 | carrar | spiffy |
| 04:06.49 | WiretapWork_ | I was having to cross-compile asterisk for i486/i586 on an i686 box that I don't want asterisk on, because I think my poor VIA C3 powered pbx would choke and die trying to compile it :P |
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| 04:58.36 | WiretapWork_ | not going to pretend I understand that |
| 04:58.55 | WiretapWork_ | my i486 compiled package is 7.5MB, the same config compiled for i686 is 23MB |
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| 04:59.52 | WiretapWork_ | no, actually I just worked it out, forgot to deselect the MOH |
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| 05:29.33 | joobie | guys where can i get check_nrpe from? it's not apart of the nagios-plugins tarbll |
| 05:40.15 | carrar | http://www.nagios.org/download/addons |
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| 06:22.43 | schmidts | good morning |
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| 06:27.45 | ChannelZ | ahoyhoy |
| 06:32.30 | wdoekes2 | morning\ |
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| 06:44.31 | ferdna | should i put my phone extensions in sip.conf or in extensions.conf? |
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| 06:50.59 | AliRezaTaleghani | hi |
| 06:51.19 | AliRezaTaleghani | can somebody give a help about voicemail application |
| 06:51.51 | AliRezaTaleghani | i am looking for a way to use channel variables in voicemail (email body) notifications |
| 06:52.27 | AliRezaTaleghani | is any body home had experience about it? |
| 06:53.35 | ChannelZ | that'd probably have to be a sourcecode hack |
| 06:54.10 | ChannelZ | or.. I wonder.. (looks) |
| 06:56.36 | joobie | guys can anyone recommend a decent quality cordless voip phone? |
| 07:00.02 | kleszcz | morning |
| 07:04.40 | Wiretap7 | I have a strange problem with voicemail |
| 07:04.50 | Wiretap7 | when I install the digium compiled debian package, it works fine |
| 07:04.54 | Wiretap7 | when I install the one I compiled myself |
| 07:05.21 | Wiretap7 | dialling *97 results in a few seconds of nothing and then a disconnect, people who would normally call to VM get reorder tone |
| 07:05.42 | Wiretap7 | and *98 results in nothing, and it having a fit in verbose |
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| 07:09.19 | Wiretap7 | anyone have any ideas? |
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| 07:28.20 | jacc0 | gooooooooog morning |
| 07:28.24 | jacc0 | :) |
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| 07:29.32 | jacc0 | is everything at digium back to normal again? |
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| 07:55.46 | verywiseman | suddenly i can't make call from zap channel 34 , when i write this command `zap show channel 34` , owner , real parameter appear "owner=zap/34-1 , real=zap/34-1" , and some times it is appearing "owner=<none>,real=<none>" , also "Echo Cancellation:" sometimes appear "128 taps unless TDM bridged, currently OFF" and sometimes "128 taps unless TDM bridged, currently ON" , where is problem? |
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| 09:09.30 | ferdna | can anyone explain to me the context in sip.conf |
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| 09:09.51 | wdoekes2 | that's where in the dialplan incoming calls land |
| 09:10.41 | wdoekes2 | (or "outgoing" if you view it from the sip-user's perspective) |
| 09:11.26 | wdoekes2 | sip.conf: [joe] context=phones |
| 09:11.45 | wdoekes2 | extensions.conf: [phones] exten => 200,1,Dial(SIP/lucy) |
| 09:12.04 | wdoekes2 | now joe can dial 200 to call lucy |
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| 09:18.48 | aberrios | I think of it as: if you've setup a sip phone in sip.conf its where that phone starts in the dialplan (extensions.conf) when they dial anything. |
| 09:21.35 | ferdna | wdoekes2, ok... i am confused |
| 09:21.36 | ferdna | with |
| 09:22.05 | ferdna | i have two itsp |
| 09:22.24 | ferdna | [itsp1] & [itsp2] in my sip.conf |
| 09:22.34 | ferdna | and they both have context=default |
| 09:22.48 | ferdna | in my sip.conf in the [general] context = default |
| 09:22.51 | ferdna | is this correct? |
| 09:23.22 | ferdna | aberrios, is this correct? |
| 09:23.24 | wdoekes2 | if general already has context=default, the two itsp peers needn't have it |
| 09:23.33 | ferdna | i see |
| 09:23.35 | ferdna | now... |
| 09:23.42 | ferdna | in my extensions.conf |
| 09:23.49 | ferdna | what should i have for my [default] |
| 09:24.07 | wdoekes2 | but you probably want put the [itspN] contexts to context=incoming instead |
| 09:24.19 | ferdna | why? |
| 09:24.21 | wdoekes2 | and create a separate [incoming] in your dialplan |
| 09:24.35 | ferdna | explain that to me please =) |
| 09:24.58 | wdoekes2 | this mitigates broken configs where the outside dials out, with your money |
| 09:25.22 | ferdna | wdoekes2, there will be only one outside dial out account |
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| 09:26.22 | wdoekes2 | I'm not saying you MUST do the [incoming], I'm saying you SHOULD |
| 09:26.48 | wdoekes2 | and you should leave the [default] empty |
| 09:27.03 | ferdna | give me a reason for that |
| 09:27.22 | wdoekes2 | what if I refuse? |
| 09:27.44 | ferdna | lol |
| 09:27.52 | ferdna | i just want to understand "the why" |
| 09:27.57 | wdoekes2 | that's always good |
| 09:28.03 | wdoekes2 | look at the BEST.PRACTICES file |
| 09:28.12 | wdoekes2 | it should be in the root of the asterisk source |
| 09:28.58 | aberrios | it helps to stop incoming calls from the itsp dropping into other contexts you may not want them to be in. |
| 09:29.23 | aberrios | am i right? |
| 09:29.44 | ferdna | oooh ok... |
| 09:31.02 | ferdna | ok |
| 09:31.06 | ferdna | can you guys test my number? |
| 09:31.23 | ferdna | when ever i call i get a buzz and then "the person you call is not available" |
| 09:32.13 | aberrios | does asterisk execute any dialplan when you phone in? |
| 09:32.56 | ferdna | i don't see anything in the cli |
| 09:34.20 | ferdna | aberrios, let me check one more time |
| 09:34.48 | wdoekes2 | ferdna: you need sufficient verbosity (core set verbose 10) |
| 09:34.58 | ferdna | i have 4 |
| 09:35.06 | ferdna | -rvvvv (4 v) |
| 09:35.10 | wdoekes2 | but probably you don't reach any context at all |
| 09:35.26 | wdoekes2 | sip set debug peer itsp1 |
| 09:35.30 | wdoekes2 | (or itsp2) |
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| 09:37.05 | ferdna | one more thing |
| 09:37.11 | ferdna | should i dial like: |
| 09:37.31 | ferdna | exten => _1NXXNXXXXXX, 1, dial(SIP/${EXTEN}@sip.broadvoice.com,30) |
| 09:37.35 | ferdna | or |
| 09:37.49 | ferdna | exten => _1NXXNXXXXXX, 1, dial(SIP/${EXTEN},30) |
| 09:37.51 | ferdna | or |
| 09:38.04 | ferdna | exten => _1NXXNXXXXXX, 1, dial(SIP/${EXTEN}@sip.broadvoice.com,30) |
| 09:38.29 | ferdna | last one i meant: exten => _1NXXNXXXXXX, 1, dial(sip.broadvoice.com/${EXTEN},30) |
| 09:38.55 | wdoekes2 | either Dial(SIP/${EXTEN}@itsp1) or Dial(SIP/itsp1/${EXTEN}) |
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| 09:40.28 | ferdna | wdoekes2, awesome |
| 09:42.33 | ferdna | wdoekes2, aberrios: i get the same thing, |
| 09:42.42 | ferdna | is not available to take your call |
| 09:43.00 | aberrios | do you need to register to the itsp? |
| 09:43.23 | ferdna | aberrios, sip show registry shows 2 sip registrations |
| 09:43.34 | ferdna | and they both say registered |
| 09:44.31 | aberrios | is there a control panel on the itsp services site to show wether they think you're registered? |
| 09:44.39 | ferdna | let me check |
| 09:47.35 | ferdna | nope it doesnt say |
| 09:48.22 | aberrios | can you dial out on it |
| 09:49.28 | ferdna | i can dial extensions |
| 09:49.39 | aberrios | can you dial out over the sip trunk? |
| 09:49.48 | *** join/#asterisk AliRezaTaleghani (~AliRezaTa@unaffiliated/AliRezaTaleghani) |
| 09:49.52 | ferdna | i havent tried yet |
| 09:53.17 | ferdna | aberrios |
| 09:53.38 | ferdna | it says that the device you are using is not registered to place calls on the network |
| 09:54.31 | ferdna | aberrios, wdoekes2: do i need to put a password for outgoing calls or something? |
| 09:55.00 | aberrios | it depends on how your itsp authenticates its you. |
| 09:56.12 | ferdna | would you call me? |
| 09:56.13 | *** join/#asterisk _zoom_ (~zero@216.131.70.133) |
| 09:56.57 | _zoom_ | hi, why asterisk doesnt release RTP ports as soon as the call is ended? |
| 10:00.25 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 10:00.35 | ferdna | aberrios, could it be packets not reaching me? |
| 10:07.18 | *** join/#asterisk wonderworld (~ww@port-92-201-76-199.dynamic.qsc.de) |
| 10:24.55 | aberrios | ferdna: it could be, looks like the itsp is struggling to reach your * box |
| 10:25.11 | ferdna | i think so |
| 10:25.13 | ferdna | :( |
| 10:25.38 | aberrios | is there a firewall between you and the itsp? |
| 10:25.49 | ferdna | yes |
| 10:26.14 | aberrios | ah right, have you changed any settings? |
| 10:27.04 | aberrios | in the firewal |
| 10:27.46 | ferdna | i forwarded some ports to my asterisk box |
| 10:28.03 | ferdna | 69, 5060-5063, 10000-20000 |
| 10:28.55 | WiretapMac | what protocols? |
| 10:29.04 | WiretapMac | and why on earth 69? |
| 10:29.19 | ferdna | UDP & TCP |
| 10:29.25 | ferdna | i know i only need udp |
| 10:31.20 | jacc0 | is everything at digium back to normal again? is the dev. SVN available? |
| 10:31.58 | nextime | 69 cause is a lovable number :) |
| 10:32.53 | ferdna | hehehhe |
| 10:33.03 | ferdna | aberrios, he said something about toll calls... |
| 10:33.23 | aberrios | ferdna: who did? |
| 10:33.33 | ferdna | tech support |
| 10:33.45 | ferdna | WiretapMac, http://www.broadvoice.com/support_faqs_network.html |
| 10:33.56 | ferdna | look for: What ports do I need to open if my BroadVoice Phone Adapter is behind a firewall? |
| 10:34.18 | ferdna | aberrios, anything i have to specify in asterisk? |
| 10:34.21 | WiretapMac | UDP/69 is TFTP |
| 10:34.48 | ferdna | WiretapMac, hehehe i know |
| 10:34.51 | ferdna | but why? |
| 10:34.55 | ferdna | do they want that port open? |
| 10:35.07 | WiretapMac | so that their ATA can grab its configuration data from their server |
| 10:35.16 | aberrios | probably for firmware updates to the "adapter" |
| 10:35.31 | ferdna | o yeah |
| 10:35.46 | aberrios | is natting enabled? |
| 10:36.19 | ferdna | aberrios, how would i know that? |
| 10:36.40 | aberrios | what do you have in the [itspN] contexts in sip.conf |
| 10:37.05 | ferdna | oh you meant the nat=yes |
| 10:37.38 | aberrios | and check your router to see if it's nat is on or not, some routers have a "re-write" SIP Headers setting too. |
| 10:37.52 | ferdna | aberrios, i am following this guide: http://www.broadvoice.com/support_install_asterisk.html |
| 10:38.32 | wdoekes2 | ferdna: what did you put in <extension>? |
| 10:38.40 | wdoekes2 | and do you have that in your [incoming] context? |
| 10:39.12 | *** join/#asterisk Wiretap (~Wiretap@unaffiliated/wiretap) |
| 10:39.47 | ferdna | wdoekes2, in the register command you mean? nothing i left it empty |
| 10:40.38 | wdoekes2 | then it's probably 's'.. so you need an exten => s,1,... in your dialplan |
| 10:40.57 | ferdna | i do have s |
| 10:41.36 | ferdna | exten => s,1,Answer() |
| 10:41.47 | wdoekes2 | okay.. and did you see any sip traffic at all? I don't see mention whether you do or don't |
| 10:42.04 | ferdna | wdoekes2, nope i do see sip traffic |
| 10:42.24 | wdoekes2 | parse error |
| 10:42.34 | ferdna | *I DO see |
| 10:43.13 | wdoekes2 | if you see sip INVITE statements, you should see more info about why it fails to match an incoming call |
| 10:44.20 | ferdna | hmmm |
| 10:44.23 | ferdna | let me paste |
| 10:44.24 | ferdna | bin |
| 10:45.09 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 10:45.11 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 10:46.45 | ferdna | wdoekes2, http://pastebin.com/RA3JJNwF |
| 10:51.13 | *** join/#asterisk MarKsaitis (~MarKsaiti@82.144.242.158) |
| 10:52.36 | aberrios | whats in extensions.conf [phones]? |
| 10:53.18 | ferdna | i have 3 includes |
| 10:53.24 | ferdna | to outgoing_calls |
| 10:53.27 | ferdna | to default |
| 10:53.30 | ferdna | to internal |
| 10:53.47 | ferdna | let me paste |
| 10:57.03 | ferdna | aberrios, http://pastebin.com/dT6wSBQS |
| 10:59.42 | aberrios | dialplan looks fine for now |
| 11:00.10 | aberrios | have you got debug and verbose on and turned up? |
| 11:00.14 | ferdna | yes |
| 11:00.22 | ferdna | i got a different message now |
| 11:00.36 | ferdna | the number that i dial doesnt exist or is invalid |
| 11:00.47 | ferdna | when dialing out of my xlite |
| 11:01.09 | aberrios | and what output from CLI? |
| 11:01.40 | ferdna | aberrios, check this out |
| 11:01.43 | ferdna | what tech support told me |
| 11:02.29 | ferdna | aberrios, http://pastebin.com/SrhpZusE |
| 11:02.36 | ferdna | that is what tech support told me |
| 11:04.24 | WiretapMac | that has nothing to do with * |
| 11:04.37 | aberrios | thats on their control panel for your account with them |
| 11:11.10 | *** join/#asterisk gh0st3r (~gh0st@72.52.127.185) |
| 11:11.49 | gh0st3r | getting this error: [May 10 17:06:17] NOTICE[8063]: chan_sip.c:14441 handle_request_invite: Call from '09400000' to extension '09400000' rejected because extension not found. |
| 11:11.57 | gh0st3r | when i try to call my DID on my SIP line |
| 11:12.10 | gh0st3r | sure its something easy as im new to asterisk |
| 11:14.50 | gh0st3r | heres my configs: http://pastebin.com/6mErmkVT |
| 11:17.24 | ferdna | aberrios, isnt that an indication on what to do so i can get mine working? |
| 11:17.38 | aberrios | maybe, log into their control panel and check their settings |
| 11:18.06 | ferdna | hmmm... they dont have any |
| 11:18.53 | ferdna | aberrios, any other ideas? |
| 11:19.15 | ferdna | 150 registration does that means anything? |
| 11:22.05 | aberrios | ask them what http://pastebin.com/SrhpZusE refers to then, coz as WiretapMac said, thats nothing to do with * |
| 11:22.58 | kaldemar | gh0st3r: not an error, but a notice. your [incoming] in extensions.conf only has exten s, but the call comes in to number 09400000. you need an extension that matches 09400000. |
| 11:23.16 | ferdna | aberrios, xlite |
| 11:24.47 | aberrios | in CLI do a "sip reload" and see if it throws any errors about registration |
| 11:26.33 | ferdna | aberrios, it just complained about: insecure=very |
| 11:26.43 | ferdna | doesnt know what "very" mode is |
| 11:27.13 | *** join/#asterisk ixyd_ (~denzs@carbon.gonicus.de) |
| 11:27.54 | ferdna | aberrios, i changed it to: insecure=port,invite |
| 11:28.45 | gh0st3r | kaldemar, for those same dialing rules to apply, how would i do it for 09400000? |
| 11:28.51 | gh0st3r | to send it to 6001? |
| 11:29.14 | *** join/#asterisk guilhermebr (~Guilherme@200.103.96.98) |
| 11:29.20 | ixyd_ | hi guys, i would like to prevent chan_sip to send out sip-notifies in some cases.... it would be great to control this inside of the dialplan, like by setting a channel variable or something like that.... is this possible in some way? |
| 11:34.23 | gh0st3r | got i going |
| 11:34.24 | gh0st3r | cheers |
| 11:35.31 | *** join/#asterisk AliRezaTaleghani (~AliRezaTa@unaffiliated/AliRezaTaleghani) |
| 11:36.20 | *** part/#asterisk AliRezaTaleghani (~AliRezaTa@unaffiliated/AliRezaTaleghani) |
| 11:38.28 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 11:39.30 | ferdna | aberrios, somehow i made a call |
| 11:39.35 | ferdna | however i heard nothing |
| 11:41.11 | aberrios | ferdna: popping out the office |
| 11:41.17 | ferdna | hey wait |
| 11:41.38 | ferdna | ${EXTEN} send everything... 9 123-4567 |
| 11:41.46 | ferdna | i need to cut off the 9, no? |
| 11:41.52 | ferdna | i rememver something like exten:! |
| 11:41.54 | ferdna | i rememver something like exten:1 |
| 11:41.59 | aberrios | it will send everything |
| 11:42.01 | aberrios | yes like that |
| 11:42.06 | aberrios | gtg :) |
| 11:42.39 | ferdna | take care |
| 11:42.43 | ferdna | and thank you for everything |
| 11:43.52 | asterisk-learner | hi, what are "Comfort Noise frame" AST_FRAME_CNG used for ? |
| 11:44.29 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 11:44.38 | ferdna | aberrios, it is working!!! |
| 11:44.40 | ferdna | i can dial out |
| 11:44.41 | ferdna | !!! |
| 11:55.45 | gh0st3r | i got dialing in working :) :) |
| 11:55.50 | gh0st3r | dtf! |
| 11:57.44 | *** join/#asterisk epart (~TECH-GENS@121.97.63.210) |
| 11:58.04 | *** join/#asterisk z4nD4R (~zandar@icm218139-orange.orange.sk) |
| 11:58.13 | z4nD4R | hi |
| 11:58.39 | ferdna | gh0st3r, awesome |
| 11:59.12 | z4nD4R | pls, how i can play some "message" both callers ? ( something in dialplan... if call is answered.. ) |
| 12:00.07 | wdoekes2 | you want to play a message to both the caller and callee before the call is bridged? |
| 12:01.06 | ferdna | z4nD4R, background command |
| 12:02.07 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 12:02.07 | z4nD4R | wdoekes2: no.. i want to play this message when user pick headset ( call is answered ) and both users can hear this message? |
| 12:02.20 | z4nD4R | ferdna: some dialplan function is not? |
| 12:02.45 | ferdna | yes |
| 12:03.09 | z4nD4R | ferdna: wich? |
| 12:03.21 | ferdna | background |
| 12:04.10 | z4nD4R | ferdna: and function answer ?? |
| 12:04.12 | z4nD4R | exten => s,1,Answer() |
| 12:05.06 | ferdna | z4nD4R, http://pastebin.com/bQiFFKRi |
| 12:06.46 | z4nD4R | ferdna: thx, but i dont know, what command baskroud make... |
| 12:07.08 | ferdna | Background(enter-ext-of-person) or PLAY |
| 12:07.15 | ferdna | those two commands will do what you want |
| 12:07.25 | kaldemar | ferdna: no they won't. |
| 12:08.25 | kaldemar | z4nD4R: there's a built-in option for Dial command to play a message to the called channel upon answer, but not both. |
| 12:08.25 | z4nD4R | exten => s,2,Playback(vm-goodbye) ? |
| 12:09.02 | kaldemar | z4nD4R: you'd need to use G() option in Dial and bridge the channels together afterwards. |
| 12:12.32 | *** join/#asterisk zapotek6 (~zap@mail.comelit.it) |
| 12:12.48 | z4nD4R | kaldemar: if i want to play message both users i need brgide channels.. |
| 12:13.23 | z4nD4R | kaldemar: this message muss not be play in me time... |
| 12:13.30 | z4nD4R | *some |
| 12:19.09 | ferdna | i am going to bed |
| 12:19.12 | sezuan | What's the best way to permanently poll for an incoming jabber message? Like configuring an extentions which receives a jabber messages with a high timeout and calling it again and again. |
| 12:19.12 | ferdna | good night guys |
| 12:20.51 | *** join/#asterisk fhmiv (~fhmiv@c-67-173-205-151.hsd1.ga.comcast.net) |
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| 12:34.30 | *** part/#asterisk Pimmetje (~Pimmetje@s5146b5f9.adsl.wanadoo.nl) |
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| 12:37.22 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 12:43.00 | *** join/#asterisk sideone (~overflow3@mail.traeger.com) |
| 12:44.14 | sideone | hey all.. im starting a new project to try to join multiple softphones together into a conference. i have been researching opensips, freeswitch, and asterisk. Which project can handle 1:many connections? |
| 12:47.18 | *** join/#asterisk marlowe (~marlowe@72.44.190.250) |
| 12:49.12 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 12:49.26 | asterisk-learner | hi, if i want 2 asterisk machines to "talk" to each other, forward calls and so |
| 12:49.48 | asterisk-learner | is it enough to add an extension and sip registry or is it mandatory to change peers.conf? |
| 12:50.58 | *** join/#asterisk coppice (~chatzilla@79.194.17.210.dyn.pacific.net.hk) |
| 12:53.04 | kaldemar | asterisk-learner: http://ofps.oreilly.com/titles/9780596517342/asterisk-OutsideConn.html#c_OutsideConnectivity_id291273 |
| 12:54.26 | kaldemar | sideone: at least asterisk can. |
| 12:58.13 | sideone | cool! |
| 12:58.36 | asterisk-learner | kaldemar: ok thx, but why is peers.conf used then ? (ther is no ref to peers.conf in the link u sent me) |
| 12:58.37 | sideone | kaldemar; great to hear! |
| 12:59.02 | sideone | i really just need to have a bunch of video sessions together |
| 12:59.11 | sideone | 1 broadcast, and many recieve |
| 12:59.17 | kaldemar | asterisk-learner: what is peers.conf? you tell me. |
| 13:00.22 | kaldemar | sideone: oh, video. asterisk doesn't do video conferencing yet. a third party app for asterisk called app_konference does as far as i know. |
| 13:00.42 | sideone | cool thanks buddy |
| 13:03.33 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 13:03.33 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 13:05.43 | asterisk-learner | mmmmmm |
| 13:08.03 | *** join/#asterisk rlfx (~lfx@post.prototechnika.lt) |
| 13:08.26 | leifmadsen | asterisk-learner: there is no peers.conf |
| 13:11.11 | zapotek6 | Hi to all, I've installed the Digium card B410P but I cannot place outgoing call (incoming was not tested). The driver seem loaded (checked lsmod, and with dmesg no warning are logged). I've only one bri channel attached to span1, the corresponding led on the card is green and In the full log of asterisk I see that No D-channels available. When I try to place a call I see in asterisk console the message "Dial failed for some reason with DIALSTATUS = CHANUN |
| 13:13.34 | *** join/#asterisk psilikon (~joel@cerberus.vicimarketing.com) |
| 13:14.21 | kaldemar | zapotek6: how did you configure the card in system.conf and chan_dahdi.conf and how are you dialing in extensions.conf? |
| 13:15.15 | zapotek6 | system.conf: |
| 13:15.21 | zapotek6 | span=1,0,0,CCS,AMI |
| 13:15.21 | zapotek6 | span=2,0,0,CCS,AMI |
| 13:15.21 | zapotek6 | span=3,0,0,CCS,AMI |
| 13:15.21 | zapotek6 | span=4,0,0,CCS,AMI |
| 13:15.21 | zapotek6 | bchan=1-2,4-5,7-8,10-11 |
| 13:15.22 | zapotek6 | dchan=3,6,9,12 |
| 13:15.22 | zapotek6 | loadzone=it |
| 13:15.23 | zapotek6 | defaultzone=it |
| 13:15.24 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 13:15.28 | leifmadsen | ~pb |
| 13:15.29 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 13:16.02 | leifmadsen | zapotek6: if dahdi_tool shows the card fine, then you still need to configure chan_dahdi.conf for asterisk to be able to use the hardware |
| 13:16.20 | leifmadsen | additionally, if this is digium hardware, you do get free support for this type of thing from them |
| 13:18.25 | zapotek6 | dahdi_tool show span 1 OK, internally clocked, irq misses 0, bipolar viol 0, tx/rx levels 0/0, total /conf/act 3/3/0 |
| 13:19.03 | leifmadsen | ok, then the hardware is setup fine |
| 13:19.10 | leifmadsen | now you have to configure asterisk to be able to access the hardware |
| 13:22.26 | zapotek6 | chan_dadhi.conf -> http://pastebin.com/eVav3j8n |
| 13:22.39 | *** join/#asterisk engrxyz (~hhghgfhgf@host81-150-217-173.in-addr.btopenworld.com) |
| 13:24.20 | zapotek6 | chan_dahdi_groups.conf -> http://pastebin.com/XaDTTivJ |
| 13:28.15 | epart | any manual how to install asterisk in ubuntu lucid |
| 13:28.22 | epart | kindly help |
| 13:29.10 | kaldemar | zapotek6: what you have under [general] in chan_dahdi.conf does not belong under [general] but under [channels]. |
| 13:31.17 | epart | kindly help... manual how to install asterisk in ubuntu lucid |
| 13:31.28 | leifmadsen | epart: you just asked like 2 mins ago |
| 13:31.39 | leifmadsen | epart: did you google "asterisk book" at all? |
| 13:31.44 | jacc0 | @epart: apt-get install asterisk |
| 13:32.03 | m4xx | i'm trying to SendDTMF(1*21#,500,500) and the call keeps ending as soon as it gets to the * |
| 13:32.03 | leifmadsen | infobot: tell epart about thebook |
| 13:32.30 | m4xx | i dont see anything on my logs so i believe the remote part is ending the call |
| 13:32.44 | m4xx | any idea why this may be? |
| 13:33.18 | m4xx | *party |
| 13:33.22 | epart | <PROTECTED> |
| 13:33.29 | leifmadsen | epart: check your messages |
| 13:33.31 | m4xx | ~thebook |
| 13:33.31 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342. Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/, or ~buybook |
| 13:33.36 | m4xx | ? |
| 13:33.40 | jacc0 | @epart: just type : apt-get install asterisk |
| 13:33.43 | jacc0 | in the console |
| 13:33.47 | jacc0 | and asterisk is installed |
| 13:34.02 | jacc0 | I can't make a shorter installation manual |
| 13:34.12 | leifmadsen | m4xx: simplify by just using SendDTMF(*) to a phone or something and see if that is the case -- then build it up each time and find out exactly what the problem is |
| 13:34.27 | m4xx | that's what i did |
| 13:34.29 | m4xx | to find out it's the * |
| 13:34.36 | zapotek6 | I've modified chan_dahdi.conf and then restarted services with /etc/init.d/asterisk stop; /etc/init.d/dahdi restart; /etc/init.d/asterisk start. But no change. Asterisk full log report No D-Channels available and placing a call asterisk console report "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 0" |
| 13:35.01 | leifmadsen | m4xx: then it may be a bug you have to file |
| 13:35.35 | jacc0 | @m4xx: check with tshark who is sending the first bye |
| 13:35.55 | jacc0 | @m4xx: tshark -R ""sip || rtpevent" |
| 13:36.11 | malcolmd | zapotek6: you've got a digium b410p? have you contacted digium support directly (http://www.digium.com/support) yet? |
| 13:36.34 | jacc0 | tshark -R "sip || rtpevent" |
| 13:36.41 | leifmadsen | malcolmd: that's what I suggested about 20 mins ago |
| 13:36.58 | malcolmd | leifmadsen: sorries, didn't see that in the backlog :D |
| 13:37.06 | m4xx | jacc0, installing now |
| 13:37.38 | leifmadsen | malcolmd: oh no worries :) |
| 13:38.24 | jacc0 | @m4xx: what dtmf mode are you using? rfc2833,info or inband? and wht asterisk version are you using? |
| 13:38.38 | m4xx | i didn't change anything |
| 13:40.03 | m4xx | i'll try setting it to auto |
| 13:42.03 | jacc0 | what asterisk version are you using? |
| 13:42.13 | m4xx | 1.8.3.3 |
| 13:42.58 | jacc0 | I've had no problems sending DTMF * in 1.8.3.3 |
| 13:43.26 | m4xx | didn't think it was asterisk |
| 13:43.46 | leifmadsen | well SendDTMF would just be using the underlying DTMF mechanisms to send DTMF |
| 13:43.50 | jacc0 | I've been working with several DTMF based protocols that use * in combination with 1.8.3.3 |
| 13:43.51 | m4xx | i'm wondering if maybe i'm sending it for too long/short, wrong mode, didn't know how to debug |
| 13:44.04 | ruyo | If I don't set master-clock in mISDN.conf does that mean Asterisk will use the pseudo dahdi clock? Does one thing have to do with the other at all? |
| 13:44.16 | leifmadsen | m4xx: enable DEBUG level logging, plus DTMF logging on the console, along with SIP debug to see what is going on |
| 13:44.49 | jacc0 | tshark rfc2833 dtmf: tshark -R "rtpevent" |
| 13:45.49 | jacc0 | tsahrk SIP-info dtmf : tshark -R "sip" |
| 13:45.51 | *** part/#asterisk ruben23 (~RLACUMBA@121.97.111.142) |
| 13:46.31 | jacc0 | if you want more detail from tshark then appand -V |
| 13:46.36 | jacc0 | *append |
| 13:50.55 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 13:52.23 | m4xx | looks like i'm hanging up =\ |
| 13:52.29 | m4xx | 298.527919 209.216.XX.XX -> 66.36.XX.XX SIP Request: BYE sip:15165551212@66.36.XX.XX:5060 |
| 13:53.22 | leifmadsen | m4xx: did you enable a feature code that says to hang up when * is pressed? |
| 13:53.26 | leifmadsen | that is a real option :) |
| 13:53.32 | m4xx | hmm |
| 13:53.34 | m4xx | let me check |
| 13:53.36 | m4xx | freatures.conf? |
| 13:53.37 | leifmadsen | could be a simple flag on the Dial(), or in features.conf |
| 13:54.43 | Freeaqingme | if I do dial(sipuser1,sipuser2) and one of them is BUSY, and the other doesn't answer. Will dialstatus then return busy or or unanswered? |
| 13:54.59 | jaytee | if I set a sip device as a persistent queue member and set the call-limit to 24 on the peer will I be able to pass up to 24 concurrent calls to that peer? (Voiceguide IVR system with Dialogic HMP 3.0 on W2K3 server) For systems with 4 licensed ports I've created 4 persistent queue members and set call-limit to 1 but I'd like to simplify my sip.conf and the config for Voiceguide if possible. |
| 13:54.59 | jaytee | I only want to queue a call if all 24 licenses are in use. |
| 13:55.18 | leifmadsen | Freeaqingme: it'd be a very easy test to reproduce that and find out on your own |
| 13:55.36 | Freeaqingme | leifmadsen, yes, but I currently have only one phone |
| 13:55.49 | leifmadsen | install a softphone |
| 13:56.10 | Freeaqingme | hmm, can do |
| 13:57.42 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 14:01.39 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:01.39 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:01.54 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 14:04.57 | ruyo | In mISDN, can setting "poll" to 128, being dsp_poll 64, cause calls to disconnect? |
| 14:05.08 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
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| 14:14.20 | m4xx | leifmadsen, i didn't see that feature but i removed features.conf to ensure i didn't miss anything. you mentioned a flag in the dial command, i don't have any flags set, is there something i should set? |
| 14:15.38 | *** join/#asterisk hexanol (~etienne@modemcable094.94-70-69.static.videotron.ca) |
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| 14:20.39 | m4xx | ah, agents.conf |
| 14:23.48 | m4xx | nope |
| 14:25.43 | Freeaqingme | I'm using dial(sip/user1&sip/user2). Now I should be abel to check using ${HASH(SIP_CAUSE,<channel-name>)}. what channel returned what status, but how do I get the channel names used? |
| 14:27.05 | *** part/#asterisk afrance (~afrance@65.117.122.45) |
| 14:27.28 | leifmadsen | use DumpChan() to see what data is available to you |
| 14:32.05 | *** join/#asterisk Kobaz (~kobaz@its.kobaz.net) |
| 14:33.35 | Freeaqingme | leifmadsen, tnx |
| 14:34.25 | jacc0 | :) nice one, didn't know that |
| 14:35.08 | leifmadsen | DumpChan() ftw |
| 14:35.13 | leifmadsen | used it more than once :) |
| 14:35.48 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 14:37.58 | Freeaqingme | leifmadsen, on 1.8.4-rc2 whenever I call (over agi) exec dumpchan I get a 'invalid or unkown command' for that, but also every command that follows. Possible bug? |
| 14:38.17 | leifmadsen | not sure |
| 14:38.37 | leifmadsen | sounds like a bug in the way you're using commands from your script |
| 14:41.20 | m4xx | i dont get, no flags are set (h or H), i set disconnect => ***0 in features.conf |
| 14:41.48 | m4xx | set endcall=no in agents.conf |
| 14:41.57 | m4xx | and still, as soon as i hit * it ends the call |
| 14:42.34 | m4xx | features show, show's that the current bind is infact set to ***0 |
| 14:46.29 | *** join/#asterisk orn (~orn@rtr1.sh23.sip.is) |
| 14:47.19 | orn | Hi, weird problem; When I connect to manager, with events on, I don't see any events, until I try to issue a command. So, I just idle there, but if I press enter, I get all the buffered up output. |
| 14:47.25 | orn | Is this a configuration issue? |
| 14:49.09 | orn | It's just on this particular asterisk box -- not on any others that I'm running |
| 14:55.17 | *** join/#asterisk chandoo (~chandoo@ool-4a5a2824.dyn.optonline.net) |
| 14:58.03 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 15:01.16 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 15:06.06 | *** join/#asterisk adyn (~adyn@pdpc/supporter/active/adyn) |
| 15:06.26 | orn | So pretty much, no events... issue an empty newline, all the buffered events show up |
| 15:10.04 | *** join/#asterisk SuPrSluG (~SuPrSluG@8.22.96.106) |
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| 15:28.38 | m4xx | anyone? =[] |
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| 15:33.08 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
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| 15:36.36 | Lipsum | is it legal to ask a question regarding cisco ip phone configuration here? |
| 15:37.48 | Qwell | Lipsum: Don't worry, we won't call the Cisco police. |
| 15:38.47 | Lipsum | ;-) |
| 15:39.01 | *** join/#asterisk moy (~moy@CPE002719f00364-CM0026f3a1701d.cpe.net.cable.rogers.com) |
| 15:39.12 | m4xx | anyone know how to disable hangup on star? |
| 15:43.32 | Lipsum | I'm building an phonebook on a SPA504G using CiscoIPPhoneMenu. If I have two SoftKeyItems on the same display, how can I figure out which menu item is marked AND which softkey is pressed? |
| 15:46.51 | m4xx | Lipsum: have you checked out http://www.cisco.com/en/US/docs/voice_ip_comm/cuipph/all_models/xsi/7_1_2/english/programming/guide/xsi712xml.html |
| 15:50.08 | Lipsum | m4xx, yeah, I've used the documentation. Problem is that my CiscoIPPhoneMenu contains all available numbers of selected subscriber (ie. mobile number, internal number, home number, etc), one number for each MenuItem. And I want one button to "dial out selected number" and one button to "transfer current call to selected number". |
| 15:50.45 | Lipsum | The problem is that using URL on the SoftKeyItem won't tell me which MenuItem which is selected... |
| 15:51.09 | m4xx | what are you trying to do? |
| 15:53.52 | Lipsum | I want to know which one of the two buttons was pressed via the target URL. |
| 15:55.43 | *** join/#asterisk cusco (~tralala@a89-152-96-250.cpe.netcabo.pt) |
| 15:55.44 | cusco | hi |
| 15:55.49 | m4xx | can you pastebin your xml file? |
| 15:55.59 | cusco | can I make it so that whenever asterisk service starts, verbose is always set to 15? |
| 15:56.12 | leifmadsen | cusco: yes, see cli.conf or asterisk.conf |
| 15:56.36 | cusco | thanks |
| 15:58.57 | m4xx | leifmadsen, i dont mean to be a thorn in your side, but i still cant seem to figure this out, what other settings could it be? |
| 15:59.38 | Lipsum | m4xx, http://pastebin.com/4PuVPUqJ |
| 15:59.47 | leifmadsen | m4xx: I have no further suggestions |
| 16:00.34 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 16:01.01 | m4xx | Lipsum, why use the softkeyitems for that? |
| 16:01.08 | Lipsum | m4xx, what would you use? |
| 16:01.15 | m4xx | i believe the url in the menu item would be triggered |
| 16:01.17 | m4xx | when you select it |
| 16:01.45 | Lipsum | m4xx, sure, I intent to use URLs in the MenuItem. But, the problem is, how can I know which of the two buttons which was pressed? |
| 16:02.35 | m4xx | just append a query string item |
| 16:03.21 | Lipsum | I don't follow. |
| 16:03.56 | m4xx | i'm not sure that your softkeys will both fire their <url> and the list items <url> |
| 16:04.43 | m4xx | have you tried QueryStringParam ? |
| 16:04.50 | Lipsum | Take a look at: http://pastebin.com/TmN9fEMf |
| 16:05.30 | Lipsum | Now, if I press "Test 1", it will open the given URL and I will know which button was pressed with $_GET['button'], but I will not know which MenuItem was selected. |
| 16:05.49 | *** join/#asterisk endemic (~endemic@2001:49f0:400e::5) |
| 16:05.59 | Lipsum | If I add a SoftKey:OK instead of the SoftKeys with URLs, I would know which item was selected, but not which button was pressed. |
| 16:06.10 | Lipsum | I need to know both... |
| 16:06.18 | m4xx | can you add QueryStringParam to the button? |
| 16:07.55 | Lipsum | Not according to the doc. |
| 16:08.00 | m4xx | just guessing |
| 16:12.35 | Lipsum | Any other suggestions? :) |
| 16:13.20 | m4xx | try adding <URL>QueryStringParam:item=1</URL> |
| 16:13.22 | m4xx | to your menu items |
| 16:14.00 | m4xx | actually |
| 16:14.07 | m4xx | n/m |
| 16:14.09 | m4xx | yeah try that |
| 16:16.23 | Lipsum | Haha, that worked like a charm! |
| 16:16.32 | m4xx | you can also do the same with teh buttons |
| 16:16.34 | m4xx | which may be easier |
| 16:16.41 | m4xx | http://www.cisco.com/en/US/docs/voice_ip_comm/cuipph/all_models/xsi/3_3_4/english/programming/guide/ip334ch3.html |
| 16:17.44 | Lipsum | Perfect! Thanks a lot, m4xx. :) |
| 16:18.03 | m4xx | n/p, i just got the same phone a few days go ;D |
| 16:22.44 | *** join/#asterisk Tozz_ (Tozz@hardwire.duocast.net) |
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| 16:35.07 | Jcook_5xData | ? for anyone we have PLantronics cs55 + Ploycom 301-501 running asterisk 1.6. The problem is when The user use the headset the calls become echoes and static, but if they pickup the handset it is fine. anyone had this problem? how did you fix it? |
| 16:40.49 | *** join/#asterisk kaii (~kh@ciphron.de) |
| 16:41.56 | kaii | what is needed to use native bri support in asterisk 1.8/dahdi 2.4.1? chan_dahdi keeps telling me: Unknown signalling method 'bri_cpe' |
| 16:42.33 | Tim_Toady | kaii libpri |
| 16:42.39 | Jcook_5xData | did you install libpri? |
| 16:42.49 | kaii | yes i did, seems i did something wrong? |
| 16:45.44 | Jcook_5xData | and you set up chan_dahdi.conf for your bri signal and switch type |
| 16:58.02 | orn | Does anyone have a clue why an AMI connection with Events: on doesn't output any data unless an attempt to enter a command is issued (empty newline will suffice) |
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| 17:08.15 | joesuffceren | Do I need to shut down my asterisk box before plugging in analog phone lines into an FXO port on a TDM410, or is okay to do that while the box is on? |
| 17:10.10 | Tim_Toady | joesuffceren no need to shut down anything, just plug it |
| 17:10.51 | joesuffceren | Tim_Toady: cool. thanks. Figured it would probably be okay, but wanted to ask before frying a $400 card. :-) |
| 17:11.01 | m4xx | chicken |
| 17:11.02 | m4xx | ;x |
| 17:12.18 | kaii | Jcook_5xData: yes i did setup chan_dahdi for euroisdn and bri_cpe[_ptmp] |
| 17:20.25 | kaii | how can i check if dahdi/chan_dahdi is compiled with bri support? |
| 17:25.29 | *** join/#asterisk Juggie (~Juggie@unaffiliated/juggie) |
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| 17:31.17 | kaii | dont get it. i recompiled and installed libpri 1.4.11.5 from sources, did the same for dahdi 2.4.1, did the same for asterisk 1.8.4-rc3. |
| 17:31.32 | kaii | still chan_dahdi says: Unknown signalling method 'bri_cpe' |
| 17:33.03 | Qwell | kaii: Do you have an updated libpri? |
| 17:33.54 | kaii | i have 1.4.11.5 from /pub/telephony/libpri |
| 17:34.14 | kaii | Qwell: *highlight* |
| 17:34.58 | Qwell | shrugs |
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| 17:36.49 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 17:40.00 | kaii | i feel stupid and stuck |
| 17:40.38 | Qwell | kaii: What steps did you take, exactly? |
| 17:40.43 | Qwell | and in which order |
| 17:40.59 | kaii | well first i forgot about libpri, installed dahdi, then asterisk. |
| 17:41.37 | kaii | Qwell: when i realized that the signalling part was missing, i started over again, beginning with libpri, then dahdi, then asterisk |
| 17:41.48 | Qwell | What exact steps did you take? |
| 17:42.45 | kaii | cd libpri-*; make && make install && cd ../dahdi-linux-complete && make clean && make && make install && cd ../asterisk && ./configure && make && make install |
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| 17:43.47 | kaii | Qwell: is there chance that there is something left over in the FS causing my build of bri support to fail? |
| 17:44.08 | Qwell | no, not if you followed those exact steps |
| 17:45.16 | kaii | Qwell: well, i'm going to start over again .. doing a make clean everywhere (also in asterisk, even if i have seen that chan_dahdi.so was rebuilt) |
| 17:45.21 | Qwell | I do notice that you didn't do a make clean in Asterisk, but that have should worked. |
| 17:45.34 | kaii | i think so too, yes |
| 17:45.34 | Qwell | the configure should have forced it |
| 17:48.03 | kaii | Qwell: maybe i need a reboot after that, because dahdi.so (and drivers) are still in memory |
| 17:48.32 | kaii | Qwell: but shouldnt chan_dahdi should recognize the bri signalling? or is that passed to the kernel module and processed there? |
| 17:48.46 | Qwell | kaii: as far as I know it's just chan_dahdi |
| 17:49.58 | kaii | weird |
| 17:50.39 | kaii | dahdi should just fail building when there is no lipri ;) |
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| 17:59.33 | WannaBeGeekster | Anyone have any suggestions on Wholesale providers for a startup VoIP provider? |
| 18:00.11 | Freeaqingme | us based? |
| 18:00.36 | *** join/#asterisk ariel_ (~chatzilla@pdpc/supporter/active/abatista) |
| 18:00.50 | WannaBeGeekster | Yes |
| 18:01.42 | kaii | Qwell: just FYI. make clean && configure && make in asterisk did the trick. weird, i have seen it recompiling chan_dahdi |
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| 18:18.36 | *** join/#asterisk b0ot (~Jinxed---@147.177.57.10) |
| 18:19.22 | b0ot | This is a really stupid question, but a lot of the configuration of Polycom phones seem to use the .cfg file. However I dont' know how to actually access the phones .cfg file |
| 18:19.36 | b0ot | I set up a ftp server on my computer and had the phone uploading logs there |
| 18:19.43 | b0ot | but I didn't know how to download the .cfg file |
| 18:19.54 | *** join/#asterisk marlowe (~marlowe@72.44.190.250) |
| 18:20.04 | b0ot | ... so How do you download the .cfg files off a polycom phone |
| 18:21.07 | paulc | b0ot: it's usually the other way round - you provide a full config file (or in multiple files) based on the skeleton/defaults.. and the phone gets it when it boots.. |
| 18:21.27 | paulc | it sounds like you've configured the phones via the web interface and now want to save a config file? or did I read it wrong? |
| 18:21.35 | b0ot | correct |
| 18:21.58 | b0ot | or I at least want to know how to get the skelton configs so I could redo the config |
| 18:24.21 | b0ot | <PROTECTED> |
| 18:24.40 | paulc | b0ot: Yeah, see http://www.polycom.com/global/documents/whitepapers/configuration_file_management_on_soundpoint_ip_phones.pdf |
| 18:25.06 | paulc | You basically get a ZIP file with latest bootloader/firmware etc, and a bunch of sample files.. and a huge big PDF for administration.. |
| 18:35.16 | b0ot | paulc, i downloaded the new firmware for my phone |
| 18:35.23 | b0ot | but it doesn't even have a sip.cfg |
| 18:36.01 | paulc | was the file you downloaded just the latest firmware (only)? no other config files etc? |
| 18:36.27 | b0ot | no it had a config folder |
| 18:36.59 | b0ot | it had a sip-basic.cfg |
| 18:37.02 | b0ot | but no sip.cfg |
| 18:41.52 | b0ot | ... is there no easy to configure a phone |
| 18:41.54 | b0ot | ... |
| 18:42.06 | b0ot | all ineed to do is change some lines in a sip.cfg file that doesn't seem to exist |
| 18:42.15 | b0ot | I know it's on my phone somewhere |
| 18:42.23 | b0ot | if I could just get it off and put it back with the lines modified |
| 18:42.28 | b0ot | I woudl be done |
| 18:42.34 | b0ot | this admin guide is 421 pages |
| 18:42.50 | b0ot | and refers to mulitple technical bullitens and white papers |
| 18:43.57 | E-bola | dont u just have to rename the basic one? |
| 18:44.10 | b0ot | where is the basic one? |
| 18:44.16 | E-bola | <b0ot> it had a sip-basic.cfg |
| 18:44.19 | b0ot | you mean the sip_basic.cfg |
| 18:45.45 | sezuan | Does anyone has an idea how I can debug gtalk. I've problems with connection to gtalk clients using googlemail + plugin. |
| 18:46.03 | b0ot | E-bola, hmm no the options weren't in the sip-basic.cfg file |
| 18:46.13 | b0ot | but they were in the sip-interop.cfg file |
| 18:46.33 | E-bola | im guessing what you want to set is advanced since the web interface apparently doesnt provide it |
| 18:52.10 | *** join/#asterisk Freeaqingme (~Freeaqing@83.160.112.249) |
| 18:53.36 | b0ot | E-bola, correct... I need to modify some dtmf settings |
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| 19:14.18 | epart | any body help\ |
| 19:14.20 | epart | exten => _81NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log) |
| 19:14.21 | epart | exten => _81NXXNXXXXXX,n,Dial(${TrunkAPN}/${EXTEN:1},,To) |
| 19:14.21 | epart | exten => _81NXXNXXXXXX,n,Hangup |
| 19:14.32 | *** join/#asterisk sourcode_ (~code@ppp-58-8-154-193.revip2.asianet.co.th) |
| 19:14.33 | epart | add caller ID in this |
| 19:15.08 | epart | kindly help |
| 19:15.36 | epart | im doing outbound and i want my caller ID be this dirct from my telco |
| 19:15.40 | m4xx | exten => 1,2,Set(CALLERID(num)=15555551212); |
| 19:15.42 | epart | exten => _81NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log) |
| 19:15.43 | epart | exten => _81NXXNXXXXXX,n,Dial(${TrunkAPN}/${EXTEN:1},,To) |
| 19:15.43 | epart | exten => _81NXXNXXXXXX,n,Hangup |
| 19:16.03 | epart | were to put it? |
| 19:16.29 | m4xx | anywhere |
| 19:16.42 | m4xx | it's priority is numbered, it'll be executed after 1 |
| 19:16.51 | *** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr) |
| 19:18.28 | carrar | put it in your AGI!! ;) |
| 19:23.25 | Jcook_5xData | I have a weird problem about once a day I think my pri is being reset. I read somewhere to add resetinterval=never to the chan_dahdi.conf witch I did but it still happens. anything else to try |
| 19:25.48 | b0ot | Why does the new polycom sip firmware not have a sip.cfg, but have a sip-basic.cfg and an sip-interop.cfg |
| 19:26.08 | b0ot | how would I get my phone to take any modifications i made in these files |
| 19:28.42 | epart | may in know how to add CID in ur outbound dialplan |
| 19:28.43 | Jcook_5xData | what ever file you are using to tell the phone which file to include I add my in the mac.cfg |
| 19:29.08 | *** join/#asterisk neurosys (~neurosys@173-9-159-177-miami.txt.hfc.comcastbusiness.net) |
| 19:30.26 | Jcook_5xData | epart, If you want it set by the telco do not pass any leave it blank. I have on ATT |
| 19:30.38 | Jcook_5xData | I have to on ATT* |
| 19:31.56 | Jcook_5xData | When I try to pass some thing to them, they try to over ride and then it bombs and my calls come up Unknown |
| 19:32.58 | *** join/#asterisk atan (~atan@unaffiliated/atan) |
| 19:33.48 | m4xx | anyone have any ideas as to why my call ends when doing SendDTMF(*) in my dialplan? I've got |
| 19:33.57 | m4xx | "disconnect => ***0" in my features.conf |
| 19:36.03 | *** join/#asterisk NightMonkey (debian-tor@pdpc/supporter/professional/nightmonkey) |
| 19:37.58 | atan | Every ~20 minutes or so my call goes blank for a second or two. I don't see anything with verbose set to 10 and debug is a little bit advanced for me. Could this be a sign of something like the server needing more memory or something? |
| 19:49.50 | Jcook_5xData | atan, how much mem do you have? It really depends on how many users you have. |
| 19:50.13 | *** join/#asterisk Workturk (~Linuturk@unaffiliated/linuturk) |
| 19:51.08 | *** join/#asterisk pdtpatrick__ (~pdtpatric@mainstwan.farheap.com) |
| 19:51.11 | Jcook_5xData | atan, I have a old p4 1.5 with 768 it can handle about 3-4 users before it get bogged down |
| 19:51.24 | atan | I think there is 256 megs in there. |
| 19:51.42 | atan | I am just confused why every few minutes the calls just goes blank for a few seconds. Almost like it's re-registering again or something. |
| 19:51.52 | atan | It might not be the server, it could be the devices trying to register again I suppose. |
| 19:52.01 | pdtpatrick__ | Question - how can i forward an extension so when calls come in to they go to another extension. Can someone please send me a wiki or guide or if chanced, can tell me where to look? Greatly appreciated. |
| 19:52.09 | *** join/#asterisk ferdna (~yup@cpe-67-10-220-35.elp.res.rr.com) |
| 19:52.43 | Jcook_5xData | I would slap another 512 in there remember you need some for the OS to |
| 19:53.09 | atan | Ahh okay then. So it very well could be the server that's crapping out on me then, not so much the phones. |
| 19:53.12 | atan | Otherwise it works great. |
| 19:53.48 | atan | Jcook_5xData, you wouldn't know of a website-based SIP phone? I'm trying to reinvent a call centre and figured it would save a few bucks to ditch the desk phones for soft phones. If it's in the browser all the better! |
| 19:54.03 | atan | I've found 'flaphone' but not many details on it. |
| 19:54.35 | *** join/#asterisk ideaman (~ihaveapla@c-98-202-74-138.hsd1.ut.comcast.net) |
| 19:54.44 | Jcook_5xData | pdtpatrick__, http://www.voip-info.org/wiki/view/Asterisk+call+forwarding |
| 19:55.06 | b0ot | Any idea what "process AVT" might do in a SPA8000 |
| 19:55.20 | pdtpatrick__ | jcook_5xData Thanks!!! |
| 19:55.23 | ideaman | Anyone: If I want to monitor a certain peer for outbound calls, is the Monitor command still the best way? |
| 19:55.42 | ideaman | to record them |
| 19:56.09 | Jcook_5xData | atan, I go with a software bade phone over web http://www.zoiper.com/support.php |
| 19:56.18 | Jcook_5xData | based |
| 19:57.13 | atan | Jcook_5xData, do you know if I can pass data to/from it or does it just load up like a seperate little java app? |
| 19:59.08 | Jcook_5xData | Are you liking to do webbase popups and app intergeneration? |
| 19:59.28 | atan | Jcook_5xData, more or less. |
| 19:59.46 | atan | I figured I'd have Asterisk hit a script to tell me it has a call, script tells it who to send it to, and prompts the users about the account. |
| 20:02.23 | Jcook_5xData | best why I can think to do that is write a small app that monitor AMI and run it on the cleint machine then have it do the action. that a look it this http://www.astassistant.com/ |
| 20:03.05 | Jcook_5xData | It can do whet I think you are looking for and may give you some ideas |
| 20:04.23 | Jcook_5xData | add some more to atleast 512 + even for a test box |
| 20:04.23 | *** join/#asterisk angryuser (53a79c5a@gateway/web/freenode/ip.83.167.156.90) |
| 20:04.40 | b0ot | Why would certain calls work with process AVT = no but other calls not, and then if I enable process AVT the calls that weren't working work, and the calls that were working don't |
| 20:05.47 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 20:07.50 | atan | ty =) |
| 20:12.52 | *** join/#asterisk jkroon (~jkroon@dsl-241-248-179.telkomadsl.co.za) |
| 20:22.07 | ferdna | guys it doesnt matter what vocodec i am using on my phones... as longest asterisk is connecting using the right one, right? |
| 20:22.28 | *** join/#asterisk WiretapWork (~Wiretap@unaffiliated/wiretap) |
| 20:23.16 | *** join/#asterisk kdekorte (~kdekorte@63-163-137-113.iprev.kci.net) |
| 20:24.00 | leifmadsen | I don't understand the question |
| 20:24.08 | leifmadsen | "right one" ? |
| 20:24.25 | carrar | or the one on the left |
| 20:24.56 | WiretapWork | never going to understand people who insist on putting their email address in their email signatures |
| 20:25.10 | m4xx | lol |
| 20:25.28 | WiretapWork | kinda like putting your web address on your website |
| 20:25.30 | carrar | it's smart to do |
| 20:25.52 | carrar | when you have MSFT people forwarding Emails without the full Email addresses |
| 20:26.06 | leifmadsen | WiretapWork: I'd rather they put their phone number so when they say, "Call me because I have no access to email" I know how to get a hold of them |
| 20:26.07 | carrar | You can find it in that users sig |
| 20:26.09 | WiretapWork | argh.... I wish I could figure out why my build of asterisk works perfectly except for voicemail |
| 20:26.20 | WiretapWork | leifmadsen, exactly |
| 20:26.46 | m4xx | $20 via paypal for someone to figure this out =\ |
| 20:27.21 | WiretapWork | I am so stumped |
| 20:27.35 | WiretapWork | when I use the digium built debian package, it all works perfectly |
| 20:27.49 | WiretapWork | when I build my own, everything works, except voicemail |
| 20:29.05 | ferdna | leifmadsen, yes... or how does it works?... so if i specify a codec in sip.conf... that is the one that will be used for all phones and itsp? |
| 20:29.22 | WiretapWork | no error messages, and doing a show voicemail users shows the mailboxes and the number of messages in them |
| 20:29.39 | ferdna | carrar, very funny |
| 20:29.40 | ferdna | :) |
| 20:29.59 | WiretapWork | (my only reason for using the custom build at all is that I need one specific patch to support my phone properly) |
| 20:31.32 | leifmadsen | ferdna: I suggest reading this chapter: http://ofps.oreilly.com/titles/9780596517342/asterisk-DeviceConfig.html |
| 20:35.20 | *** join/#asterisk luckman212_phone (~luckman21@mobile-166-137-137-034.mycingular.net) |
| 20:36.01 | JonathanRose | svn update |
| 20:36.04 | atan | Hmm, anyone use red5phone? Trying to use it it just for SIP but can't get it figured out. |
| 20:36.11 | ferdna | leifmadsen, ok i will |
| 20:36.57 | WiretapWork | leifmadsen, hate to be a pain, but would you happen to have any ideas with regards to my issue? |
| 20:37.39 | WiretapWork | literally nothing has changed in my config... and I only moved from 1.8.3 to 1.8.3.3 |
| 20:38.41 | leifmadsen | WiretapWork: I don't know what your issue is |
| 20:38.56 | leifmadsen | I've been working on other things and only briefly looking in here |
| 20:39.03 | WiretapWork | leifmadsen, ah, ok |
| 20:39.18 | ferdna | and i have another question... |
| 20:39.51 | ferdna | when i get calls from my itsp provided number... it says that can find extension number and the phone number |
| 20:40.04 | ferdna | *to my |
| 20:40.23 | WiretapWork | well... if I use the digium-built .debs, everything works perfectly, but if I build my own (with everything enabled except sounds, since I already have those installed on the target system), voicemail doesn't work, and by doesn't work I mean you get 3 seconds of silence then hangup if you dial *97 or *98, and if your call rings through to voicemail you get the reorder tone |
| 20:41.29 | b0ot | Could someone explain to me what is means to processs AVT |
| 20:42.38 | luckman212_phone | avt is rfc2833 dtmf i believe |
| 20:43.25 | luckman212_phone | its a type of inband dtmf, but uses special rtp packets to signal dtmf instead of "audio" |
| 20:44.15 | *** part/#asterisk epart (~TECH-GENS@121.97.63.210) |
| 20:44.47 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 20:46.51 | WiretapWork | leifmadsen, the patch I applied was this one, which I wouldn't have thought would have affected VM https://issues.asterisk.org/file_download.php?file_id=28973&type=bug |
| 20:46.57 | luckman212_phone | does anyone know of a good guide to setting up phones behind a nat where the asterisk server doesnt have to proxy media? or just some good general guides on various NAT setups (eg phone<->LAN<->asterisk<->NAT<->Wan/internet |
| 20:47.51 | WiretapWork | luckman212_phone, doesn't really need a guide :P |
| 20:48.03 | WiretapWork | externip=YOUREXTERNALIP |
| 20:48.06 | WiretapWork | nat=yes |
| 20:48.10 | luckman212_phone | or phone<->Nat<->internet<->NAT<->asterisk |
| 20:48.36 | WiretapWork | and make sure ports 5060/UDP and an RTP range are forwarded and unfiltered to your * box |
| 20:48.52 | luckman212_phone | WiretapWork: yes but there are lots of gotchas |
| 20:49.01 | WiretapWork | (also make sure to define localnet=SUBNETID/SUBNETMASK) |
| 20:49.29 | WiretapWork | luckman212_phone, well, yes, but in a normal situation the above works |
| 20:49.44 | WiretapWork | it does help if you have a firewall/router that supports SIP-ALG too |
| 20:49.55 | *** join/#asterisk rgagnon (~rgagnon@rrcs-71-42-183-54.sw.biz.rr.com) |
| 20:49.58 | *** part/#asterisk rgagnon (~rgagnon@rrcs-71-42-183-54.sw.biz.rr.com) |
| 20:50.02 | luckman212_phone | my problem these days is i've got my * box behind the same nat router as my sip phone |
| 20:50.19 | WiretapWork | why is that a problem? |
| 20:50.27 | WiretapWork | thats what most people have |
| 20:50.31 | luckman212_phone | i am trying NOT to have my asterisk box proxy all the media |
| 20:50.50 | WiretapWork | set 'nat=no' on the peer? |
| 20:50.55 | luckman212_phone | i want my phone to be able to send media direct somehow |
| 20:51.07 | WiretapWork | and also make sure your localnet is set |
| 20:51.23 | luckman212_phone | WiretapWork: yep but then the RTP doesnt get thru |
| 20:51.56 | WiretapWork | luckman212_phone, if localnet is set properly, and nat=no on the peer, why wouldn't it? |
| 20:52.06 | WiretapWork | obviously you need to make sure nat is off on the phone |
| 20:52.21 | luckman212_phone | Ive heard of some setups where people have some kind of "symmetric rtp" or something so the ports would be open/fwd'ed to the right device on the Lan during a call |
| 20:52.56 | WiretapWork | ports opened/forwarded? |
| 20:52.57 | WiretapWork | um |
| 20:53.06 | WiretapWork | if you're behind NAT there's no such thing as 'forwarding' ports |
| 20:53.22 | WiretapWork | are you sure there isn't another NAT between the phone and asterisk? |
| 20:53.32 | luckman212_phone | yes positive |
| 20:53.56 | luckman212_phone | e.g phone is 192.168.1.226 and asterisk is 192.168.1.248 |
| 20:54.04 | WiretapWork | no firewalls, nothing? |
| 20:54.25 | luckman212_phone | asterisk is running iptables |
| 20:55.14 | luckman212_phone | the nat router is too but its not restricting anything specific-- there are some port fwds for SIP (port 5060) to the asterisk server |
| 20:56.09 | luckman212_phone | in my rtp.conf i have rtpstart=16384 and rtpend=16403 |
| 20:56.09 | *** join/#asterisk dfamorato (~dfamorato@173-9-190-190-miami.txt.hfc.comcastbusiness.net) |
| 20:57.21 | WiretapWork | so iptables is in use or simply installed and free of rules? |
| 20:57.50 | luckman212_phone | its installed and there are some rules for port fwds |
| 20:58.33 | luckman212_phone | WiretapWork: hey thanks for the help my doctor just called me in (was in waiting rm) so gotta run. will pop back in later |
| 20:58.44 | WiretapWork | kk |
| 20:58.44 | luckman212_phone | :-/ |
| 20:58.53 | WiretapWork | so there IS another NAT |
| 20:59.00 | WiretapWork | its just..... on the same box as the *, wtf |
| 21:01.11 | ferdna | oh, btw... if i use G729 and G723 what is the deal with licensing stuff? will it perform poorer if i dont use a license? |
| 21:01.24 | *** join/#asterisk MarKsaitis (~MarKsaiti@82.144.242.158) |
| 21:01.41 | WiretapWork | ferdna, you can't use g.729 without a license |
| 21:02.38 | leifmadsen | a license is required for use, yes |
| 21:02.54 | leifmadsen | it will perform so poorly that it won't even work without a license :) |
| 21:03.21 | ferdna | WiretapWork, where can i get one? how much does it cost |
| 21:03.27 | leifmadsen | www.digium.com |
| 21:03.29 | ferdna | and how do you install this license? |
| 21:03.30 | leifmadsen | $10 per license |
| 21:03.47 | WiretapWork | one license per channel |
| 21:03.54 | ferdna | awesome =) |
| 21:03.57 | leifmadsen | http://www.digium.com/en/products/g729codec.php |
| 21:07.31 | WannaBeGeekster | Is there a good "big picture" overview of running a VoIP company that anyone can suggest for me? |
| 21:08.00 | ideaman | If I want to record a certain peer for outbound calls, is the Monitor command the best way? |
| 21:08.10 | _Corey_ | overview of running a VoIP company? what do you mean by "voip company"? |
| 21:09.58 | WannaBeGeekster | I guess that does over generalize quite a bit |
| 21:10.12 | WannaBeGeekster | Basically Vonage. Not a wholesaler |
| 21:11.16 | _Corey_ | in that case, I don't know of a "guide" per se |
| 21:11.28 | WannaBeGeekster | Yes, that is what I thought. |
| 21:11.54 | _Corey_ | There are several turn-key platforms out there though |
| 21:12.20 | WannaBeGeekster | Really? I never thought about that part honestly |
| 21:13.20 | WannaBeGeekster | Any suggestions on some good ones? |
| 21:14.19 | _Corey_ | Ours is a home-brew involving Asterisk/Kamailio and a bunch of billing/accounting/provisioning interfaces |
| 21:15.11 | WannaBeGeekster | Do you guys do any consulting? |
| 21:18.59 | *** part/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 21:19.16 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 21:19.21 | jkroon | ferdna, WiretapWork, leifmadsen - re g729 - the above isn't entirely correct, you need a license to transcode between G.739 and other codecs, you can function in pass-through mode without a license. |
| 21:20.37 | jkroon | WannaBeGeekster, just one piece of advise: understand *, even if you end up using some other solution on top of it. |
| 21:21.07 | ferdna | jkroon, oooohhh.... |
| 21:21.25 | leifmadsen | jkroon: right, which is exactly what he was asking for |
| 21:21.40 | leifmadsen | G.723 to G.729 |
| 21:21.58 | ferdna | leifmadsen, nope... i ask about those two codecs independently |
| 21:22.13 | pabelanger | burn |
| 21:22.25 | _Corey_ | WannaBeGeekster: I don't want to solicit you here, though we do |
| 21:22.42 | pabelanger | WannaBeGeekster: #asterisk-consultants |
| 21:22.46 | leifmadsen | ferdna: then you were not very clear about that |
| 21:23.05 | _Corey_ | WannaBeGeekster: There was someone at ITEXPO east this year that had an interesting framework solution but I'm having trouble finding where I bookmarked them |
| 21:23.12 | WannaBeGeekster | _Corey_: Got it. Thanks |
| 21:23.27 | ferdna | leifmadsen, yes i was: |
| 21:23.28 | ferdna | [15:01] <ferdna> oh, btw... if i use G729 and G723 what is the deal with licensing stuff? will it perform poorer if i dont use a license? |
| 21:24.00 | WannaBeGeekster | Well we just have a unique in road to a specific market. We have extra servers just sitting there waiting to be utilized. We just need the services setup so we can make the money. |
| 21:24.20 | leifmadsen | ferdna: that is not clear, as you're saying, "if I use G729 and G723" which reads as "using them at the same time" |
| 21:24.58 | ferdna | i never said "at the same time" |
| 21:25.00 | ferdna | :D |
| 21:25.12 | ferdna | ok, lets continue with life.... |
| 21:25.32 | leifmadsen | oh phew, I almost didn't |
| 21:25.48 | pabelanger | leifmadsen: hehe |
| 21:25.51 | _Corey_ | WannaBeGeekster: just remember, routing calls is half the battle... there's much more to the "rest" than many realize |
| 21:25.51 | ferdna | hehehehe |
| 21:28.08 | *** join/#asterisk marlowe (~marlowe@ip68-100-147-177.dc.dc.cox.net) |
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| 21:30.19 | pdtpatrick__ | what's the default asterisknow username and password for the index.html page? |
| 21:30.30 | WannaBeGeekster | Sure, billing support etc. Can be a real nightmare I am sure. |
| 21:30.32 | pdtpatrick__ | freepbx/fpbx doesn't work |
| 21:31.16 | _Corey_ | WannaBeGeekster: I finally found the link... |
| 21:31.20 | _Corey_ | Have a look at http://www.2600hz.org/ |
| 21:31.36 | _Corey_ | Hope it helps |
| 21:32.04 | WannaBeGeekster | Sweet. Thanks man. I do apprecaite it |
| 21:35.04 | *** join/#asterisk jrrose (~Jon@207.98.203.74) |
| 21:40.35 | *** join/#asterisk [T]ank (~Tank@206.71.78.158) |
| 21:41.45 | [T]ank | is it possible to do a NoOp(<value of history-info from sip header)? I am reading about asterisks func sip_header and wondering if something like that can be used in the dialplan its self. Could that be possible? |
| 21:42.53 | leifmadsen | dialplan functions are very much used in the dialplan :) |
| 21:43.01 | leifmadsen | core show functions like SIP |
| 21:43.54 | [T]ank | understood, but can i pull just one line out of it like the history_info line? ***heading to my CLI to do a core show funcions*** |
| 21:44.16 | leifmadsen | depends on what the function provides |
| 21:44.30 | leifmadsen | you probably want SIP_HEADER()_ |
| 21:45.41 | [T]ank | ill give it a shot, thanks |
| 21:46.34 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 21:57.12 | [T]ank | im looking for somewhere that will document the list of sip header names that asterisks function sip_header will be able to use. looking on the wiki and in asterisk core show func... not finding anything. Does anyone know of a list of names i am able to use in the function? |
| 21:59.13 | [T]ank | NoOp(${SIP_HEADER(History-Info)}) does not return anything, but NoOp(${SIP_HEADER(Via)}) does return values. |
| 22:04.34 | leifmadsen | look at the SIP trace and see if that header exists |
| 22:05.41 | [T]ank | its not in the sip trace |
| 22:05.48 | [T]ank | but, its in the header being sent from an Avaya system. |
| 22:06.06 | [T]ank | is a sip header not the same from system to system? |
| 22:06.07 | leifmadsen | if it's not in the SIP trace then it's not there |
| 22:07.05 | [T]ank | k, hang on... is sip trace and sip debug different? or are they the same thing? |
| 22:07.34 | leifmadsen | same thing |
| 22:07.37 | leifmadsen | 'sip trace' is not a command |
| 22:07.44 | leifmadsen | unless you add it to cli_aliases.conf :) |
| 22:07.45 | [T]ank | thats why i asked. |
| 22:07.55 | [T]ank | was just mkaing sure |
| 22:08.03 | *** join/#asterisk jksM (jks@193.189.93.254) |
| 22:09.43 | [T]ank | what i am trying to do is transfer a call from Avaya to Asterisk. But I need to retain the original dialed number the client used to dial into Avaya. Avaya sends this info in their sip header as 'History-Info'. is there possibly another way that I could still get this info? I am just looking at the one function because that is what i found. |
| 22:10.21 | *** join/#asterisk bdfoster (~bdfoster@unaffiliated/bdfoster) |
| 22:11.30 | leifmadsen | I don't see why Asterisk would be stripping any headers, so if it isn't there then I'd suggest it isn't being sent |
| 22:12.05 | [T]ank | interesting perspective... i will look at it from that point of view too. Thanks leif |
| 22:18.49 | leifmadsen | you can verify with a tshark capture |
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| 23:28.47 | atan | Anyone here aware of a simple little SIP client which is built inside flash? |
| 23:29.08 | atan | I find tons of references to them on Google but most are owned by a company just for their service or are nowhere to be found |
| 23:32.42 | *** join/#asterisk ariel_ (~chatzilla@pdpc/supporter/active/abatista) |
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| 23:39.09 | Freeaqingme | atan, well, I probably am not allowed to say this, but you can just copy their applets, and if the address is hardcoded, should not be too hard to change |
| 23:39.20 | Freeaqingme | either way, if you find something open source, I'd be interested to hear about it ;) |
| 23:39.33 | atan | Freeaqingme! long time no see :) |
| 23:39.47 | *** join/#asterisk Zelic (~Zel@ip68-104-95-42.lv.lv.cox.net) |
| 23:39.48 | atan | I found one but I can't get the thing to work with SIP alone. Maybe you can take a look at it with me? |
| 23:39.57 | atan | http://code.google.com/p/red5phone/ |
| 23:40.16 | atan | Am I wrong to think this needs the red5 server? |
| 23:40.27 | atan | I can't understand what the Red5 server even is to be honest. |
| 23:41.22 | Freeaqingme | atan, no clue |
| 23:41.29 | Freeaqingme | the page suggewsts you do need a red5 server indeed |
| 23:41.32 | Freeaqingme | whatever red5 may be ;) |
| 23:42.04 | Zelic | Hello. I need assistance on resetting a password for an extension on the PBXbox. Can anyone please help me? |
| 23:42.24 | atan | Freeaqingme, then there is http://flashphoner.com/ but it's like $2000 |
| 23:44.46 | Freeaqingme | looked at http://code.google.com/p/siprtmp/ ? |
| 23:45.04 | atan | Hmm! |
| 23:45.12 | atan | I have now. Sweet. |
| 23:45.38 | p3nguin | zelic: That doesn't make very much sense to me. What is the actual problem? |
| 23:46.03 | Freeaqingme | atan, red5 is a flash streaming server btw |
| 23:48.07 | Freeaqingme | atan, try sparkweb |
| 23:48.21 | Zelic | THank you penguin. I'm sorry if it wasn't very clear. I have a PBX box setup with extensions for each phone. Each phone has an independent voicemail. Currently, most voicemails have the login as the extension: ie, 6999, password is 1236999. So if it was 7001, it would be 1237001. A previous employee of the company configured one of the extensions to have a different password. I would like to |
| 23:48.22 | Zelic | know if it is possible to reset this by logging into the PBX through the IP address on the browser, if there is a master extension I can log into.. or if there is a manual switch on the phone to reset the device. I hope this makes it more clear. |
| 23:49.17 | p3nguin | What do the extensions do that requires a password? |
| 23:49.50 | Zelic | It is to check the mailbox of voicemails |
| 23:50.27 | p3nguin | If you dialed extension 6999, you get voicemail for what phone? |
| 23:50.31 | Zelic | Unless the user logs in with Username and password on the phone, they won't be able to check their voicemail |
| 23:51.13 | p3nguin | If the only thing you want to do is change the voicemail password for a given mail box, that is done in voicemail.conf. |
| 23:51.34 | p3nguin | If that's all you want to do, you've made this much harder than it needs to be. |
| 23:52.04 | Zelic | how would I get to voicemail.conf? |
| 23:52.13 | Zelic | I'm sorry I am new to this technology |
| 23:53.14 | p3nguin | ssh to the Asterisk host. Open /etc/asterisk/voicemail.conf in any text editor that you can use. Make the change, and save the file. Exit. |
| 23:53.57 | Freeaqingme | if you dont know what text editor to use, use: nano /etc/asterisk/voicemail.conf |
| 23:55.20 | Zelic | Could you please assist me on starting the SSH process? |
| 23:55.25 | Zelic | I am not sure how to do that |
| 23:56.00 | p3nguin | What OS is your Asterisk system running? |
| 23:56.28 | Freeaqingme | Zelic, if your company is reliant on this pbx I'm not sure it would be smart to do this process with the little knowledge you have (no offense) |
| 23:58.07 | Zelic | none taken. Thank you. Our company recently had an IT change, the old IT company setup the PBX box, but there were internal security issues, so we migrated to a new IT company |
| 23:58.42 | Freeaqingme | have the new it company fix it? |
| 23:59.02 | Zelic | Unfortunately, the new company does not have the knowledge for the PBX system |
| 23:59.12 | Freeaqingme | you can open an ssh connection using putty btw (assuming you're on windows) |
| 23:59.21 | Freeaqingme | but you'll need a username/password, username is probably root |
| 23:59.25 | Zelic | I really need my employee to be up soon because she is one of the lawyer in my lawfirm |
| 23:59.33 | Zelic | and I can't wait for the IT company to get their act together |
| 23:59.39 | *** join/#asterisk LemensTS (~matthew@adsl-70-238-163-82.dsl.stlsmo.sbcglobal.net) |
| 23:59.52 | Zelic | lawyers* |