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00:51.56 | mpruett | anyone here? |
01:03.03 | Hmmhesays | not really |
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01:48.00 | Brijn | Anyone seen the D-Link DPH-540 already? |
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02:38.08 | TheCops | Agent can have multiple call at the same time ?! |
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02:39.20 | bhearsum | anyone here using a nortel i2002 (fw 0603B60) with asterisk? |
02:39.51 | bhearsum | i'm not getting any sound from my phone |
02:39.58 | bhearsum | but when i place calls the phone on the other end rings |
02:44.10 | VeNoMouS_ | lol i just had a rant on eyebeams forum |
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02:54.07 | tzafrir_laptop | anything wrong with the digium mailing lists? |
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02:59.58 | _murf_ | hmmmm. Late Sunday night(US) early monday morn (Europe) must be a slow time on this channel....! |
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03:03.55 | tengulre11 | Hi,all, which channel discuss H.323 protocol? |
03:09.04 | mrdigital | anyone use Zoom |
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03:38.47 | Klydal | anyone available to help a newb? Im trying to setup FWDout on my pbx and Im not sure how to make out going calls |
03:46.47 | P-NuT | Hey all. |
03:47.03 | P-NuT | Making outbound calls to other asterisk boxes... |
03:47.19 | P-NuT | is that done by making an extension and then dialing it? |
03:48.25 | P-NuT | like this? |
03:48.26 | P-NuT | exten => 750,1,Dial(IAX2/s@ipadresshere,30,t) |
03:48.26 | P-NuT | exten => 750,3,Hangup |
03:48.39 | P-NuT | is that right? |
03:51.39 | [TK]D-Fender | P-NuT : missing step 2 |
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03:58.52 | Dimitripietro | Anyone could help me compiling the sangoma drivers ? |
03:59.32 | mogorman | ./configure ; make ; make install ^_^ |
04:00.08 | Dimitripietro | You never compiled sangoma driver for sure :-) |
04:00.34 | mogorman | but you are right |
04:00.39 | mogorman | i dont have any sangoma hw |
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04:02.18 | websae_ | sagnoma is the BEST :) |
04:02.47 | mogorman | but i am biased |
04:02.49 | [TK]D-Fender | if mogorman were to say Sangoma 3 times his head would explode ;) |
04:02.54 | mogorman | working for their competitor |
04:03.05 | mogorman | lol [TK]D-Fender |
04:03.13 | mrdigital | [TK]D-Fender: ever hear of Zoom? |
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04:04.10 | Dimitripietro | <websae_> You are using sangoma driver ? |
04:04.53 | Dimitripietro | Have you ever had any error concerning udev ? |
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04:06.09 | mogorman | your rules are probabl bad Dimitripietro |
04:06.26 | Dimitripietro | I don't know anything about udev |
04:06.51 | mogorman | well i dont know anything about sangoma, but you probably have /etc/udev/rules.something |
04:06.59 | mogorman | and it needs to have correct entries |
04:07.35 | Dimitripietro | I do have /etc/udev/rules.d/ but I don't know what the correct entries look like |
04:07.54 | Dimitripietro | I was ruinning my system with a TDM400 and everything was fine |
04:07.57 | mogorman | there is probably sample in sangoma source |
04:09.37 | Dimitripietro | See as there is a samples, i'M gonan take a llok |
04:09.39 | Dimitripietro | look |
04:10.14 | mogorman | okies |
04:10.20 | mogorman | hope you can get it working |
04:11.04 | [TK]D-Fender | mrdigital : Yup |
04:12.09 | Dimitripietro | Everything from the sample is already present in the udev rules file :-( |
04:12.48 | mogorman | only thing i can think of is your udev is yuck |
04:12.52 | mogorman | but id have to see |
04:14.31 | russellb | you will not see! |
04:14.37 | mogorman | well yeah |
04:14.41 | mogorman | im about to go to bed |
04:14.43 | russellb | :) |
04:14.44 | mogorman | tired |
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04:17.53 | P-NuT | hey all, |
04:18.10 | P-NuT | who can dial my FWD number for me? |
04:19.58 | Qwell | P-NuT: the fwd callback app can |
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04:28.45 | P-NuT | yeah, I tried that, but am getting nothing.. |
04:29.02 | P-NuT | so I was wondering if somebody could dial me and tell me what error they get. |
04:29.12 | P-NuT | maybe I've not got soemthing right. |
04:30.33 | NotJohnDavid | hm |
04:30.56 | P-NuT | hm? |
04:31.59 | NotJohnDavid | dial you for what? |
04:32.36 | littleball | hello, i am looking for small size box for linux, who can suggest? |
04:38.16 | P-NuT | NotJohnDavid: Well, I tried it from the website, and it never called. So I was wondering if someone could try dialing my number and tell me what result they get. |
04:38.24 | P-NuT | NotJohnDavid: Make sense? |
04:39.30 | NotJohnDavid | what is your # |
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04:44.44 | variable_office | how can you change the voicemail message? |
04:50.08 | littleball | hello, anyone using spandsp to send out fax? |
04:54.25 | VeNoMouS_ | <PROTECTED> |
04:54.25 | VeNoMouS_ | Jul 3 16:48:08 WARNING[10138]: app_getdtmf.c:95 getdtmf_exec: GET_DTMF_DIGITS was set to [54654] |
04:54.25 | VeNoMouS_ | <PROTECTED> |
04:54.25 | VeNoMouS_ | <PROTECTED> |
04:54.25 | VeNoMouS_ | <PROTECTED> |
04:54.30 | VeNoMouS_ | err shit wrong window |
04:55.30 | mpruett | Russelb I have an odd one for you |
04:55.30 | Sedorox | darn.. no passwords :p |
04:55.46 | VeNoMouS_ | Sedorox : lol |
04:56.39 | mpruett | I have two boxes setup (I believe) exactly the same |
04:57.33 | mpruett | On one box when I make a sip connection you can't hear the other side when they speak but they can hear you |
04:57.49 | mpruett | The other box works fine |
04:58.01 | VeNoMouS_ | checked your firewall rules/ |
04:58.05 | VeNoMouS_ | rtp allowed in both ways? |
04:58.24 | mpruett | If both callers use Meetme both sides can communicate fine |
04:58.32 | VeNoMouS_ | sip would've established the call, but if only one rtp is going |
04:58.39 | VeNoMouS_ | that would say to me your blocking a stream |
04:58.43 | mpruett | I turned of my firewall - problem still exist |
04:59.00 | VeNoMouS_ | yes but meetme is a conf |
04:59.15 | VeNoMouS_ | it dont try neg the streams |
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05:00.22 | mpruett | VeNoMous: So your thinking a firewall is blocking me somewhere? |
05:00.53 | mpruett | I don't believe my CoLo is filtering me but I could check |
05:01.19 | mpruett | The box that is working fine is local the one that isn't is at CoLo |
05:03.56 | VeNoMouS_ | rtp debug ip <hostnamehere> |
05:04.32 | *** part/#asterisk littleball (n=littleba@26.203-123-30.leased.qala.com.sg) |
05:06.16 | VeNoMouS_ | well |
05:06.21 | VeNoMouS_ | err wrong window ffs |
05:07.03 | mpruett | VeNoMous: You didn't mean that for me correct? |
05:07.52 | VeNoMouS_ | the rtp |
05:07.52 | VeNoMouS_ | yes |
05:07.59 | VeNoMouS_ | the "well" no |
05:08.14 | mpruett | ok let me try that |
05:15.56 | mpruett | OK now the stupid question |
05:16.08 | mpruett | How do I use this command |
05:16.40 | mpruett | All I get is the following |
05:16.41 | mpruett | Usage: rtp debug [ip host[:port]] |
05:16.41 | mpruett | <PROTECTED> |
05:17.53 | mpruett | In the ip host:port - I put the IP of the ATA I am using and 10000 as the port |
05:22.50 | mpruett | VeNoMous: I turned on rtp debug |
05:23.45 | mpruett | VeNoMous: then I made a call - I can see RTP packets to and from host |
05:34.28 | VeNoMouS_ | on both sides? |
05:34.40 | VeNoMouS_ | i mean on both boxes? |
05:34.44 | mpruett | Yes |
05:36.18 | mpruett | Another clue - I just made a call and for a couple seconds it worked fine then it went back to how I described b4 in mid call |
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05:37.59 | VeNoMouS_ | err in mid call? |
05:38.21 | VeNoMouS_ | weird |
05:38.35 | mpruett | Yes we could hear each other fine for a couple seconds then I couldn't hear him talking but he could still hear me |
05:39.38 | VeNoMouS_ | <mpruett> In the ip host:port - I put the IP of the ATA I am using and 10000 as the port |
05:39.40 | VeNoMouS_ | no |
05:39.46 | VeNoMouS_ | u should use the ip of the remote machine |
05:40.46 | mpruett | OK - will it give me different info that just "rtp debug"? |
05:40.54 | mpruett | s/than/that |
05:41.25 | mpruett | Let me try |
05:42.58 | muppetmaster | <PROTECTED> |
05:43.02 | muppetmaster | <PROTECTED> |
05:43.03 | muppetmaster | <PROTECTED> |
05:43.03 | muppetmaster | azz |
05:43.05 | muppetmaster | <PROTECTED> |
05:44.26 | muppetmaster | Apologies, my son got a hold of the keyboard |
05:45.54 | russellb | that's awesome :) |
05:46.01 | mpruett | lol - and he is already saying dirty words!! A techy in the works! |
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06:05.21 | stephane_ | jour |
06:09.28 | P-NuT | bonjour. |
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06:24.17 | muppetmaster | Anyone else here have the Nokia E61? |
06:25.01 | file | in an alternate universe I do |
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06:25.46 | muppetmaster | We all do in that case |
06:26.42 | Corydon76-home | but only in separate parallel universes |
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06:28.08 | VeNoMouS_ | <PROTECTED> |
06:28.08 | VeNoMouS_ | Jul 3 18:21:40 WARNING[16012]: channel.c:787 channel_find_locked: Avoided initial deadlock for '0x8143078', 10 retries! |
06:28.08 | VeNoMouS_ | <PROTECTED> |
06:28.08 | VeNoMouS_ | Jul 3 18:21:48 WARNING[16029]: app_getdtmf.c:73 getdtmf_exec: Digits Entered Were [123456987**789***000**7] |
06:28.08 | VeNoMouS_ | Jul 3 18:21:48 WARNING[16029]: app_getdtmf.c:78 getdtmf_exec: Digit Limit set of [3] |
06:28.10 | VeNoMouS_ | Jul 3 18:21:48 WARNING[16029]: app_getdtmf.c:81 getdtmf_exec: GET_DTMF_DIGITS was set to [123] |
06:28.12 | VeNoMouS_ | arse |
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06:28.28 | VeNoMouS_ | thats the prob with having to bitchx's open |
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06:39.41 | blkremedy | does anyone here know of a simple program that can be used to clone a asterisk install. |
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06:41.39 | VeNoMouS_ | dd |
06:41.41 | VeNoMouS_ | :P |
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06:50.09 | SplasPood | does AEL1 support any type of #include style syntax? |
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06:53.54 | P-NuT | Hey all, if I try to call anyone from my IAX extension, (external to my network) I get congestion and it just hangs up.. What am I doing wrong? |
06:54.23 | P-NuT | I have port 4569 open, do I need anything else? |
06:54.39 | L|NUX | allow guest |
06:54.39 | L|NUX | :) |
06:54.58 | P-NuT | hmm.. |
06:55.02 | P-NuT | I thougth I did... |
06:55.04 | P-NuT | lets see.. |
06:55.10 | P-NuT | of |
06:55.13 | P-NuT | oh ok, |
06:55.31 | P-NuT | so in iax.conf under general, I want allow=guest ? |
06:55.32 | L|NUX | okay |
06:55.50 | P-NuT | yeah? |
06:56.05 | L|NUX | na |
06:56.07 | L|NUX | sip.conf |
06:56.12 | P-NuT | oh... |
06:56.17 | P-NuT | um.. |
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06:56.24 | P-NuT | why sip.conf? |
06:56.36 | L|NUX | because sip.conf have it |
06:56.40 | P-NuT | oh ok. |
06:56.42 | P-NuT | thsanks |
06:56.45 | L|NUX | basically you can do like this |
06:56.51 | L|NUX | add a user in your iax.conf like this |
06:56.55 | L|NUX | [linux] |
06:57.00 | L|NUX | type = friend |
06:57.04 | L|NUX | username = linux |
06:57.12 | L|NUX | secret = ****** |
06:57.16 | L|NUX | host = dynamic |
06:57.26 | L|NUX | context = default |
06:57.35 | L|NUX | callerid = "Linux Calling" |
06:57.45 | L|NUX | and when you want to dial to your network |
06:57.48 | L|NUX | you can dial using this |
06:57.53 | L|NUX | in your extensions.conf |
06:58.37 | L|NUX | exten => 1,1,Dial(linux:******@host/exten,60,tr) |
06:58.44 | L|NUX | simple :> |
06:58.48 | P-NuT | hmm.... |
06:58.51 | P-NuT | ok then... |
06:58.57 | L|NUX | reload |
06:58.59 | L|NUX | and then call |
06:59.54 | P-NuT | do I need secret in there? |
06:59.58 | L|NUX | yeah |
07:00.00 | L|NUX | you need |
07:00.01 | L|NUX | :) |
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07:01.11 | P-NuT | ok, so in ,Dial(linux:******@host/exten,60,tr) linux:****** is the username and password. |
07:01.31 | L|NUX | yeah |
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07:02.14 | P-NuT | so if I wanted to call digium lets say, it would be exten => 1,1,Dial(linux:******@misery.digium.com/6000,60,tr) |
07:02.17 | P-NuT | is that right? |
07:04.07 | P-NuT | and also, where in sip.conf do I put the allow=guest. [General] or [authentication] ?? |
07:04.43 | L|NUX | yeah |
07:05.00 | L|NUX | well see general section of your sip.conf |
07:05.24 | SheriF_WorK | what kind of softphone supports G729 or G726 or G723.1 and also supports SIP INFO in DTMF mode ? |
07:06.09 | VeNoMouS_ | eyebeam |
07:07.02 | L|NUX | SheriF_WorK : but g723 is not for sale |
07:07.22 | L|NUX | SheriF_WorK : you can get it if you buy 250 + copies |
07:07.24 | SheriF_WorK | L|NUX: there is an opensource G729 and G723 |
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07:07.40 | L|NUX | SheriF_WorK : codec is different thingy |
07:07.48 | L|NUX | SheriF_WorK : but softphone is different |
07:08.06 | SheriF_WorK | L|NUX: yes :-s good point ok i want a phone supports G729 :-s |
07:08.19 | L|NUX | SheriF_WorK : eyebeam |
07:08.31 | SheriF_WorK | L|NUX: any other options ? |
07:08.34 | muppetmaster | LINUX: G729 is proprietary and licensed. |
07:08.37 | muppetmaster | No way around that |
07:08.56 | muppetmaster | You may only get a binary from Digium or others |
07:09.08 | SheriF_WorK | http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/ |
07:09.31 | L|NUX | muppetmaster : i know but there is some open source thingy arround with intel ipc .... |
07:09.42 | L|NUX | as SheriF_WorK given link |
07:09.51 | muppetmaster | LINUX I don't think so, if so, it would be a violation of licensing for G729 |
07:10.06 | L|NUX | SheriF_WorK : use X-pro |
07:10.16 | SheriF_WorK | Please note that this code is available for you to download for education purposes only |
07:10.23 | L|NUX | muppetmaster : might be but no one keep this sites down |
07:11.17 | SheriF_WorK | L|NUX: no other options for a softphone ? |
07:11.35 | L|NUX | well there are some opensource :0 |
07:12.39 | L|NUX | SheriF_WorK : http://www.portsip.com/ |
07:12.50 | L|NUX | this have g723 + g729 :) |
07:14.42 | DrkShdw | L|NUX: basically, you are getting a multi-thousand dollar PBX system for free. don't go pirating the g729 codecs. hell, it's only $10/channel. |
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07:15.15 | DrkShdw | Sorry. I stayed quiet long enough. :) |
07:15.52 | L|NUX | DrkShdw : did i said any thing :( |
07:16.06 | L|NUX | DrkShdw : i am not the person i just said yes i have seen |
07:16.23 | L|NUX | DrkShdw : i my self have digium g729 license |
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07:30.10 | SplasPood | Can anyone point me in the right direction as to how I can use Asterisk realtime static meetme.conf ? |
07:33.02 | Bert- | I bought one but as I'm student, I could have used G729 without paying for taht, no ? |
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07:33.47 | *** join/#asterisk tparcina (n=tparcina@lns01-0003.dsl.iskon.hr) |
07:34.07 | tparcina | hi channel! |
07:34.21 | Bert- | and another question : If I want to put my asterisk on a new computer, what about the license I bought ? |
07:34.24 | Bert- | is it portable ? |
07:34.28 | Bert- | hi tparcina |
07:35.01 | tparcina | i head beautifull weak in brussel. it's realy a nice town. hopefully i'll go back thate again... |
07:35.25 | *** part/#asterisk angom_h (n=angom@red-corp-200.76.251.26.telnor.net) |
07:35.30 | DrkShdw | Bert-: you can transfer the license to another machine once. after that, you have to contact digium |
07:37.22 | Bert- | okay |
07:37.31 | Bert- | anyway, it is only 10$ :) |
07:37.57 | DrkShdw | per channel, yes |
07:38.03 | Bert- | yep |
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07:38.18 | Bert- | and I bought one for the compagny I'm working for (I'm a trainee) |
07:38.31 | Bert- | but I don't want to give them my own license |
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07:44.05 | tparcina | hi Bert, it's quite today - everybody goone to vacation? |
07:47.05 | Bert- | ?? |
07:47.17 | Bert- | from my side, I've to go to work :( |
07:47.26 | Bert- | have a good day here :) |
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07:57.48 | muppetmaster | So, I posted this on the E61 @ Nokia: http://discussion.forum.nokia.com/forum/showthread.php?t=83985 |
07:57.57 | muppetmaster | About to take it back to FNAC if I can not get inbound working on this thing. |
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08:18.16 | _4d4m_ | hi all.. anyone able to offer some advice on billing app's for *? |
08:19.23 | _4d4m_ | have looked at plenty of options but am not sure which is right for me |
08:19.40 | _4d4m_ | couple of hundred pre-pay accounts to handle |
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08:20.17 | _4d4m_ | nothing too flash.. not bothered about UI.. something lightweight, flexible, and known to work well. |
08:20.38 | _4d4m_ | am running * realtime static, flat csv cdr at the mo |
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08:41.26 | hwt | has anybody managed to compile spandsp rxfax on recent asterisk? |
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09:09.15 | Bert- | hop |
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09:09.53 | TeePOG | good morning |
09:10.04 | TeePOG | Hi FuriousGeorge! |
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09:42.41 | Bert- | according to you guys, what is the "best" gui to use asterisk ?? |
09:43.16 | dpryo | vim |
09:43.21 | dpryo | and asterisk -r |
09:43.39 | Bert- | hahaha |
09:43.42 | Bert- | I totally agree |
09:44.08 | Bert- | but I want a kind f graphical interface to add/remove a phone, etc ... |
09:44.25 | Bert- | beacause I not sure to work for my company for ever :) |
09:46.51 | hwt | Bert-: there aren't any good, IMHO. |
09:47.00 | hwt | Bert-: for stock *. |
09:48.46 | *** part/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
09:50.34 | Bert- | I tried freepbx, but it never worked wih Free account |
09:50.53 | Bert- | and Trixbox ... how to say ... it sux |
09:50.56 | Bert- | :) |
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09:54.34 | dlynes_laptop | Good morning, peeps |
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10:26.39 | frenzy | hey all |
10:26.57 | frenzy | where can I get standard pbx voice overs? |
10:27.05 | frenzy | the free stuff |
10:27.29 | MrChimpy | get a microphone and speak into it |
10:27.40 | frenzy | hahaha |
10:28.04 | MrChimpy | there's the standard asterisk-sounds set |
10:28.12 | MrChimpy | same place you got asterisk |
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10:30.47 | damned | can anybody suggest if asterisk is technically able to initiate voice call and if it would be taken - connect person to another number? |
10:31.27 | dlynes_laptop | damned: read up on the wiki about how to do call files |
10:32.44 | mosty | damned: i have seen click to call firefox extenions that use asterisk |
10:32.56 | mosty | so i guess so |
10:33.04 | damned | dlynes_home: the actual question is about ability to force asterisk to call from a command or a script. |
10:33.28 | dlynes_laptop | damned: yeah...like I said...use call files |
10:33.38 | dlynes_laptop | damned: you can use a script to generate them |
10:33.46 | damned | dlynes_home: ok. thnx. |
10:33.55 | dlynes_laptop | damned: asterisk checks the spool directory for the call files periodically |
10:34.00 | VeNoMouS_ | damned : perl script |
10:34.12 | dlynes_laptop | damned: when it finds one, it automates a process based on the info in that call file |
10:34.21 | VeNoMouS_ | http://search.cpan.org/~jhiver/Asterisk-LCR-0.08/lib/Asterisk/LCR/Dialer.pm |
10:36.42 | MrChimpy | aye, i've used call files. easy. |
10:38.07 | MrChimpy | like the music on hold feature, but a looped sample which starts when the dial starts |
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10:43.12 | dlynes_laptop | MrChimpy: why not write it yourself? |
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10:54.06 | *** join/#asterisk MatsK (n=mats@141.221.181.62.in-addr.dgcsystems.net) |
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11:09.37 | MatsK | Is there anyone that has experience with pattern matching with letters instead of numbers ? |
11:10.02 | MatsK | example: exten => _smith,1,Answer |
11:12.33 | MatsK | and how is dotts interpreted in letter pattern, example: exten => _john.smith,1,Answer |
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11:14.57 | dlynes_laptop | MatsK: probably \. |
11:16.42 | MatsK | you mean that it's possible to escape it ? |
11:17.01 | *** part/#asterisk P-NuT (n=P-Nut@CPE-60-227-93-75.nsw.bigpond.net.au) |
11:18.28 | kay2 | u can't |
11:19.19 | MatsK | kay2: Thx I thought so, I have to ask for that feature then ;-) |
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11:28.49 | MrChimpy | dlynes_laptop: that's what I meant when I said "MrChimpy suspects he will be writing it himself" |
11:29.19 | MrChimpy | why not write it myself? it'd be a bit silly to re-do it if someone has done it already. |
11:31.36 | __chris | when storing callerID names/numbers in * using database put cidname how are these actually stored? I'm trying to back them all up but can't seem to find them anywhere, grep knownstorednumber doesn't return anything either |
11:34.08 | vgster | astdb |
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11:57.30 | knobo | I have a diva 2.02 PCI card. lspci shows it to me, but I'm not shure which tools to use to configure it |
11:58.00 | knobo | capiinfo says "capi not installed" |
11:58.25 | knobo | capi modules are loaed. I can se them with lsmod |
11:58.53 | knobo | divacapi is loaded and in use by the kernelcapi module |
12:04.23 | MrChimpy | wtf? |
12:04.38 | MrChimpy | 1.2.9.1 doesn't build! |
12:04.47 | MrChimpy | make[1]: Leaving directory `/root/asterisk-1.2.9.1/channels' |
12:04.55 | MrChimpy | xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx |
12:05.05 | MrChimpy | chan_zap.c: In function `pri_dchannel': |
12:05.05 | MrChimpy | chan_zap.c:9038: error: structure has no member named `call' |
12:05.07 | muppetmaster | MrChimpy - Build for me |
12:05.07 | *** join/#asterisk cjk (n=cjk@80.92.64.103) |
12:05.15 | muppetmaster | I do have a problem with SVN TRUNK though.... |
12:05.18 | MrChimpy | i grabbed the latest zaptel in case that fixed it |
12:05.19 | muppetmaster | Pretty unstable |
12:05.28 | cjk | hi, does anyone know a good website to do reverse number lookups? |
12:05.31 | MrChimpy | 1.2.9.1 unstable? |
12:05.45 | muppetmaster | MrChimpy - No, 1.2.9.1 is fine, SVN TRUNK is unstable |
12:05.52 | MrChimpy | ah, ok |
12:06.18 | MrChimpy | quick google said it builds without zaptel and iax trunking |
12:06.29 | muppetmaster | I was at Astricon last week and Kevin Flemming said they are trying to get v1.4 beta out this week. Seems a stretch. |
12:06.30 | *** join/#asterisk fenlander (n=fenlande@82.152.81.57) |
12:06.33 | MrChimpy | which are the two major portions that my application uses |
12:07.23 | MrChimpy | i think i may end up sticking with a crusty old version and forgetting about submitting stuff to digium |
12:08.18 | *** join/#asterisk rogier (n=rogier@83-67-dsl.ipact.nl) |
12:08.36 | MrChimpy | ah |
12:08.39 | MatsK | MrChimpy: And after that will it be more stable ?! |
12:08.46 | MrChimpy | i need to update libpri too. |
12:08.56 | MrChimpy | matsk: it's stable with what i'm using now |
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12:11.36 | MrChimpy | at least if i screw it up i know what i've done :) |
12:12.28 | MrChimpy | live platform is too critical to go playing upgrade frenzy on |
12:17.14 | MatsK | Well, I have a "dogfood" platform that is upgraded first and then after a test period is the production platfrom upgraded. |
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12:25.48 | knobo | anyone has a eicon Diva card? |
12:26.37 | MrChimpy | testing for me is something of a problem, as we have very heavy call volumes which I can't replicate - so any upgrade is a case of trying it and having a spare pair of trousers for when it goes wrong |
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12:31.18 | MatsK | I just use SIPP to simulate some load and we have also some inhouse usage on the dogfood platform so it's not impossible to emulate your heavy call volume ;-) |
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12:32.07 | mosty | __chris: they can be stored in sip.conf or iax.conf for those types of clients, or set on the clients themselves, i think |
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12:59.20 | [TK]D-Fender | Katty: Mew |
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13:08.20 | Katty | morning mister fender (= |
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13:16.52 | trelane_ | I'm having severe IRQ problems, is there a way to manually steer interrupts via insmod? |
13:17.22 | VeNoMouS_ | only if the module supports it |
13:17.23 | MooingLemur | lemme guess.. a dell server :P |
13:17.28 | VeNoMouS_ | ie its been written into the code |
13:17.40 | VeNoMouS_ | MooingLemur comming it has to be a ibm! |
13:17.53 | VeNoMouS_ | ibm == its buggered mate |
13:18.14 | VeNoMouS_ | hp dl 140 > * |
13:19.05 | MooingLemur | I think that's biting us too. Our asterisk boxes are dell 1750s, and they get pretty choppy with disk access |
13:19.24 | trelane_ | MooingLemur, how'd you know? |
13:19.33 | VeNoMouS_ | what kernel u guys running? |
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13:19.57 | MooingLemur | dells tend to route everything to one IRQ |
13:20.02 | trelane_ | MooingLemur, I'm runing on 830's |
13:20.05 | MooingLemur | pain in the butt |
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13:20.23 | skraelings001 | hi everyone |
13:20.27 | trelane_ | MooingLemur, I've got most of the bios steering disabled and am manually assigning IRQ's however network and smbus and raid and ide seem to want the same IRQ's |
13:20.30 | VeNoMouS_ | better yet, time to go smoke a fat one |
13:21.49 | trelane_ | MooingLemur, I can get asterisk working or I can get a stable system (the other option is to put the wctdm24xxp against the ide controller |
13:22.18 | skraelings001 | i'm trying to do an agi application and so far i don't know what format channelname should have. anyone know? is this like the output in show channels ? |
13:22.44 | skraelings001 | the command is CHANNEL STATUS [channelname] |
13:22.44 | VeNoMouS_ | well what u writing it in genious? |
13:22.50 | VeNoMouS_ | c,perl, python? |
13:22.54 | MooingLemur | I don't really have the answer. The only problem I'm having is with rxfax and IRQs screwing that up |
13:22.56 | skraelings001 | python |
13:23.09 | VeNoMouS_ | dont know python ure out of luck |
13:23.11 | MooingLemur | shared IRQs that is |
13:23.38 | skraelings001 | i didn't give reference, cause i don't think is language-dependent |
13:23.39 | VeNoMouS_ | and the channel is the rtp stream |
13:23.45 | trelane_ | MooingLemur, what's sharing for you? |
13:25.01 | MooingLemur | digium card and scsi controller |
13:25.04 | VeNoMouS_ | skeffling |
13:25.05 | VeNoMouS_ | http://home.cogeco.ca/~camstuff/agi.html#CHANNELSTATUS |
13:25.12 | skraelings001 | i'm thinking is like for everycall channelname's format is SIP/101-234w3fs or IAX2/102-239fsd |
13:25.13 | VeNoMouS_ | ^^ pyython agi |
13:25.17 | trelane_ | MooingLemur, both are discrete cards? (physical hardware and not onboard) |
13:25.21 | VeNoMouS_ | CHANNEL STATUS Zap/9-1 |
13:25.21 | VeNoMouS_ | Return the status of channel Zap/9-1 |
13:25.30 | MooingLemur | scsi is onboard |
13:25.48 | VeNoMouS_ | its amazing when you type in "asterisk CHANNEL STATUS python" into google |
13:25.49 | trelane_ | MooingLemur, there's options in bios to move irq's |
13:26.17 | MooingLemur | I'm on vacation anyway.. I'll deal with that when I get back home :P |
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13:26.51 | trelane_ | MooingLemur, I'M NOT! |
13:26.52 | trelane_ | :-p |
13:27.53 | skraelings001 | VeNoMouS_ : i guess so, thanks |
13:34.46 | VeNoMouS_ | np |
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13:36.08 | skraelings001 | VeNoMouS_ : it's what i expected, it would work with zap, but sip or iax assign a random set of numbers and letters to tech/ext, this is the problem |
13:37.39 | VeNoMouS_ | skraelings001 not really |
13:37.41 | VeNoMouS_ | because if you had read |
13:37.44 | VeNoMouS_ | u would have seen |
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13:37.45 | VeNoMouS_ | Return the status of the specified channel. If no channel name is specified, return the status of the current channel. |
13:38.12 | VeNoMouS_ | unless u want a certain channel |
13:38.35 | skraelings001 | VeNoMouS_ : certain channel |
13:40.30 | VeNoMouS_ | skraelings001 so call the var # ${CHANNEL} |
13:40.33 | VeNoMouS_ | skraelings001 so call the var ${CHANNEL} |
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13:55.38 | BertZ | how to see which codec is used for an active call please ? |
13:57.44 | a1fa | Anybody know a provider that sells $1did + 1c/min? |
13:59.18 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
13:59.26 | BertZ | hmm |
13:59.34 | BertZ | who works with G79 codecs here plz ? |
13:59.45 | BertZ | I have a lot of errors about VAD |
14:00.18 | malcolmd | turn off VAD on your endpoint or the connecting gateway |
14:00.23 | malcolmd | VAD isn't supported by Asterisk |
14:01.36 | BertZ | hmm |
14:01.55 | BertZ | VAD is not supported by me , as I don't know what is it :) |
14:02.17 | coppice | Voice Activity Detection |
14:02.19 | *** join/#asterisk Pepse (n=pepse@ip68-109-169-37.ph.ph.cox.net) |
14:02.23 | BertZ | okay |
14:02.24 | BertZ | thx :) |
14:04.04 | BertZ | hm |
14:04.40 | BertZ | there is no options about VAD on my endpoint. It is a Nextone SoftSwitch. Does someone ever worked with asterisk, G729 and Nextone plz ? |
14:06.06 | coppice | its usually selectable. maybe they call it something else in the config |
14:09.21 | BertZ | no sip option to specify no VAD in asterisk ?? |
14:09.35 | *** join/#asterisk Hmmhesays (i=negative@66.173.103.110) |
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14:11.07 | BertZ | supoprt said to me that I've to disable VAD on the GW, not on the Nextone |
14:11.18 | BertZ | but I see no option about VAD in Asterisk : |
14:11.20 | BertZ | :( |
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14:12.22 | coppice | there's no option for VAD in Asterisk, as it doesn't have support for VAD |
14:12.55 | BertZ | I use a Grand Stream maybe there is an option ... |
14:13.24 | *** join/#asterisk vgster (n=vgster@host217-45-221-53.in-addr.btopenworld.com) |
14:13.37 | Hmmhesays | likely assumption |
14:13.57 | Hmmhesays | If you "need" to use VAD, here is a clue... don't use VOIP |
14:14.07 | HuSoft | Is there a way I can make asterisk recognize calls using context names? for example, a user normally calls with the url, sip:1234@10.0.0.3, Is it possible to call: sip:husoft@10.0.0.3 ? (husoft is the context name for the extension 1234). |
14:14.22 | coppice | there's nothing wrong with true VAD |
14:14.23 | Hmmhesays | of sorts |
14:14.24 | *** join/#asterisk Qb3rt (n=jhgjkgui@kyle.colba.net) |
14:14.40 | Qb3rt | my litle problem (question) ---> http://pastebin.ca/77848 |
14:15.26 | Hmmhesays | yeah but generally people who are asking about it have something like a 14.4k connection with about 1500ms ping between endpoints |
14:15.32 | BertZ | hmmm on voip-info.org, G723 and G729 must be disabled :( |
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14:16.00 | BertZ | when I disable G723/G29, all works fine :( |
14:16.16 | BertZ | but I want to use it, as it need less bandwith |
14:16.27 | Hmmhesays | Then you need to read more |
14:16.34 | BertZ | sure |
14:16.46 | Hmmhesays | for g.729 you either need a license, or use it passthru only |
14:16.51 | Hmmhesays | g.723 is passthru only |
14:17.30 | rob0 | Qb3rt: if [ -z "$1" ] ; then exit 1 ; fi |
14:17.50 | rob0 | you could also test to be sure it's a directory |
14:17.56 | rob0 | "help test" |
14:17.58 | BertZ | Hmmhesays : Grand Stream has G723/G729, so no problem for passthrough |
14:18.07 | BertZ | I bought a G729 license from digium |
14:18.30 | *** join/#asterisk kay2 (n=ashdown@sd-420.dedibox.fr) |
14:18.31 | BertZ | but I can test G723 and G729, as I'm only testing asterisk. |
14:18.42 | Hmmhesays | That last sentence made no sense |
14:18.50 | BertZ | ? |
14:19.11 | BertZ | I'm a student, I'm trying to see how asterisk works, then make a summary of it |
14:19.20 | BertZ | I don't have to pay for G729 |
14:19.30 | BertZ | written on digium's website if I remember |
14:19.36 | BertZ | anyway |
14:19.40 | BertZ | I bought my own license |
14:19.44 | BertZ | I want it to work |
14:19.52 | BertZ | the problem is VAD, not G729 license |
14:19.59 | BertZ | I'll find :) |
14:20.02 | HuSoft | Is there a way I can make asterisk recognize calls using the extension description? for example, a user normally calls with the url, sip:1234@10.0.0.3, Is it possible to call: sip:husoft@10.0.0.3, sip:voicemail@10.0.0.3, etc...? |
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14:21.41 | Qb3rt | rob0 thanks |
14:22.42 | mosty | husoft: yes, via the dialplan |
14:23.37 | Hmmhesays | I love it when people ask the same question over and over again |
14:23.41 | Hmmhesays | even though I answered it |
14:23.47 | Hmmhesays | HuSoft: yes |
14:23.54 | trelane_ | Hmmhesays, it's called generating extra content :) |
14:24.02 | mosty | husoft: use Goto to jump to a different context and Dial to dial a particular device |
14:24.15 | trelane_ | Hmmhesays, it's called saying the same thing over and over again because the old grey matter upstairs died years ago |
14:24.22 | trelane_ | ;) |
14:24.32 | Hmmhesays | I have some friends like that, too many drugs in college |
14:24.59 | HuSoft | mosty, ok, thanks. |
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14:25.04 | iq | Hi |
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14:28.31 | rene- | hi, i am looking for steve totaro of totaro tech and asteriskhelpdesk, has anyone seen it? his web pages have disappeared this side of the internet? i am hoping someone knows his irc handle |
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14:33.26 | cytrak | is there a way to review the recorded grettings without having to always re-record them |
14:35.42 | a1fa | Anybody know a provider that sells $1did + 1c/min? |
14:36.06 | Nivex | no, but I bet the wiki might |
14:36.14 | a1fa | couldnt find it |
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14:37.45 | trelane_ | do the zaptel modules support IRQ steering, I'm still working around an IRQ conflict on this damnable dell server |
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14:40.34 | cytrak | trelane: check which devices are using IRQs and disable them on the BIOS |
14:41.03 | a1fa | maybe he needs those |
14:41.04 | cytrak | trelane: that what I did on mine, I got no usb , serial, parralel , they are all disabled |
14:41.20 | a1fa | trelane_ : hey |
14:41.25 | a1fa | just switch irqs in bios |
14:41.27 | a1fa | etc |
14:42.00 | cytrak | well my bios didn't give me the option to actually assign irqs |
14:42.05 | cytrak | i wish I had |
14:42.07 | a1fa | update bios |
14:42.18 | cytrak | even with that |
14:43.02 | cytrak | do you know if there is a way to review the recorded grettings without having to always re-record them ? |
14:43.08 | *** join/#asterisk nfi|ermes (n=nfi_erme@217.220.121.62) |
14:43.15 | nfi|ermes | hi all |
14:43.22 | cytrak | my users bitch so much about all these little things |
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14:44.02 | [TK]D-Fender | cytrak : You mean VM announcements? |
14:44.16 | trelane_ | alfa: been there done that, I'm suffering from braindead hardware |
14:44.37 | cytrak | yeah I guess so , I'm talking about the busy and unavailable messages that you record |
14:44.44 | trelane_ | alfa: this dell likes moving the smbus with the zap card |
14:45.11 | cytrak | [TK]D-Fender: was that what you meant ? |
14:45.35 | [TK]D-Fender | cytrak : You could jsut make your own little dialplan script to listen to them |
14:45.37 | a1fa | trelane_: it doesnt have a jumper on it |
14:45.55 | trelane_ | a1fa, no |
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14:46.15 | a1fa | hehe |
14:46.18 | a1fa | sorry dude |
14:46.18 | trelane_ | it's a server board, I can steer IRQ's in bios but regardless of what I do the smbus controller follows the pci slot that the zap card's in |
14:46.21 | a1fa | disable something in bios |
14:46.32 | trelane_ | everything that can be disabledi s |
14:46.33 | a1fa | disable that esd |
14:46.37 | a1fa | or whatchmaqall it |
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14:47.08 | cytrak | [TK]D-Fender: ok , so the voicemail app won't be able to |
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14:48.03 | cytrak | [TK]D-Fender: I thought about doing that but then I know I'm gonna hear a lot of bitching about "oh we got call another extension to be able to do that .." |
14:48.17 | a1fa | [TK]D-Fender : you are the one who recomended that $1 did + 1c a minute |
14:48.53 | [TK]D-Fender | a1fa : nope. |
14:48.58 | a1fa | yeah it was you |
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14:50.19 | a1fa | i no lie, no lie |
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14:50.54 | [TK]D-Fender | a1fa : Sorry, I don't do PSTN termination like that...l don't have anyone to recommend to anyone really, so couldn't be me. |
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14:53.12 | nfi|ermes | i would like to enable the caller to transfer the call, but this doesn t happen |
14:53.31 | Hmmhesays | what a fantastically vague statement |
14:54.02 | nfi|ermes | lol |
14:55.40 | RoyK[at] | ~nickometer RoyK[at] |
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14:57.14 | knobo | anyone knows how asterisk 1.0.7 works with isdn4linux? |
14:58.16 | knobo | or if it is better to use capi |
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14:59.20 | RoyK[at] | at least |
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15:00.47 | RoyK[at] | knobo: any particular reason you want to use 1.0.7? or isdn4linux? |
15:01.19 | *** part/#asterisk RoyK[at] (n=roy@80.109.196.173) |
15:01.21 | knobo | RoyK[at]: People force me to use debian stable |
15:01.33 | *** join/#asterisk RoyK[at] (n=roy@chello080109196173.3.graz.surfer.at) |
15:01.45 | RoyK[at] | knobo: any particular reason you want to use 1.0.7? or isdn4linux? |
15:02.10 | RoyK[at] | knobo: 1.0.7 is quite old, and i4l is perhaps some of the worst shite I've ever touched |
15:02.12 | *** join/#asterisk visba (n=dca[lapt@sta-208-139-193-162.rockynet.com) |
15:02.20 | RoyK[at] | use bristuff or visdn instead. perhaps capi |
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15:02.48 | knobo | well, i4l is the the only way to get the eicon Diva 2.02 card going |
15:02.56 | knobo | and 1.0.7 is in debian stable |
15:02.58 | RoyK[at] | then get another card |
15:03.26 | knobo | which one do you recomend? |
15:03.53 | RoyK[at] | http://www.komplett.no/k/ki.asp?sku=119006 |
15:03.55 | RoyK[at] | that one |
15:04.06 | RoyK[at] | works well with bristuff and visdn |
15:04.26 | RoyK[at] | i've heard some people have even managed to make it work with capi, but I don't have any idea how |
15:07.11 | RoyK[at] | you can also use more of them in a system, but they generate a bloody storm of interrupts, so it might not be so good |
15:07.28 | *** join/#asterisk nf1 (n=nf1@vpn-pppoe-213-240-242-81.megalan.bg) |
15:07.30 | RoyK[at] | two or three should work, though |
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15:09.38 | smackus2 | has anyone had experience with an e&m winkstart T1 em_w? I have set one up, but it does not show caller ID. Is there some other setting that has to be used to send it? I can get it to work with my PRI's |
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15:25.19 | trelane_ | are digium's offices closed? the pbx is being evil and noone's available at the operator extensions |
15:25.44 | Hmmhesays | being evil? |
15:25.49 | Hmmhesays | feed it some children |
15:25.59 | trelane_ | Hmmhesays, I would but it's in atlanta iirc |
15:26.06 | trelane_ | that would require driving |
15:26.09 | Hmmhesays | what type of pbx? |
15:26.20 | trelane_ | I'd hope they're using asterisk |
15:26.26 | Skarmeth | hi all |
15:26.32 | Hmmhesays | Hello |
15:27.35 | malcolmd | we're effectively closed; there are only 3 people here today. |
15:27.51 | Skarmeth | I am trying to set up a E1 ISDN PRI with Asterisk and TE110P, I can call mobile phones (local and long distance), I can call tollfree numbers (0800), I can call long distance |
15:27.57 | Sonderblade | how long does it usually take for softphones to reregister to a server when you reboot that server? |
15:28.00 | Skarmeth | but not local 8 numbers |
15:28.10 | Hmmhesays | Sonderblade: whatever the timeout is? |
15:28.21 | Hmmhesays | looks like a DP problem Skarmeth |
15:28.31 | Sonderblade | Hmmhesays: that is why i said *usually*.... |
15:28.43 | Hmmhesays | there is no "Usually" |
15:28.54 | Hmmhesays | each endpoint is going to likely have a different timeout |
15:29.28 | Sonderblade | Skarmeth: you need to add the area code to local numbers |
15:29.31 | Hmmhesays | if there is a specific softphone you are looking for, just download it and look at the default |
15:29.33 | Qb3rt | Sonderblade: let say after 30minutes if it is not registering you have a problem |
15:29.51 | Sonderblade | Qb3rt: thanks |
15:30.12 | Hmmhesays | heh |
15:30.30 | Hmmhesays | I see what I get for giving decent advice, instead of a generic answer |
15:31.54 | *** part/#asterisk mosty (i=mostynm@60-241-198-194.static.tpgi.com.au) |
15:32.12 | rob0 | I have children and am in driving range from Digium! |
15:32.41 | Hmmhesays | 30 minutes, I don't think I've ever seen an endpoint have 1800 seconds for a registration time out |
15:33.03 | Hmmhesays | you see a lot of 60, 180 and 3600's |
15:33.54 | NotJohnDavid | salviadud: i got the SPA3k working |
15:34.15 | *** join/#asterisk SpaceBass (n=sp@static-71-251-230-6.rcmdva.fios.verizon.net) |
15:34.28 | Qb3rt | Hmmhesays: yeah! true |
15:34.31 | SpaceBass | anyone using linksys phones? Anyone get auto answer working |
15:34.50 | Hmmhesays | model number FOO! |
15:35.03 | Hmmhesays | haha |
15:35.11 | SplasPood | Does anyone know if it's possible to use RealTime to store meetme.conf ? |
15:35.13 | SpaceBass | SPA-941 and 942 |
15:35.45 | Qb3rt | bye A+ |
15:36.36 | salviadud | NotJohnDavid, how you do it? |
15:38.29 | Hmmhesays | SpaceBass: you still just messing around with voip at home? |
15:38.54 | NotJohnDavid | salviadud: posted to a forum and someone explained it to me. I knew that the SPA3k handled PSTN like Voip. the way it does pass thru is encode the PSTN and then place a call to the FXS |
15:39.11 | Skarmeth | Sonderblade, not here in brazil |
15:39.15 | SpaceBass | pretty much |
15:39.28 | salviadud | NotJohnDavid, so what did you tweak at the end? |
15:39.31 | NotJohnDavid | salviadud: what I didn't know that the only codec that it can use is g711. If you set Line1 to something else it just doesn't work. doesn't warn you about it... no documentation anywhere |
15:39.44 | SpaceBass | got some voip related projects for work up my sleeve...but I have to sneak that stuff in |
15:40.02 | *** join/#asterisk _problem_ (n=lokesh_k@estrela.nortenet.pt) |
15:40.10 | NotJohnDavid | slaviadud: so for passthru to work both PSTN and Line1 have to be set to g711. AND when I emailed tech support (Sipura/linksys) they didn't mention this in the 4-5 emails that were exchanged |
15:41.00 | Skarmeth | Sonderblade, may be something related to pri dial plan |
15:41.45 | NotJohnDavid | salviadud: thanks for the help. I just think it comes down to poor documentation on their part |
15:42.00 | Hmmhesays | SpaceBass why is that? |
15:42.24 | Hmmhesays | Send them my way, I'm in between projects at the moment |
15:42.42 | SpaceBass | im not in IT at all....its hard to relate VoIP to healthcare :) |
15:43.17 | Hmmhesays | what do you do in healthcare? |
15:44.03 | SpaceBass | we have a TON of conference calls...I'm on like 5 a day...so I'd like to have a * box recieve the e-mail invite for the conf call, schedule a job that dials into the call and records it, then converts to MP3 and uploads to a sharepoint site |
15:44.31 | Hmmhesays | That can be arranged |
15:44.41 | Hmmhesays | imap or pop3? |
15:44.41 | SpaceBass | I'm a revenue cycle consultant...fancy way of saying I help hospitals with patient processing, registrations, billing, etc |
15:44.43 | salviadud | I hate linksys support |
15:44.49 | SpaceBass | salviadud, oxymoron |
15:45.04 | SpaceBass | linksys doesnt support anything...they just put you on hold and transfer you to india |
15:45.14 | SpaceBass | Hmmhesays, imap |
15:45.22 | Hmmhesays | That can be arranged |
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15:45.51 | Hmmhesays | I have an asterisk box that retrieves email and plays mp3's based on subject line |
15:45.55 | Hmmhesays | its not a far leap to what you want |
15:45.55 | SpaceBass | i was working on it...got hung up trying to get postfix to check the mail and parse it |
15:46.03 | SpaceBass | not far at all |
15:46.19 | Hmmhesays | its pop3 but the switch to imap is trivial |
15:46.47 | Hmmhesays | I can get it done for you for a bit of monetary compensation |
15:46.51 | SpaceBass | these are all outlook/exchange invites... we'd (company) have to establish a standard for putting the phone number and access code into the meeting request |
15:47.19 | Hmmhesays | you could have a script parse the email easily enough |
15:47.27 | Hmmhesays | or have a form to autogenerate the email |
15:47.35 | Hmmhesays | a web form type deal |
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15:48.22 | SpaceBass | well, its got to be seamless....my co-workers arnt going to use outlook and a web form...im the only office geek :) |
15:49.01 | a1fa | lool |
15:49.05 | Hmmhesays | then they'd have to put the access number and time into the email |
15:49.14 | Hmmhesays | you could have a script parse just bout any format they could think of |
15:49.28 | Hmmhesays | that would be no big deal |
15:49.46 | SpaceBass | typically we put the number and access code into the location field in the metting request |
15:50.01 | SpaceBass | when you get one via imap or pop (IE not outlook) all of that is there in cleartext |
15:50.27 | SpaceBass | and if you can fix my site-to-site VPN while you are at it..... :) |
15:50.44 | Hmmhesays | what vpn? |
15:50.59 | Hmmhesays | SpaceBass that would be cake to pull info out of that email |
15:51.06 | SpaceBass | using freeswan via IPcop |
15:51.18 | Hmmhesays | what are your two vpn endpoints? |
15:51.58 | SpaceBass | both are IPcop routers...the problem is one endpoint gets a private IP for its WAN address (but has a Public IP that points to it) |
15:52.25 | Hmmhesays | I would use a couple wrt routers with openvpn |
15:52.27 | SpaceBass | and Im going to build a freenas box today...if I can get out of this hangover :) |
15:52.38 | SpaceBass | I think OpenVPN probably works a lot better |
15:52.42 | Hmmhesays | probably? |
15:52.52 | SpaceBass | well, I havent played with it to confirm :) |
15:52.55 | *** part/#asterisk NotJohnDavid (i=dave@c-68-47-199-178.hsd1.tn.comcast.net) |
15:53.11 | Hmmhesays | I run all kinds of network traffic over openvpn |
15:53.16 | Hmmhesays | it...just...rocks |
15:53.34 | SpaceBass | I need to play with it |
15:54.08 | Hmmhesays | well the fact that you can run it on an $80 router and have it be stable.. |
15:54.09 | SpaceBass | to access my home network I use windows 2003 as the VPN server...works fine...but for this net-to-net that I'm trying to set up OpenVPN might be the way to go |
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15:54.31 | Hmmhesays | yes, it would be |
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15:55.53 | Hmmhesays | openvpn has some kickass features too |
15:56.01 | [TK]D-Fender | Hmmhesays : OpenWRT? |
15:56.05 | SpaceBass | but I'm pretty partial to IPcop....although I think Endian Firewall (which is an IPcop clone) uses OpenVPN |
15:56.12 | Hmmhesays | i've used it on sat connections before that have a really small MTU |
15:56.29 | Hmmhesays | you can fragment the packets and send them across |
15:56.36 | Hmmhesays | [TK]D-Fender: yeah |
15:57.14 | SpaceBass | trying to figure out the most cost effective way to do this FreeNAS box.... a few 300gb drives mirrored, or several 150gb drives spanned and mirrored |
15:57.35 | Hmmhesays | what are you trying to accomplish? |
15:57.58 | SpaceBass | massive and redundant storage :) |
15:58.14 | jbalcomb | SpaceBass Raid 10 (one-zero) |
15:58.32 | [TK]D-Fender | Hmmhesays : I'd think that a small MTU would absolutely kill Sat because of latency and packet overhead.... |
15:59.11 | SpaceBass | VPN over a sat connection sucks! |
15:59.47 | Hmmhesays | [TK]D-Fender: its a guess and check type deal |
15:59.54 | Hmmhesays | and SpaceBass: not really |
16:00.54 | SpaceBass | maybe its just that satellite connections suck in general :) |
16:01.04 | Hmmhesays | mostly yeah |
16:01.30 | Hmmhesays | but when you're in a grass hut and you have phone service where you used to use that dude down the road that could run fast... |
16:01.51 | SpaceBass | true |
16:02.03 | SpaceBass | we have a family farm and sat is all we can get there...and it does beat dialup |
16:02.15 | SpaceBass | but its very expensive and still not that fast... |
16:02.29 | Hmmhesays | and you don't need use binoculars and hope the hut down the roads window is open to see boobies |
16:02.34 | Hmmhesays | satellite is pretty damn nice |
16:02.41 | SpaceBass | lol |
16:04.43 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
16:05.03 | salviadud | i have pansat |
16:05.10 | salviadud | i wish they had a fix... |
16:05.11 | Hmmhesays | So anyhoo SpaceBass, if you want some help on that project let me know. I'm going to be available for awhile |
16:06.36 | SpaceBass | I will! |
16:07.02 | SpaceBass | I'm not working today...had to fedex my laptop to Ga to be reimaged...but when I get it back thats on my list of stuff to paly with |
16:07.32 | Hmmhesays | you use googletalk at all? |
16:07.47 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
16:07.56 | SpaceBass | i had it loaded on my XP box...but haven't played with it in a while |
16:09.11 | *** join/#asterisk neoXite (n=neoxite@xdsl-87-78-77-206.netcologne.de) |
16:09.41 | Hmmhesays | ahh ok, I have mine going most of the time |
16:09.46 | Hmmhesays | pretty kickass messenger |
16:09.51 | neoXite | hi, i have problems building asterisk 1.2.9.1 on dapper drake |
16:10.05 | *** join/#asterisk Bert- (n=bert@LAubervilliers-151-12-81-84.w193-252.abo.wanadoo.fr) |
16:10.14 | Bert- | hello again :) |
16:10.24 | *** join/#asterisk smacku1 (n=smackus@c-67-169-248-217.hsd1.ut.comcast.net) |
16:10.33 | neoXite | i installed all the necessary packages (build-dep from version 1.2.7.1) but gcc gives me an error during building |
16:10.37 | Bert- | can I run Idefisk on a windows CE plateform please ? |
16:10.48 | Bert- | or only on windows XP/2K ? |
16:10.51 | Hmmhesays | is there a build for windows CE? |
16:10.54 | neoXite | anyone here experienced similar problems? |
16:11.21 | Hmmhesays | I've had gcc errors before, however if you don't tell us the error then your question will never be answered |
16:11.38 | Bert- | what is the gcc error ? |
16:11.52 | neoXite | just a sec, i had it copied somewhere |
16:11.59 | smacku1 | I am using inbound digits from an 800 number to route calls, and I am capturing CDR Data into mysql database. But I am not finding any data to identify those calls by incoming digits. does cdr trap incoming digits? if so, by what name? |
16:12.00 | Hmmhesays | there is an echo in here, lol |
16:12.29 | SpaceBass | Hmmhesays, it is great...unfortunatly I dont know anyone else using it :) .... aslo on a mac most of the time |
16:12.39 | Hmmhesays | ahh ok |
16:12.43 | SpaceBass | I know iChat can connect to google talk...just been lazy about it |
16:12.58 | Hmmhesays | it uses jabber protocol |
16:13.33 | Bert- | hey it's 6 p.m !! I'm still at work |
16:13.36 | Bert- | i'm crazy ... :) |
16:13.46 | Bert- | see ya here ;) |
16:14.01 | smacku1 | does the cdr not store dnis? |
16:14.59 | eKo1 | smacku1: not by default |
16:15.12 | smacku1 | how can I change that? |
16:16.19 | neoXite | okay, should i paste in here or is that against channel rules? |
16:16.25 | eKo1 | Hack the cdr_* module that you're using. |
16:16.35 | Hmmhesays | heh you *could* do that |
16:16.40 | Hmmhesays | or use custom cdrs |
16:17.02 | smacku1 | ok, I am even newer to databases than I am asterisk... is there a newbie answer? |
16:17.07 | *** join/#asterisk Ludo_ (n=Ludo@obelix.zoxx.net) |
16:17.16 | eKo1 | I've never used custom cdrs... |
16:17.31 | Hmmhesays | pay me money and me give you answer |
16:17.32 | Hmmhesays | lol |
16:17.42 | neoXite | gcc says 'chan_zap.c:9160: error: ‘pri_event_setup_ack’ has no member named ‘call’' |
16:18.16 | neoXite | i tried building the same source on debian/testing and it compiled fine |
16:18.41 | smacku1 | I am open to that... but with a different approach. I have offered others payment for work. I need a quote. The end result is I need to trap dnis in my cdr, how much for you to do it? |
16:18.54 | neoXite | but on ubuntu/dapper it fails with that error |
16:19.05 | Hmmhesays | where is the call coming from? |
16:19.21 | smacku1 | i have 10 different 800 numbers coming in from all over. |
16:19.28 | smacku1 | sending the last 4 digits as dnis |
16:19.42 | smacku1 | i need to bill companies based on the 800 number dialed |
16:19.50 | Hmmhesays | i'm guessing if you're a n00b to this you're probably using asterisk@home distro? |
16:19.56 | smacku1 | nope |
16:20.07 | smacku1 | this is a 80+ agent call center |
16:21.10 | Hmmhesays | what shows up in the SRC field of your CDR? |
16:21.34 | smacku1 | hang on... gotta connect to vpn to get it |
16:22.00 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
16:22.52 | Ludo_ | Hi, I have one asterisk PABX with one public ip, if I connect directly to this server from outside or inside of the network it's ok |
16:23.11 | Ludo_ | now I would like try to connect to this server in using a client behind a nat |
16:23.22 | Ludo_ | do I need a stun server or not? |
16:23.24 | Ludo_ | to do it? |
16:23.37 | Hmmhesays | not |
16:24.47 | Ludo_ | ok just to activate nat=yes in my sip.conf? |
16:25.56 | Hmmhesays | worth a shot |
16:26.07 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
16:26.24 | Ludo_ | what do you mean? |
16:27.57 | *** part/#asterisk smacku1 (n=smackus@c-67-169-248-217.hsd1.ut.comcast.net) |
16:28.03 | *** join/#asterisk smacku1 (n=smackus@c-67-169-248-217.hsd1.ut.comcast.net) |
16:28.15 | *** join/#asterisk muppetmaster (n=jasongoe@169.red-81-184-73.user.auna.net) |
16:28.20 | muppetmaster | Hello |
16:28.26 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
16:28.37 | muppetmaster | Has something changed with the codec disallow/allow in terms of ulaw/alaw as of v1.2.9.1? |
16:29.10 | muppetmaster | I have disallow=all and allow=ulaw, and yet when I call with Xlite with G711u enabled on its side I get a notification that no codecs are available. |
16:29.23 | RoyK[at] | try alaw |
16:30.01 | muppetmaster | Okay |
16:30.16 | muppetmaster | I did, but will do again to be sure |
16:30.19 | *** part/#asterisk rene- (n=rene-@dsl-200-67-175-250.prod-empresarial.com.mx) |
16:30.20 | muppetmaster | As it had the same problem |
16:31.21 | Qwell[] | Corydon-w: ;) |
16:31.39 | *** join/#asterisk Gamercjm (n=chris@pool-71-254-175-66.lsanca.fios.verizon.net) |
16:31.44 | Corydon-w | :-P |
16:33.12 | Corydon-w | Qwell[]: 7 days |
16:33.28 | Qwell[] | 6 |
16:33.36 | Qwell[] | until I get there, anyhow |
16:33.39 | RoyK[at] | knobo: ping |
16:33.41 | RoyK[at] | ~ping |
16:33.43 | jbot | pong |
16:33.43 | Corydon-w | Qwell[]: you're coming to Nashville on Sunday? |
16:33.47 | RoyK[at] | muppetmaster: i'm running x-lite with alaw |
16:33.50 | RoyK[at] | works splendidly |
16:33.50 | Qwell[] | Corydon-w: nope, HSV |
16:33.53 | RoyK[at] | and we run 1.2.9.1 with several thousand customers |
16:33.55 | RoyK[at] | so if alaw didn't work, our customer support centre would kill me |
16:33.57 | RoyK[at] | :) |
16:34.05 | Qwell[] | Corydon-w: You'll have to drive up, to harrass me :P |
16:34.10 | Qwell[] | or..down, as it were |
16:34.12 | *** join/#asterisk BugKham (i=CKGLOB@125.24.0.136) |
16:34.15 | Corydon-w | Qwell[]: I meant until you'd have me to spoon |
16:34.18 | Qwell[] | :P |
16:34.19 | *** part/#asterisk BugKham (i=CKGLOB@125.24.0.136) |
16:34.19 | jbalcomb | muppetmaster: i'm running x-lite with ulaw |
16:34.27 | muppetmaster | Not sure why it is not working... |
16:34.29 | jbalcomb | works splendidly |
16:35.08 | jbalcomb | anyone using gnophone? |
16:35.18 | muppetmaster | ulaw nor alaw is working |
16:35.29 | muppetmaster | If I do an allow all, then it works |
16:35.41 | jbalcomb | muppetmaster what codec does it use when it works? |
16:35.46 | muppetmaster | ulaw |
16:36.04 | jbalcomb | whats the trouble then really? |
16:36.16 | muppetmaster | The fact that I want to select that it only uses that codec |
16:36.19 | jbalcomb | granted its queer but disallowing codecs isnt the biggest issue ever |
16:36.27 | muppetmaster | When I try that, then it tells me no go, as no supported codec is available |
16:36.49 | Qwell[] | in which order are you allowing/disallowing codecs? |
16:36.52 | jbalcomb | its like worrying about whether the TP unrolls from the front or the back |
16:36.58 | muppetmaster | disallow=all |
16:37.00 | muppetmaster | allow=ulaw |
16:37.03 | Qwell[] | jbalcomb: sorry, but that DOES matter |
16:37.13 | jbalcomb | =) |
16:37.17 | *** join/#asterisk rene- (n=rene-@dsl-200-67-175-250.prod-empresarial.com.mx) |
16:37.28 | muppetmaster | Then get: Jul 3 18:15:56 NOTICE[13210]: chan_sip.c:3691 process_sdp: No compatible codecs! |
16:37.29 | jbalcomb | Qwell[]: I prefer over-the-top |
16:37.34 | Qwell[] | as you should |
16:37.55 | jbalcomb | Qwell[]: It only makes sense. ;) |
16:38.18 | neoXite | ok nevermind, fixed it. my zaptel-source package apparently was too old |
16:38.30 | Qwell[] | I could go into detail on why, but it's incredibly offtopic, and disturbing :P |
16:38.59 | RoyK[at] | muppetmaster: as i said, i use disallow=all,allow=alaw and it works with all clients i've tried |
16:39.11 | muppetmaster | RoyK[at] I have no idea what is going on here. |
16:39.14 | muppetmaster | Very strange |
16:39.23 | muppetmaster | I have used it before no problem, but puking as you can see |
16:39.26 | RoyK[at] | what does sip debug say? |
16:39.34 | *** join/#asterisk Eggplant (i=No@dsl-216-155-213-228.cascadeaccess.com) |
16:39.47 | muppetmaster | RoyK[at] Too much traffic at the moment to run a debug, need to isolate it |
16:39.52 | RoyK[at] | pastebin your config, then |
16:39.52 | RoyK[at] | and debug output |
16:39.52 | RoyK[at] | s/debug/verbose/ |
16:39.56 | jbalcomb | muppetmaster: run the command in asterisk and see what codecs it think are available. Anyone know what that command is? |
16:40.06 | muppetmaster | show codecs |
16:40.06 | RoyK[at] | sip debug peer xxxxx |
16:40.14 | RoyK[at] | muppetmaster: won't help much |
16:40.37 | RoyK[at] | muppetmaster: show translation will show what codecs are actually loaded |
16:40.46 | jbalcomb | muppetmaster: too much traffic for debug? HA! I run debug with 46 zap channels and 150 sip users at midday on wednesday1 |
16:40.54 | RoyK[at] | shite. this internet connection is bad..... |
16:40.56 | RoyK[at] | 100 packets transmitted, 76 packets received, 24% packet loss |
16:40.56 | RoyK[at] | round-trip min/avg/max/stddev = 59.958/121.233/880.028/107.601 ms |
16:41.08 | muppetmaster | shows ulaw and alaw both present and thea bility to translate |
16:41.21 | RoyK[at] | then it's a config problem. pastebin your config and verbose output |
16:41.23 | RoyK[at] | ~pb |
16:41.24 | jbot | it has been said that pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/ |
16:41.38 | RoyK[at] | if you can't isolate it, setup another box and try there |
16:41.46 | jbalcomb | the jbot entry for pb should be shorter |
16:42.08 | RoyK[at] | ~jbalcomb |
16:42.27 | jbalcomb | ~TP RoyK[at] |
16:42.50 | RoyK[at] | jbot: tp? |
16:42.51 | jbot | [tp] I AM THE GREAT CORNHOLIO! I NEED TP FOR MY BUNGHOLE! |
16:42.52 | jbalcomb | haha.. dude, i just threw toilet paper all over you, remotely. |
16:43.10 | jbalcomb | Feel the pang. |
16:43.16 | jbalcomb | (tm) |
16:43.31 | jbalcomb | jbot: yermom? |
16:43.45 | *** part/#asterisk smacku1 (n=smackus@c-67-169-248-217.hsd1.ut.comcast.net) |
16:43.49 | neoXite | jbot: orly? |
16:43.50 | jbot | YARLY |
16:44.00 | jbalcomb | jbot: wtf? |
16:44.01 | jbot | who? |
16:44.23 | jbalcomb | jbot: Zombie Jesus? |
16:44.24 | jbot | ACTION summons an army of the undead to eat Jesus |
16:44.32 | jbalcomb | oh damn.... |
16:44.45 | RoyK[at] | :) |
16:44.58 | RoyK[at] | not too much left, though |
16:45.21 | skraelings001 | what does this mean ? Jul 3 10:51:53 NOTICE[13638] channel.c: Dropping incompatible voice frame on Local/202@INTERNACIONAL-5175,2 of format alaw since our native format has changed to slin |
16:46.24 | RoyK[at] | skraelings001: 1.2? |
16:46.43 | skraelings001 | yes |
16:46.54 | RoyK[at] | what codec? |
16:48.38 | skraelings001 | alaw | ulaw | gsm |
16:48.50 | RoyK[at] | skretry disallow=all, then allow=alaw, and try again |
16:49.09 | RoyK[at] | ~codecs |
16:49.10 | jbot | extra, extra, read all about it, codecs is http://snipurl.com/wiki_codecs |
16:49.10 | RoyK[at] | ~codec |
16:49.12 | jbot | hmm... codecs is http://snipurl.com/wiki_codecs |
16:51.38 | RoyK[at] | jbot: codecs is also If you have audio/codec problems, first try to 'disallow=all' and 'allow=all' and see if that works |
16:51.40 | jbot | okay, RoyK[at] |
16:53.24 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
16:54.23 | RoyK[at] | ~codecs |
16:54.25 | jbot | [codecs] http://snipurl.com/wiki_codecs. If you have audio/codec problems, first try to 'disallow=all' and 'allow=all' and see if that works |
16:54.30 | RoyK[at] | ~codec |
16:54.31 | jbot | i heard codecs is http://snipurl.com/wiki_codecs. If you have audio/codec problems, first try to 'disallow=all' and 'allow=all' and see if that works |
16:56.05 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
16:56.48 | *** join/#asterisk Bullseye_Network (n=info@216.143.192.69) |
16:58.34 | *** join/#asterisk HuSoft (n=apo@227stb47.codetel.net.do) |
16:58.42 | *** join/#asterisk ToyMan (n=stuq@74-32-9-135.dsl1.mdl.ny.frontiernet.net) |
16:59.49 | skraelings001 | RoyK[at] : i found that is related with MoH, seems can't get back to previous format |
17:03.22 | HuSoft | Is this the way to support both num/alpha dialing, e.x.: sip:husoft@localhost ?: exten => husoft, 1, dial(SIP/1234) |
17:04.55 | *** join/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net) |
17:06.48 | *** part/#asterisk HuSoft (n=apo@227stb47.codetel.net.do) |
17:08.16 | Ludo_ | I have nat=yes activated, I try to call my asterisk server from a sip client behind nat, I see in log some activity, but I don't heard nothing on my sip client |
17:08.28 | Ludo_ | my sip client is behind a fon router |
17:08.57 | RoyK[at] | skraelings001: what sort of MoH format? |
17:09.00 | skraelings001 | RoyK[at] : having diferents formats for same file seems to be the answer and also low-down cpu usage |
17:09.16 | skraelings001 | mp3 |
17:09.24 | Ludo_ | do you think it's the fon router the problem? |
17:09.33 | Ludo_ | I get a 408 timeout |
17:10.29 | SpaceBass | anyone useing the linksys phones (SPA-941 and 942) gotten auto answer to work? |
17:11.06 | jbalcomb | SpaceBass I have three but i dunno if the auto answer works, sorry. |
17:13.10 | *** join/#asterisk Tsop (n=Tsop@69.155.81.24) |
17:13.41 | jbalcomb | SpaceBass: ours do not auto-answer. |
17:13.45 | Tsop | good morning all |
17:15.26 | SpaceBass | hummm |
17:17.41 | Tsop | i am new to linux world, i have just finish installing debian and i already love it ;), i would like to install asterisk and use it at home just for my personal use? do i need to subscribe to any service or this is compeletly free? |
17:18.14 | Qwell[] | Tsop: It's free as in beer (and libre), for any usage |
17:18.22 | Qwell[] | feel free to use it in a corp, or whatever |
17:18.28 | Qwell[] | just...I suggest you read the GPL |
17:19.37 | Tsop | Qwell, so with just my dsl connection and my linux box i will be able to make phone calls? |
17:19.45 | Qwell[] | not quite |
17:19.59 | Tsop | what do i need to get started |
17:20.08 | Qwell[] | ~docs |
17:20.09 | jbot | from memory, docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
17:20.11 | Qwell[] | ~wikis |
17:20.13 | jbot | rumour has it, wikis is http://www.voip-info.org |
17:20.13 | *** join/#asterisk Sponge_bob (n=None@cpe-66-27-162-13.socal.res.rr.com) |
17:21.07 | Tsop | i had a linksys RT31P2 but i couldn't get it to work , they Vonage told me it's lock to them and i called linksys and they also told me i'll have to call vonage |
17:21.08 | *** join/#asterisk syle (n=blah@unaffiliated/syle) |
17:21.14 | Sponge_bob | what's the best way to test out a meetme room for echo? |
17:21.25 | Qwell[] | Sponge_bob: call it, and listen for echo |
17:21.35 | Sponge_bob | i need 23 calls |
17:21.41 | Qwell[] | call it 23 times? |
17:21.52 | Qwell[] | or use a call file for 22 of them |
17:22.32 | *** join/#asterisk file2 (n=file@out.clearnet.com) |
17:22.39 | Qwell[] | eeps |
17:22.41 | Sponge_bob | well i found that with 2-5 users the calls seems to be ok. if i get a couple more some hear their own voice |
17:22.45 | file2 | moon |
17:22.50 | file2 | er mooo |
17:23.22 | skraelings001 | Can anyone shed light on what these warnings mean >channel.c: Avoided initial deadlock for '0x829c350', 10 retries!<, how they can be avoided, and if they are something to worry about? |
17:23.57 | Sponge_bob | Qwell[]: how can i test a meetme with a call file? |
17:24.19 | file2 | magic |
17:24.40 | *** join/#asterisk argos73 (n=mike@w010.z208036240.chi-il.dsl.cnc.net) |
17:24.40 | Sponge_bob | tell me how the magic works |
17:25.01 | Tsop | Qwell, thanks, i know i'll def. need to read , but as i told u i'm a n00be, can u just point me to a link that you'll give to someone that know nothing about asterisk and want to set it up? |
17:26.34 | file2 | ooh nice outside |
17:26.48 | Qwell[] | file2: sorry, you aren't allowed outside |
17:27.15 | Qwell[] | close your window too...no peeking |
17:27.18 | file2 | why not :( |
17:27.25 | Qwell[] | because I'm mean |
17:27.33 | file2 | ah ic |
17:27.41 | RoyK[at] | ~rtfs |
17:27.42 | jbot | i guess rtfs is probably read the f*cking source... |
17:27.42 | file2 | makes sense |
17:28.25 | *** join/#asterisk Ellegon (n=sbryant@72.164.50.72) |
17:28.54 | *** join/#asterisk evisu (n=hIRC@bzq-88-152-238-38.red.bezeqint.net) |
17:29.48 | Skarmeth | Sonderblade, I am back. I was trying to call a number of my E1 and it was sucessful (I can see that the call goes out), but the outgoing and the inconming channels was hang up. I have a TE110P on IRQ 24 and a TDM04B on IRQ 25... it make any sense? |
17:29.55 | trelane_ | is there a document somewhere listing options to pass to insmod on zap modules |
17:30.00 | Ellegon | quick question... I have the Zap dev kit. When ever I try to make a call from Zap/1 (FXS) to Zap/4 (FXO) I can hear the person on the call fine but they complain that I am muffaled |
17:30.15 | Ellegon | this problem is not there if I call from one of my pap2's |
17:30.21 | Ellegon | to Zap/4 |
17:31.15 | Ellegon | Also the call is fine from Zap/1 to any port on the Pap'2 |
17:33.40 | *** part/#asterisk SpaceBass (n=sp@static-71-251-230-6.rcmdva.fios.verizon.net) |
17:38.15 | *** join/#asterisk donpaolo (n=donpaolo@pri-214-b7.codetel.net.do) |
17:39.06 | Tsop | i don't know where to get this information please? |
17:39.46 | donpaolo | Hi guys! I get this error when calling through mutualphone: WARNING[5015]: channel.c:2691 ast_channel_make_compatible: No path to translate from SIP/mutualphone-5e3b(256) to SIP/oficinamision-a47a(4) |
17:39.48 | Tsop | i need to know what do i need in addition of installing asterisk to make phone calls , do i have to sign with a voip provider |
17:40.03 | donpaolo | and then: WARNING[5015]: channel.c:2691 ast_channel_make_compatible: No path to translate from SIP/oficinamision-a47a(4) to SIP/mutualphone-5e3b(256) |
17:40.03 | donpaolo | Jul 3 13:38:25 WARNING[5015]: app_dial.c:1572 dial_exec_full: Had to drop call because I couldn't make SIP/oficinamision-a47a compatible with SIP/mutualphone-5e3b |
17:40.22 | file2 | can not transcode between g729 and ulaw |
17:40.48 | donpaolo | What have I to change? I followed the instructions on mutualphone site |
17:40.56 | RoyK[at] | perhaps you haven't bought any g.729 codecs..... |
17:41.12 | file2 | you have to buy a license if you want to transcode |
17:41.28 | RoyK[at] | $10 per concurrent transcode |
17:41.31 | RoyK[at] | iirc |
17:42.07 | *** join/#asterisk smacku1 (n=smackus@c-67-169-248-217.hsd1.ut.comcast.net) |
17:42.17 | donpaolo | file, are you speaking to me? |
17:42.26 | file2 | yes |
17:43.03 | smacku1 | is there any way for a queue once a call has timed out and been delivered to another extension to be able to detect that the other extension did not answer and then transfer the call to a different extension? |
17:43.29 | RoyK[at] | donpaolo: yes, it's $10 per concurrent transcode |
17:44.05 | file2 | so nice outside |
17:44.13 | donpaolo | RoyK[at], do you mean that I can't communicate with them without buying a g.729 license? |
17:45.04 | *** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka) |
17:46.37 | *** part/#asterisk tgrman (n=jcmoore@picard.ojc.nuvio.com) |
17:46.45 | RoyK[at] | donpaolo: yes |
17:46.52 | RoyK[at] | donpaolo: except in passthrough |
17:46.57 | donpaolo | RoyK[at], how? |
17:47.02 | RoyK[at] | how what? |
17:47.05 | RoyK[at] | how to buy? |
17:47.11 | RoyK[at] | or how to use passthrough? |
17:47.50 | *** join/#asterisk jcmoore (n=jcmoore@picard.ojc.nuvio.com) |
17:47.54 | donpaolo | how to use pasthrough |
17:48.19 | RoyK[at] | just setup client a to use g.729, client b to use g.729, disallow=all, allow=g729 |
17:48.44 | RoyK[at] | asterisk doesn't understand much of the audio, and thus cannot playback() or playtones() or anything, but the users can communicate |
17:49.23 | RoyK[at] | trelane: man modinfo :) |
17:49.54 | donpaolo | RoyK[at], I set up * disallow=all allow=g729, but mutualphone gives me those error I put above... |
17:50.53 | RoyK[at] | excactly what are you trying to do? |
17:51.14 | *** join/#asterisk Johnnie (n=john@pdpc/supporter/active/Johnnie) |
17:52.48 | Ellegon | quick question... I have the Zap dev kit. When ever I try to make a call from Zap/1 (FXS) to Zap/4 (FXO) I can hear the person on the call fine but they complain that I am muffled. this problem is not there if I call from one of my pap2's |
17:52.53 | smacku1 | if I wanted to replace time: GotoIfTime(9:00-17:00| with *, would it be GotoIfTime(*| or GotoIfTime(*-*|? |
17:53.00 | Ellegon | Also the call is fine from Zap/1 to any port on the Pap'2 |
17:54.03 | donpaolo | RoyK[at], I want to connect to mutualphone, and I set up sip.conf according to their instructions in http://www.mutualphone.com/asterisk.htm , I can register with them but I can't place a call, I get errors: "ast_channel_make_compatible: No path to translate from SIP/mutualphone-01fa(256) to SIP/oficinamision-fbf6(4)" - "chan_sip.c:2542 sip_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 256/256)" - "chann |
17:54.03 | donpaolo | el.c:2691 ast_channel_make_compatible: No path to translate from SIP/oficinamision-fbf6(4) to SIP/mutualphone-01fa(256)" - "app_dial.c:1572 dial_exec_full: Had to drop call because I couldn't make SIP/oficinamision-fbf6 compatible with SIP/mutualphone-01fa" |
17:54.37 | donpaolo | smacku1, the 1st |
17:54.47 | smacku1 | thank you much |
17:55.59 | RoyK[at] | donpaolo: then asterisk is trying to transcode, and not to do passthrough |
17:56.04 | RoyK[at] | dunno why |
17:57.29 | *** join/#asterisk Johnnie (n=john@pdpc/supporter/active/Johnnie) |
17:58.24 | *** join/#asterisk TripleFFFF (n=TripleFF@147-102.mc.cite.net) |
17:58.24 | TripleFFFF | : hey guys.. got something weird.. got a pap2 client.. his calls drop every 300 seconds.. no firewall , hes dmz'ed.. im wondering if asterisk could be thinking that call nevr asnwered and still ringing and drops after 300 ? |
17:59.17 | *** join/#asterisk Johnnie (n=john@pdpc/supporter/active/Johnnie) |
17:59.20 | TripleFFFF | anyone ? |
18:00.47 | cypromis | none |
18:01.02 | dlynes_home | TripleFFFF: 300 seconds every single time? |
18:01.05 | TripleFFFF | nonewhat ? |
18:01.06 | TripleFFFF | yeah |
18:01.09 | TripleFFFF | well 300 after pickup |
18:01.15 | TripleFFFF | so 301 to 325 |
18:01.20 | TripleFFFF | depending on how long it rang |
18:01.25 | TripleFFFF | but its that im sure |
18:01.31 | TripleFFFF | cant be always 300 sec |
18:01.40 | TripleFFFF | and got people doing 3000-4000 secs no prob |
18:01.46 | TripleFFFF | its the linksyspap2t na im sure |
18:01.53 | dlynes_home | TripleFFFF: is he on dialup, or broadband? |
18:01.58 | TripleFFFF | broadband |
18:02.04 | *** join/#asterisk |marv0997| (i=marv0997@190.4.2.83) |
18:02.16 | dlynes_home | TripleFFFF: ppoe, or regular? |
18:02.28 | TripleFFFF | cable |
18:02.28 | *** join/#asterisk dec_ (n=tom@ppp206-151.lns1.adl2.internode.on.net) |
18:02.38 | dlynes_home | is it ppoe, though? |
18:02.41 | dlynes_home | erm |
18:02.47 | dlynes_home | pppoe i mean? |
18:02.50 | TripleFFFF | no |
18:03.09 | RoyK[at] | pppoipoatmoatoe |
18:03.28 | dlynes_home | point-to-point-protocol-over-ethernet |
18:03.54 | TripleFFFF | vonage is ok |
18:03.58 | TripleFFFF | anyother is ok |
18:03.59 | TripleFFFF | just me |
18:04.00 | TripleFFFF | lol |
18:04.05 | dlynes_home | ah, ok |
18:04.06 | TripleFFFF | but using the linksys pap2t na, |
18:04.34 | *** part/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
18:04.39 | dlynes_home | is he the only one you're connecting to in the same manner? |
18:04.52 | donpaolo | RoyK[at], increible, I can connect to mutualphone with gsm codec! |
18:04.53 | TripleFFFF | no |
18:05.21 | dlynes_home | TripleFFFF: is he the only one that you have that's not portmapped? |
18:05.36 | TripleFFFF | what you mean portmapped ? |
18:05.49 | TripleFFFF | he just tried to dmz since it was making no sense |
18:06.16 | TripleFFFF | its the 300 sec that worrying me |
18:06.21 | dlynes_home | port forwarding udp port 5060 on your router to your machine |
18:06.21 | TripleFFFF | makes me think its on my side |
18:06.28 | TripleFFFF | udp or tcp ? |
18:06.38 | dlynes_home | ^^^^^^^^^^^^^^^^^^^^ |
18:06.41 | dlynes_home | read :) |
18:06.56 | TripleFFFF | i mean i tought was tcp |
18:06.58 | Ellegon | Is there a way to make a Zap fxo port wait 2 rings before picking up? |
18:07.01 | TripleFFFF | but hes DMZ so no matter |
18:07.09 | TripleFFFF | wait(2) |
18:07.09 | dlynes_home | TripleFFFF: i'm asking about yoru side, not his |
18:07.23 | TripleFFFF | ? my box is direct on net |
18:07.30 | TripleFFFF | im runnign asterisk 1.2.9.1 |
18:07.32 | dlynes_home | TripleFFFF: is it firewalled, though? |
18:07.33 | TripleFFFF | on wan |
18:07.35 | TripleFFFF | no |
18:07.38 | TripleFFFF | direct |
18:07.56 | dlynes_home | is it running in a vm on a windows boxen? |
18:08.02 | TripleFFFF | cents 4.2 |
18:08.04 | TripleFFFF | centos |
18:08.14 | TripleFFFF | dual xeons 2 giger |
18:08.22 | TripleFFFF | basically as it should |
18:08.23 | TripleFFFF | ;) |
18:08.35 | dlynes_home | and when you installed it, and it asked about a firewall, you told it none, right? |
18:09.10 | TripleFFFF | no idea |
18:09.15 | TripleFFFF | but iptables is off |
18:09.24 | dlynes_home | type iptables -nL |
18:09.38 | TripleFFFF | bash: iptables: command not found |
18:09.46 | dlynes_home | /usr/sbin/iptables -nL |
18:10.05 | TripleFFFF | <PROTECTED> |
18:10.19 | TripleFFFF | no iptables in there |
18:10.22 | dlynes_home | what do you mean which iptables? there is only one iptables |
18:10.23 | TripleFFFF | located none neither |
18:10.33 | TripleFFFF | lol |
18:10.38 | Tsop | i need to know what do i need in addition of installing asterisk to make phone calls , do i have to sign with a voip provider? |
18:10.42 | TripleFFFF | <PROTECTED> |
18:10.57 | dlynes_home | Tsop: either that, or have an analog line, and get a tdm card |
18:11.10 | dlynes_home | TripleFFFF: i know that...I'm not stupid |
18:11.12 | TripleFFFF | hmm |
18:11.13 | TripleFFFF | <PROTECTED> |
18:11.13 | TripleFFFF | Chain INPUT (policy ACCEPT) |
18:11.13 | TripleFFFF | target prot opt source destination |
18:11.13 | TripleFFFF | <PROTECTED> |
18:11.13 | TripleFFFF | Chain FORWARD (policy ACCEPT) |
18:11.14 | TripleFFFF | target prot opt source destination |
18:11.18 | TripleFFFF | Chain OUTPUT (policy ACCEPT) |
18:11.20 | TripleFFFF | target prot opt source destination |
18:11.22 | salviadud | pastebin |
18:11.22 | TripleFFFF | weird |
18:11.25 | salviadud | for the love of god |
18:11.25 | TripleFFFF | sorry |
18:11.26 | a1fa | god damn you |
18:11.28 | dlynes_home | TripleFFFF: but sbin directories are not usually in your path |
18:11.35 | a1fa | fucking ah!!!!!!!! |
18:11.40 | Tsop | lol |
18:11.45 | a1fa | wtf. paste bin you jack the fuck ass |
18:11.58 | dlynes_home | a1fa: dood...get it straight |
18:12.02 | TripleFFFF | a1fa sirry |
18:12.02 | dlynes_home | a1fa: it's triple fuck ass :p |
18:12.21 | salviadud | yeah, nique ta mere, pastebin |
18:13.24 | Tsop | dlynes_home, thank you, from my understanding if i got a tdm card , and used asterisk and called internationl will this be charged on my analog line? |
18:13.30 | a1fa | some croatian hax0r? |
18:14.15 | dlynes_home | TripleFFFF: is he running a firewall on his end? |
18:14.24 | TripleFFFF | hes dmz';ed |
18:14.28 | TripleFFFF | demilitarized |
18:14.29 | dlynes_home | Tsop: yes |
18:14.40 | dlynes_home | Tsop: doesn't matter...still going through a firewalll |
18:14.50 | dlynes_home | Tsop: erm...mistell |
18:14.59 | dlynes_home | TripleFFFF: doesn't matter...still going through a firewall |
18:15.00 | TripleFFFF | trough |
18:15.08 | dlynes_home | TripleFFFF: is it a real firewall, or a router? |
18:15.10 | TripleFFFF | could be a router table flush of something |
18:15.33 | Tsop | dlynes_home, so yes i will be charged? or it was a yes for TripleFFFF ? |
18:15.40 | dlynes_home | Tsop: that was a yes for you |
18:16.10 | dlynes_home | Tsop: assuming your outgoing calls were going out on analog, and not voip |
18:16.21 | Tsop | dlynes_home, i thought i could use asterisk to make free phone calls |
18:16.34 | dlynes_home | Tsop: not free, necessarily |
18:16.39 | dlynes_home | Tsop: more like inexpensive |
18:16.56 | TripleFFFF | lol |
18:17.01 | Tsop | dlynes_home, my friend is calling me from africa and he told me he doesn't pay anything using this system |
18:17.16 | TripleFFFF | caus he hacked someone lol |
18:17.27 | TripleFFFF | nothing free in life.. even death .. gota pay the toll bro |
18:17.28 | dlynes_home | Tsop: until everyone starts using dundi and enum, and all calls go voip |
18:17.34 | dlynes_home | Tsop: voip will not be 100% free |
18:17.49 | dlynes_home | Tsop: he could be calling for free |
18:18.02 | dlynes_home | Tsop: if he's doing sip direct calling, and not dialing a pstn number |
18:18.41 | Tsop | he is using his pc and calling land # |
18:19.05 | dlynes_home | Tsop: then either he's bullshitting you, or he's stealing long distance from someone |
18:19.24 | TripleFFFF | also dlynes_home.. hes call trough vonage are ok |
18:19.25 | smacku1 | i am trying to make it so that when a queue times out it tries an extension not logged into the queue. If that extension does not answer, I want it to go to another extension's voicemail. Here is what I tried. It did not work. Can anyone tell me if this is possible, and if so how to adjust my dial plan? http://pastebin.ca/78012 |
18:19.28 | TripleFFFF | so the router should be ok |
18:19.29 | dlynes_home | Tsop: or he's found one of those extremely rare free call places |
18:19.29 | *** join/#asterisk file2 (n=IrcNet@out.clearnet.com) |
18:20.02 | dlynes_home | TripleFFFF: he's using a pap2, or a pap2-na? |
18:20.09 | TripleFFFF | pap2 t -na |
18:20.19 | Tsop | dlynes_home, he mentiond something about gizmo project and use it to get sip # |
18:20.41 | _problem_ | smacku1: use n option to exit from queue to dial an extension not in that q |
18:20.43 | dlynes_home | wtf is a pap2 t-na? |
18:21.01 | TripleFFFF | bah |
18:21.02 | smacku1 | ok, let me adjust and have you double check me |
18:21.02 | dlynes_home | oh nvm...the 't' is short for toast? |
18:21.05 | TripleFFFF | upgrades twice ram etc |
18:21.13 | TripleFFFF | Firmware Version: 3.1.10(LSc) |
18:21.14 | dlynes_home | oh |
18:21.22 | TripleFFFF | Product Name: PAP2T |
18:21.44 | Tsop | TripleFFFF, is this is a linksys router |
18:21.51 | TripleFFFF | yes |
18:22.12 | smacku1 | _problem_: it exits the queue fine, and dials the extension fine, but then goes into that extensions voicemail rather than trying the next extension. does the n option change that? |
18:22.13 | Tsop | TripleFFFF, luck i am stuck with a TR32P2 can't get it to unlock |
18:22.22 | TripleFFFF | Register Expires: 300.. could it be after 300 it cant reregister ? |
18:22.35 | _problem_ | smacku1:no |
18:22.53 | _problem_ | u already understood what i means.. |
18:24.05 | Tsop | dlynes_home, so is there is a cheep voip service? and what is the advantage to use asterisk ? |
18:24.26 | smacku1 | _problem_: yeah, it exits to the first extension fine. What I was hoping is that if it did not answer, it would try a second extension rather than leave a voicemail at the first extension. Kind of like a hunt group, but I did not want to go that way if there was another option. |
18:24.52 | *** join/#asterisk freebsd_fan (n=ebola@i-83-67-73-117.freedom2surf.net) |
18:25.52 | _problem_ | smacku1: no dear i dont know other way ..i was also facing the similar problem...which i little bit solved with n option...perhaps somebody else could help |
18:26.30 | *** join/#asterisk brc_ (n=brc_@pdpc/supporter/basic/brc) |
18:26.32 | smacku1 | ok |
18:26.33 | *** join/#asterisk Samoied (n=Samoied@ip47092.static.poa.virtua.com.br) |
18:27.36 | smacku1 | _problem_: so in your scenario using the n option, how does your call flow go? Once you exit the queue, what happens? |
18:29.01 | _problem_ | smacku1: once it exits from a q it goes to an extension..after timout at that extension it goes to another q |
18:29.42 | smacku1 | can you pastebin what that looks like? |
18:30.04 | _problem_ | smacku1: sure hold on |
18:30.17 | Tsop | dlynes_home, i am just wondering what will be the use for asterisk? if i sign up for a voip service i could use my phone rihgt? |
18:31.39 | _problem_ | smacku1: http://pastebin.ca/78026 |
18:32.58 | Tsop | anyone please cleaify , i am just wondering what will be the use for asterisk? if i sign up for a voip service i could use my phone rihgt? |
18:39.04 | TripleFFFF | oh |
18:39.05 | TripleFFFF | dlynes |
18:39.20 | TripleFFFF | i see a REGISTER after the 300 sec and a 401 unauthorized |
18:41.50 | TripleFFFF | maybe not |
18:43.38 | *** join/#asterisk evisu (n=hIRC@bzq-88-152-238-38.red.bezeqint.net) |
18:46.09 | smacku1 | _problem_: thanks for the suggestion, I am going to try the n option and see what it does. |
18:46.23 | smacku1 | based on what I am reading about it, it makes sense that this is the correct way to do what I need. |
18:46.50 | _problem_ | smacku1: ok best of luck...ya |
18:47.56 | *** join/#asterisk clive- (n=pirch@dsl-145-18-135.telkomadsl.co.za) |
18:55.14 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
18:55.59 | *** part/#asterisk Samoied (n=Samoied@ip47092.static.poa.virtua.com.br) |
19:05.07 | Tsop | is it possable i can just use cisco ip phone 7940 instade of tdm card ? |
19:05.35 | clive- | tsop, yes, with ztdummy for timming |
19:06.18 | TripleFFFF | INVITE Expires: |
19:07.52 | Tsop | clive-, please guide me? why will i use asterisk? what is the benifit ? |
19:09.37 | clive- | tsop there are many reasons...for fun, for business, for ivrs, voicemails, etc etc |
19:10.25 | nortex | Tsop, THe reason/benifit of asterisk is to have a full pbx connected to the voip service. If you just want a SIP device to ring when the calls come into you voip service you don't need asterisk. |
19:11.33 | Tsop | i am really very confused right now? |
19:12.35 | clive- | tsop if you just want to call your girlfreind with your ip phone and thats all, you dont need asterisk |
19:12.38 | clive- | lol |
19:12.55 | nortex | clive-, hahaha |
19:13.07 | Tsop | clive-, she is over sea thu |
19:13.08 | TripleFFFF | anyone know how to factory reset a pap2 ? |
19:13.32 | nortex | Tsop, What do you want your IP phone to do? |
19:14.16 | Tsop | TripleFFFF, i found the info u want last night in google, try vonage forum they have a user name and password that will rest to factory |
19:14.30 | TripleFFFF | darn.. the shit pap2 t na is sending me byes at exactly 300 sec.. .every time |
19:14.45 | Tsop | nortex, basiclly to be able to call my familly over sea, and for fun too |
19:17.15 | nortex | Tsop, You won't "need" Asterisk for calling overseas if you have an IP phone/ ATA and a voip service |
19:17.36 | TripleFFFF | tsop not vonage |
19:17.41 | TripleFFFF | its a unloked pap2 t na |
19:17.50 | *** join/#asterisk Trazz (i=Trazz@c-67-163-92-37.hsd1.il.comcast.net) |
19:17.52 | nortex | If you want to have features like clive- mentioned then Asterisk is pretty cool. |
19:17.56 | *** join/#asterisk DarKnesS_WolF (n=wolf@82.201.232.126) |
19:18.04 | clive- | tsop I hope she is not here in south africa, the adsl here sucks for voip |
19:18.30 | Tsop | clive-, no north africa |
19:18.38 | Tsop | and yes it suck there too |
19:18.38 | nortex | Tsop, Have you set your IP phone up for the voip service yet? |
19:18.59 | clive- | tsop welcome to african internet |
19:19.02 | Tsop | nortex, not yet i just started yesturday |
19:19.26 | Tsop | clive-, i know man the last time i was there they only used dial up , lol |
19:19.47 | Tsop | normsteel, thats why i need ur help guys to be directed on the right track |
19:20.09 | Tsop | how can i shop for a voip |
19:23.44 | Tsop | ??? |
19:24.08 | clive- | tsop what do you want to buy |
19:24.58 | Tsop | clive i want to be able to call north africa for the cheapest price what do i need to get , i am in usa now |
19:26.00 | clive- | tsop well you either need to find a voip provider who can offer you cheap calls to the country you need, or you need a callingcard |
19:27.08 | Tsop | clive-, which voip u have |
19:27.40 | clive- | tsop or if theer is decent internet in the place, you can get another voip phone and call hers with yours |
19:28.15 | rob0 | FWD doesn't like me. :( I'm certain the FWD number and password are correct, but my IAX2 registration is being rejected. |
19:29.21 | clive- | -I forgot to "make insta;;"...scary |
19:29.29 | rob0 | I even reset the password! |
19:29.46 | TripleFFFF | so no one know why the pap2 sends a bye every friggin 300 seconds ? |
19:31.25 | *** join/#asterisk mpruett (n=mpruett@24-240-203-82.static.stls.mo.charter.com) |
19:32.10 | Tsop | clive-, i am haveing a hard time to show her how to open skype ? do u think i'll be able to show her how to setup voip |
19:32.13 | Tsop | lol |
19:33.37 | TripleFFFF | shit |
19:33.43 | nortex | Tsop, You may have a problem there :) |
19:33.51 | *** join/#asterisk Johnnie (n=john@pdpc/supporter/active/Johnnie) |
19:34.21 | rob0 | Hmmm, maybe I should try FWD with SIP. |
19:35.41 | *** join/#asterisk postel_ (n=jp@unaffiliated/postel) |
19:36.07 | Tsop | nortex, yeah so thats why i was trying to setup up somethig for cheat |
19:37.55 | smacku1 | so in my cdr i have a cdr record for the extension, as well as the agent. so it has essentially doubled the call record. is there a way around this? |
19:38.20 | Tsop | clive-, can i pm u |
19:39.25 | clive- | sure |
19:42.14 | Tsop | clive-, did u get my pm |
19:43.04 | *** part/#asterisk m4rkl4r (n=markp@outboundemail.uneta.com) |
19:43.42 | rob0 | Hmmm, after a bunch of rejections it just started working ... |
19:45.09 | *** join/#asterisk docelmo (n=docelmo@55-65.126-70.tampabay.res.rr.com) |
19:45.26 | smacku1 | also, do exten => values have to be numeric? or can they be # and * also? |
19:45.45 | docelmo | You can use # or * in the extensions |
19:45.54 | docelmo | Anything sip can pass including alpha values |
19:46.17 | smacku1 | can you see if there is anything that would keep this from working?: exten => #1,1,VoicemailMain(@progrexion) |
19:46.57 | docelmo | ya the @progrexion |
19:47.46 | docelmo | should be user@context |
19:47.49 | docelmo | not @context |
19:48.19 | smacku1 | exten => 8500,1,VoicemailMain(@progrexion) works just fine. The @progrexion specifies which voicemail context to log into |
19:48.20 | [TK]D-Fender | docelmo : No, it should be valid as a way to seperate #'s |
19:48.37 | [TK]D-Fender | docelmo : As he's running amnulti-tennent system |
19:49.13 | docelmo | Hmm.. |
19:49.36 | [TK]D-Fender | docelmo : that way it still promts for the box but in a specific VM context. |
19:49.54 | docelmo | ya.. I thought asterisk didnt handle VM that wa |
19:49.55 | docelmo | way |
19:50.05 | smacku1 | yeah, works great. |
19:50.25 | [TK]D-Fender | smacku1 : only reason it might not work would be dialplan context based. |
19:50.29 | smacku1 | I have 5 different companies all with their own context. working smooth after a weekend of overhaul |
19:50.37 | [TK]D-Fender | smacku1 : Or something stopping the phone from dialing that exten. |
19:51.02 | smacku1 | hmmm. interesting thought, because I get absolutely no CLI output when dialing it. |
19:51.18 | smacku1 | maybe the phone its self |
19:51.29 | smacku1 | xlite soft phone |
19:51.48 | docelmo | I know linksys would kill it. You have to fudge the dialplan in the phone to make that work |
19:52.22 | [TK]D-Fender | smacku1 : either the phone of context setup |
19:52.49 | [TK]D-Fender | docelmo : Linksys shouldn't have anything to say about that at all... its all * dialplan... |
19:53.32 | docelmo | [TK]D-Fender check the Linksys ATA's trust me they will screw with it. I believe the supura's would also. I had a hell of a time getting *98 to work correctly |
19:54.03 | [TK]D-Fender | docelmo : *.T|X.T|#.T |
19:54.29 | [TK]D-Fender | docelmo : I never had any kind of dialplan issue with them.... |
19:54.29 | *** join/#asterisk Vorondil (n=jkiffmey@miro.voltaiccommerce.com) |
19:54.29 | docelmo | smart ass |
19:54.31 | [TK]D-Fender | :D |
19:54.42 | docelmo | I should have asked you 6 months ago |
19:54.45 | [TK]D-Fender | docelmo : You had me at "smart" ;) |
19:55.01 | docelmo | its ok.. I dont work there anymore.. :) |
19:56.10 | docelmo | I work for another ITSP now |
19:56.33 | docelmo | ANYWHO.. any news on astricon yet? speakers and such? |
19:57.10 | skraelings001 | what mean that dchan is provisioned? |
19:57.14 | [TK]D-Fender | docelmo : I can tell you Olle WON'T be there.... |
19:58.25 | file | too early to know aboot speakers... |
19:58.46 | Corydon-w | aboot? |
19:58.50 | [TK]D-Fender | file : Silly Canuckian! |
19:58.53 | file | :D |
19:58.58 | file | [TK]D-Fender: it's beautiful here today! |
19:58.58 | *** part/#asterisk muppetmaster (n=jasongoe@169.red-81-184-73.user.auna.net) |
19:59.17 | [TK]D-Fender | file : Mississauga : If you don't like the weather, wait 10 minutes. |
19:59.27 | file | ha |
19:59.28 | file | nice |
20:00.07 | [TK]D-Fender | file : And dispicably true throughout my vacation. We had EVERYTHING. In a day : heavy fog, clear skies & sun, rain, and .... HAIL. |
20:00.17 | *** join/#asterisk joat (n=joat@ip70-160-147-169.hr.hr.cox.net) |
20:00.36 | [TK]D-Fender | file : I was just waiting for the walls to start bleeding... |
20:00.46 | file | [TK]D-Fender: sacrifice a goat! |
20:00.47 | docelmo | [TK]D-Fender yes sad.. I heard there was a fall out |
20:01.23 | docelmo | Sounds like the weather in Tampa |
20:01.36 | docelmo | Were in the Hurricane/Rainy season right now |
20:04.56 | *** join/#asterisk Arno[Slack] (n=hellSOUN@master.infinityperl.org) |
20:06.50 | *** join/#asterisk jcmoore (n=jcmoore@picard.ojc.nuvio.com) |
20:08.04 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-246-145.buckeyecom.net) |
20:08.46 | gambolputty | Hi. I am trying to install the mysql addons for * trunk and they don't get selected for compiling. Could someone help? |
20:09.33 | TripleFFFF | any idea on pap2 false hangups ? |
20:09.43 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
20:11.37 | *** join/#asterisk Krush (n=Krush@213.9.109.181) |
20:13.48 | Krush | doesnt zaptel support DESTDIR on make install? |
20:14.15 | docelmo | look in the make file |
20:14.34 | docelmo | and why not say hi and chill a bit before hammering with questions.. |
20:14.37 | docelmo | be polite.. :) |
20:14.44 | smacku1 | is there a quick easy to install app out there that an agent on the floor can see if they are logged into a queue or not? |
20:15.00 | docelmo | Nope.. |
20:15.06 | docelmo | Could script one |
20:15.10 | docelmo | fairly easily |
20:15.24 | smacku1 | it would just use the manager interface? |
20:15.25 | Krush | docelmo, ok. sorry |
20:16.06 | websae | Krush: are you going to ClueCon? |
20:16.19 | websae | and what about you docelmo are you planning to attend? |
20:16.35 | [TK]D-Fender | smacku1 : AMI would work, or just dump 'asterisk -rx "show queues"' and parse it out. |
20:16.35 | docelmo | Cluecon? Doubt it. When is it? |
20:16.56 | Krush | but make install with destdir ends up in: /bin/sh: /tmp/package-zaptel//etc/udev/rules.d/zaptel.rules - file does not exist |
20:16.59 | smacku1 | you said dump :-D |
20:17.29 | websae | Aug 1-3 |
20:17.47 | [TK]D-Fender | smacku1 : Yes, like I did to the bodies of those people who pissed me off last week :F |
20:18.39 | smacku1 | ahhhh yes, one of those may have been me? |
20:19.08 | [TK]D-Fender | smacku1 : No, you're still talking, but I give you 5 mins tops ;) |
20:19.16 | smacku1 | ah |
20:19.18 | smacku1 | run |
20:20.10 | smacku1 | thats just what happens when you log in while on vacation ;) |
20:21.05 | [TK]D-Fender | No longer on vacation... back with a mountain of work ready forme. |
20:21.08 | websae | docelmo: i better see you there |
20:21.10 | Krush | the makefile contains INSTALL_PREFIX:=$(DESTDIR) |
20:24.46 | docelmo | I dunno we will see.. Depends mainly on the new company |
20:25.14 | docelmo | They are all about sending me out from what I have seen so far.. Im going somewhere in august then october and possibly VON in september |
20:25.42 | smacku1 | is there a command that shows why an agent has been logged out? |
20:25.50 | dpryo | lol |
20:26.08 | skraelings001 | pls, help with directed pickup application |
20:26.08 | docelmo | no |
20:26.16 | smacku1 | figured as much |
20:26.16 | dpryo | I would like a command that shows why an agent didn't answer the call. |
20:26.22 | smacku1 | i see the error |
20:26.25 | smacku1 | Jul 3 13:54:08 NOTICE[12176] chan_agent.c: Agent 'Dan Black' didn't answer/confirm within 15 seconds (waited 16) |
20:26.35 | *** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it) |
20:26.40 | smacku1 | just wondered if anything showed that like the -rx "show agents" |
20:27.02 | dpryo | grep "didn't answer" /var/log/asterisk/messages |
20:27.02 | dpryo | :D |
20:27.04 | [TK]D-Fender | dpryo : "Agent Busy Masturbating in Opposite Sex's Bathroom, Back Later" |
20:27.09 | docelmo | hay [TK]D-Fender you got any experience with post dialing? like into a PBX or something? |
20:27.20 | [TK]D-Fender | docelmo : "post dialing"? |
20:27.32 | docelmo | like the call is established then you dial thru a IVR to reach someone |
20:27.55 | [TK]D-Fender | docelmo : as in scripted? |
20:28.23 | *** part/#asterisk a1fa (n=a1fa@207.210.210.202) |
20:28.32 | docelmo | yes |
20:28.54 | rob0 | oh wait, that would be an app, not a chan |
20:28.56 | dpryo | hehe |
20:29.02 | Krush | hmm. the zaptel makefile does not seem to work correctly on destdir, as it does not create the necessary subdirs |
20:30.13 | smacku1 | asterisk seems to be running, but I cannot get any output when i type in a cli command? any idea what is up and how to fix it? |
20:30.28 | smacku1 | ie i type: slk-apbx-01*CLI> show agents |
20:30.28 | smacku1 | slk-apbx-01*CLI> show channels |
20:30.28 | smacku1 | slk-apbx-01*CLI> |
20:30.31 | docelmo | Im not sure if its possible.. I keep running shit thru my head and nothing seems to click to make it work |
20:31.59 | smacku1 | without restarting asterisk is there a way to fix it |
20:32.09 | docelmo | isnt there something in the dial command to send DTMF after the call is established? |
20:32.57 | [TK]D-Fender | docelmo : There is an option for post DTMF dial, but nothing "too bright". Not really viable. |
20:33.07 | gambolputty | Hi. I am trying to install the mysql addons for * trunk and they don't get selected for compiling. Could someone help? |
20:33.27 | [TK]D-Fender | docelmo : its just dumps the DTMF instantly, not waiting or detecting connect no really allowing paus between IVR options. |
20:33.45 | [TK]D-Fender | BBIAB, heading home. |
20:37.51 | *** join/#asterisk Cresl1n (n=matt@gateway.digium.com) |
20:38.19 | anthm | you *can* insert w's for 500ms pauses if you wish |
20:41.02 | smacku1 | ok, so if i call my extension where I have AgentCallbackLogin, suddenly it is not working 100%. Nothing has changed since it did work a few minutes ago except for when I ran asterisk -rx "show queues" |
20:41.07 | smacku1 | now when you call it: |
20:41.29 | smacku1 | you get the prompt to log in, but then it never gets to the point where it asks for pin or extension. |
20:41.36 | *** join/#asterisk nagl (n=nagl@86.59.54.237) |
20:41.37 | smacku1 | the call does not die, it just does not do anything. |
20:41.56 | smacku1 | any help? |
20:43.22 | *** join/#asterisk prh (n=paul@X80.mjr.org) |
20:44.51 | smacku1 | anyone? |
20:46.41 | smacku1 | is there a stop command to this? asterisk -rx "show queues" |
20:47.25 | smacku1 | reload app_queues.so gives me the error "the previous command didnt finish yet" |
20:47.34 | smacku1 | I think that may be causing my issue |
20:48.30 | smacku1 | now I am getting the previous reload did not complete yet. |
20:48.33 | smacku1 | wtf |
20:48.36 | smacku1 | any ideas? |
20:50.59 | mrtwister | *CLI> stop nowJul 3 11:50:17 ERROR[11949]: res_config_mysql.c:615 mysql_reconnect: MySQL RealTime: Failed to connect database server asterisk on . Check debug for more info. |
20:51.11 | mrtwister | how to read debug info |
20:51.46 | smacku1 | that is the output you got from a stop now command? |
20:52.00 | Corydon-w | mrtwister: please update your -addons source |
20:52.54 | mrtwister | i installed mysql-dev and compiled addons |
20:53.10 | Corydon-w | When? |
20:53.35 | smacku1 | anyone know how to stop a reload? |
20:53.51 | Corydon-w | smacku1: 'stop now' |
20:53.59 | smacku1 | does that not stop asterisk? |
20:54.03 | Corydon-w | Yes |
20:54.13 | smacku1 | cant stop asterisk 49 calls on the lines |
20:54.19 | smacku1 | i need to stop the reload |
20:54.22 | smacku1 | is it not possible? |
20:54.51 | smacku1 | that has to be a way |
20:54.51 | Corydon-w | Okay, then. You cannot stop the reload without shutting down Asterisk. |
20:54.57 | Corydon-w | Sorry. It wasn't designed to be stoppable. |
20:55.09 | Corydon-w | Then again, it's not supposed to take that long, either |
20:55.12 | smacku1 | crap |
20:55.30 | skraelings001 | mrtwister: are you working with real time ? |
20:56.05 | mrtwister | yes |
20:56.15 | mrtwister | with realtime |
20:56.29 | *** join/#asterisk Greek-Boy (n=Greek-Bo@193.220.93.162) |
20:56.47 | Corydon-w | mrtwister: that particular error message was addressed last week. |
20:57.36 | *** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it) |
20:57.41 | Corydon-w | So update your addons to the latest SVN 1.2. |
20:58.38 | Greek-Boy | whats a good solution for someone that uses multiple phones? to quickly set which phone you're at so someone just has to remember one extension? |
20:59.15 | mrtwister | why i cannoit use stable addons |
20:59.16 | mrtwister | :) |
20:59.22 | *** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it) |
20:59.38 | Corydon-w | mrtwister: there is no such thing as "stable addons" |
20:59.43 | mrtwister | i use 1.2.9.1 |
21:00.05 | Corydon-w | addons are not at release 1.2.9.1 |
21:00.48 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
21:00.54 | Corydon-w | That's why I'm telling you to run the release branch of addons |
21:01.27 | skraelings001 | mrtwister: what is your res_mysql.conf ? |
21:01.44 | clive- | what does chan_phone do, my asterisk keeps failling to laod up on that |
21:01.54 | mrtwister | [general] |
21:01.54 | mrtwister | dbhost = localhost |
21:01.54 | mrtwister | ;dbname = asterisk |
21:01.55 | mrtwister | dbname = billing |
21:01.55 | mrtwister | dbuser = root |
21:01.55 | mrtwister | dbpass = amoeba0819 |
21:01.57 | mrtwister | dbport = 3306 |
21:01.59 | mrtwister | like that |
21:02.15 | PakiPenguin | hello there , i have a tdm2400p , and i need to mark a specific callerid to an incoming fxo channel ( like whenever a call comes in on channel1 , i need it to say 7085441700 on caller id by default , so i can differentiate between calls ) , i did this in zapata-chanels.conf -> http://pastebin.ca/78171 , but its not sending me the number |
21:02.25 | skraelings001 | clive : standard linux telephony API |
21:02.28 | Corydon-w | ~pb |
21:02.30 | jbot | i guess pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/ |
21:02.53 | clive- | skraelings what can cause it to crash on that? |
21:02.58 | PakiPenguin | can anyone help me please |
21:03.12 | [TK]D-Fender | PakiPenguin : Set each FXO channel to have its own incoming context and in the "s" exten, set it at the start. |
21:04.15 | PakiPenguin | [TK]D-Fender, amp :( .. i wanted to do that .. but we have amp here and its needed since the user managing wont be capable at all |
21:04.57 | [TK]D-Fender | PakiPenguin : if you ahve to ask, the answer is "ditch AMP" |
21:05.07 | PakiPenguin | :) yeah i know |
21:05.47 | [TK]D-Fender | PakiPenguin : I presume you can define each channel seperately. From there you should be able to define a "custom script" at the end of which you can "goto" a defined IVR |
21:06.12 | skraelings001 | clive: what does it say when fails? |
21:06.32 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
21:06.52 | skraelings001 | mrtwister: socket exists? /var/run/mysqld/mysqld.sock |
21:07.58 | clive- | skraelings : Jul 3 22:31:45 WARNING[3134]: loader.c:554 load_modules: Loading module chan_phone.so failed! |
21:08.05 | *** join/#asterisk marv0997 (i=marv0997@190.4.2.83) |
21:08.18 | mrtwister | solved. |
21:08.31 | mrtwister | needed to define socket in res_mysql.conf |
21:08.35 | skraelings001 | mrtwister : good! |
21:08.43 | mrtwister | not working witout :) |
21:09.19 | skraelings001 | mrtwister : what is your modules.conf ?? |
21:10.12 | mrtwister | from make samples |
21:10.23 | mrtwister | it is installed just 20 minutes ago |
21:10.32 | mrtwister | at ubuntu dapper |
21:11.34 | skraelings001 | mrtwister: you haven't touched anything since?=+ |
21:12.34 | skraelings001 | mrtwister: how do you start asterisk? asterisk or asterisk -vvvvv |
21:13.24 | *** part/#asterisk Vorondil (n=jkiffmey@miro.voltaiccommerce.com) |
21:13.26 | mrtwister | asterisk, then asterisk -vvvdddr |
21:14.13 | skraelings001 | killall asterisk and try asterisk -vvvvv , what's the output? |
21:14.29 | mrtwister | no, all ok now :) |
21:14.44 | skraelings001 | ok |
21:15.10 | *** join/#asterisk russellb_ (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
21:15.10 | *** mode/#asterisk [+o russellb_] by ChanServ |
21:16.58 | docelmo | ~seen trixter |
21:17.01 | jbot | trixter <n=trixter@65-165-167-217.du.volcano.net> was last seen on IRC in channel #asterisk, 2d 15h 40m 14s ago, saying: 'but the fact that its there and can be spread by bites means that this year they are likely to have more cases where people get infected'. |
21:22.59 | *** join/#asterisk nextime (n=nextime@213-140-6-103.ip.fastwebnet.it) |
21:24.23 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
21:24.33 | *** join/#asterisk nortex (n=breeves@snapper.titanspecialties.com) |
21:26.26 | *** join/#asterisk Mattwj2005 (n=Matt@user-12l3n74.cable.mindspring.com) |
21:26.59 | Mattwj2005 | hey guys...what do you think of net neutrality? |
21:27.10 | file | I'm neutral on the issue. |
21:27.20 | *** join/#asterisk blebleble (i=godie@caesar.godie.net) |
21:27.24 | dlynes_home | is that where your net gets neutered? |
21:28.24 | Mattwj2005 | the telephone company wants a two tier Internet |
21:28.38 | *** join/#asterisk ariel_ (n=Ariel@dsl-20-177.cofs.net) |
21:29.08 | Mattwj2005 | on with high priority traffic for their traffic and companies that pay extra and one for everything else |
21:29.09 | gambolputty | No ISP should block or degrade any service that competes with theirs |
21:29.18 | rob0 | FWD: IAX or SIP? I think I have it working with IAX but it seems flaky, someone on FWD forums said SIP is better. |
21:29.40 | Mattwj2005 | I just wonder how this is going to affect VoIP? |
21:30.03 | dlynes_home | Mattwj2005: *shurg* aol thinks they can get away with it for email |
21:30.08 | gambolputty | Imagine an ISP blocking all UDP 5060 traffic. |
21:30.34 | Corydon-w | That's why God made port 5070 |
21:32.07 | Mattwj2005 | I work in Networking.....I think it is just going to be more headaches |
21:32.54 | dlynes_home | Mattwj2005: phone up the fcc...maybe they'd like to hear about a phone company that wants preferential network treatment |
21:33.17 | rob0 | Maybe they've already been paid. |
21:33.22 | dlynes_home | that, too |
21:33.32 | Mattwj2005 | it America....yeah probably |
21:33.34 | Mattwj2005 | *in |
21:34.47 | Mattwj2005 | The job of the phone company should be to get my voice or data from point A to B.....and then people like myself should be concerned with everything else |
21:35.37 | nortex | But that might cut into their precious profits :0 |
21:36.21 | Mattwj2005 | good point |
21:39.32 | *** join/#asterisk mooodi (i=mooodi@bouncer.ikhost.com) |
21:40.18 | Mattwj2005 | I think in the end...even if it does pass it will eventually bit them in the behind....I am sure they will get a lot of complains from users and companies like the one I work for.....in the end the good will win |
21:40.48 | mpruett | Hello Everyone! |
21:41.01 | Mattwj2005 | hi mpruett :) |
21:41.10 | mpruett | I have a tricky one for you guys |
21:41.47 | mpruett | Please take a look at http://pastebin.ca/78191 - I put the problem here with everything I have done to this point |
21:42.05 | *** part/#asterisk mooodi (i=mooodi@bouncer.ikhost.com) |
21:42.07 | mpruett | Didn't want to clutter up everything in here ;) |
21:43.16 | file | I assume you set nat=yes? |
21:43.21 | mpruett | Yes |
21:43.46 | Mattwj2005 | mpruett where are you from? |
21:44.00 | mpruett | Missery |
21:44.19 | mpruett | or Missouri whichever |
21:44.32 | file | mpruett: well, a packet has to come from the NATted device before the media can be sent to it's non-private IP address and it doesn't look like it sent one |
21:44.35 | Qwell[] | You can't spell your state? |
21:44.40 | file | mpruett: silence suppression on? VAD? |
21:44.49 | Mattwj2005 | oh okay....you have a similar name to one of my old college teachers.....just thought I would check :) |
21:44.52 | *** join/#asterisk gorauskas (n=gorauska@66-224-20-131.atgi.net) |
21:45.01 | mpruett | Play on words - obviously not a funny joke |
21:45.55 | mpruett | file: I do not believe I have silence suppression on |
21:46.27 | mpruett | file: I know how to do this from ATA - Is there a way to do at Asterisk level? |
21:46.42 | file | Asterisk doesn't support it |
21:47.01 | file | but in order for your audio to travel, both sides have to send at least one packet of audio to Asterisk |
21:47.09 | mpruett | file - I have this set to "No" on ATA |
21:47.32 | file | it will then change the destination IP address and port for both sides from it's private IP/port to the NATted public IP/port, and audio should flow |
21:48.51 | *** part/#asterisk gorauskas (n=gorauska@66-224-20-131.atgi.net) |
21:50.17 | smacku1 | is anyone here using e&m wink start T1s? |
21:51.01 | smacku1 | I cannot get the caller id to work with them |
21:51.12 | mpruett | file: How can I tell if this is happening? Does this behavior change with MeetMe? |
21:51.12 | smacku1 | so i am wondering if there is any special configuration that i need to consider. |
21:51.25 | smacku1 | it all works with my pris but not e&m |
21:52.19 | file | mpruett: you look at rtp debug, and when a packet comes in from the phone that is NATted... all outgoing packets should then switch to that source IP address and port |
21:52.46 | file | mpruett: do you have canreinvite=no ? |
21:53.29 | mpruett | file: I have canreinvite=yes |
21:53.52 | file | that's going to cause issues with NATted devices... weird things can happen |
21:53.56 | *** join/#asterisk paolob-parroquia (n=paolob-p@pri-214-b7.codetel.net.do) |
21:54.05 | mpruett | Let me switch and retest |
21:56.05 | paolob-parroquia | Guys, I have my asterisk server working, but I can't understand how can I do the following: I'm the secretary, after answering a call for my boss I must ask him whether to pass him the call or not, but if I perform a transfer I can't ask the boss the permission to pass him the call. How can I do it? |
21:57.10 | file | paolob-parroquia: attended/supervised transfer? |
21:57.43 | mpruett | file: YOU ARE THE MAN!!!!! |
21:57.53 | paolob-parroquia | file, no, if I do an attended transfer, when my boss ansers he get directly the call |
21:58.10 | file | paolob-parroquia: that's a blind transfer |
21:58.14 | mpruett | file: 4 tests all good - will continue to test but thanks for the help!!! |
21:58.24 | *** join/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net) |
21:59.20 | mpruett | file: This is the second time you have helped me out with an issue - I seriously appreciate your help |
22:02.15 | paolob-parroquia | file, ok, I understand. But after the boss answer my attended call, what are the options? |
22:02.45 | smacku1 | does anyone know about callerid on E&M? there references out there that suggest it is done differently than normal, but the link I found to the details is dead |
22:02.56 | *** join/#asterisk jcmoore (n=tgrman@c-71-199-75-134.hsd1.ks.comcast.net) |
22:03.29 | file | paolob-parroquia: I've never done that kind of transfer lol |
22:03.56 | blebleble | anyone ever see 'Unable to pass the full buffer onto the device file. -1 bytes of 2 written: Resource temporarily unavailable' from iaxmodem? |
22:04.12 | smacku1 | are you referring to doing a "warm transfer"? where you announce the transfer, then hang up the call? |
22:04.21 | smacku1 | paolob-parroquia: |
22:04.26 | file | warm... attended... supervised... |
22:04.30 | [TK]D-Fender | paolob-parroquia : Attended transfer means you call the 3rd party, they see YOUR caller ID, they answer, you ask if they want the call, if so you typically press the transfer button again and the call gets passed off |
22:04.54 | smacku1 | that is what i was typing |
22:06.37 | *** join/#asterisk BZBW (n=wlwzhang@ip67-153-142-109.z142-153-67.customer.algx.net) |
22:07.56 | paolob-parroquia | [TK]D-Fender, ok, thnx |
22:09.29 | smacku1 | after a while of asterisk running, I get the issue where stuff just doesnt do anything when i type it. Like: |
22:09.29 | smacku1 | slk-apbx-01*CLI> reload |
22:09.29 | smacku1 | slk-apbx-01*CLI> show channels |
22:09.29 | smacku1 | slk-apbx-01*CLI> |
22:09.45 | smacku1 | when an agent tries to log in at this point it only prompts them for there agent id, and then does nothing. |
22:09.49 | smacku1 | what could be happening. |
22:09.54 | smacku1 | second time in two hours |
22:10.03 | smacku1 | no one is doing anything on the system except using it |
22:17.09 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-63-230.cybersurf.com) |
22:17.19 | *** part/#asterisk mpruett (n=mpruett@24-240-203-82.static.stls.mo.charter.com) |
22:20.14 | *** join/#asterisk |marv0997| (i=marv0997@190.4.2.86) |
22:20.18 | smacku1 | http://pastebin.ca/78217 |
22:20.34 | smacku1 | are the calls that are down supposed to still be there? |
22:21.33 | *** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com) |
22:22.00 | smacku1 | can i kill calls from the cli? |
22:24.00 | smacku1 | did everyone leave? |
22:24.09 | clive- | smaku soft hangup |
22:24.34 | smacku1 | ok, and do i specify a channel? or extension? |
22:25.08 | clive- | yes |
22:25.15 | clive- | try it until it works:) |
22:25.17 | smacku1 | looks like channel |
22:28.46 | smacku1 | ok, so i have done soft hang up on all three lines that show down, ring and down. if that did not kill them, whats next to try? |
22:29.12 | [TK]D-Fender | smacku1 : "restart now" |
22:29.26 | smacku1 | suck! second time in 2 hours. |
22:29.35 | smacku1 | any idea as to why this keeps locking up? |
22:29.40 | smacku1 | I think it is that same extension |
22:33.11 | *** join/#asterisk luke-jr_ (n=luke-jr@2002:1891:f657:0:20e:a6ff:fec4:4e5d) |
22:33.48 | dlynes_home | [TK]D-Fender: a101 is working nice and smooth on our main softswitch now :) |
22:34.10 | dlynes_home | [TK]D-Fender: no more spurious hdlc framing errors |
22:34.58 | [TK]D-Fender | dlynes_home : Good to hear. |
22:35.14 | [TK]D-Fender | dlynes_home : Hints on the cause? |
22:39.51 | smacku1 | this looks to me like the calls are all locking up on the queuing and agent logins. am I seeing this correctly? http://pastebin.ca/78237 |
22:42.37 | *** join/#asterisk bigmac4444 (n=mtur2848@CPE-124-177-67-147.qld.bigpond.net.au) |
22:43.09 | bigmac4444 | good morning/evening all |
22:45.18 | bigmac4444 | i have a quick question |
22:46.35 | bigmac4444 | i need to redirect an inbound pstn call to another number. |
22:50.39 | bigmac4444 | *grabs a number and standings in line |
22:50.53 | bigmac4444 | stands* |
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22:56.03 | Bullseye_Network | bigmax4444: like this? http://pastebin.ca/78254 |
22:56.12 | Bullseye_Network | mac not max |
22:56.13 | Bullseye_Network | lol |
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23:07.24 | dlynes_home | [TK]D-Fender: probably shitty hardware, defective card, or slow cpu |
23:07.29 | dlynes_home | [TK]D-Fender: or a combination thereof |
23:10.59 | [TK]D-Fender | All hail the C3! ;) |
23:11.24 | dlynes_home | [TK]D-Fender: not that bad :) |
23:11.55 | [TK]D-Fender | Dunno... C4 is at least explosive ;) |
23:12.01 | dlynes_home | Celeron 2Ghz |
23:12.18 | dlynes_home | running 10 g729 channels |
23:12.49 | [TK]D-Fender | not terrible... |
23:13.06 | dlynes_home | and even then most of the time |
23:13.25 | rob0 | Oh man, I had a helluva time trying to get * working on a C3. |
23:13.33 | dlynes_home | I was usually maxing out at three channels (7 of those licences I don't think were ever used) |
23:13.58 | dlynes_home | rob0: I've had quite good experience getting it to work on a C3 |
23:14.23 | dlynes_home | rob0: but try to use any digium hardware on it, forget it |
23:14.34 | dlynes_home | rob0: the shared interrupts kill the digium hardware |
23:14.38 | rob0 | I failed because the TDM400 card didn't work on that motherboard. Not truly PCI 2.2 compliant. |
23:14.44 | rob0 | yes |
23:15.32 | rob0 | My newest * box is x86_64 ... heaven! |
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23:20.54 | bryanfe | Question - after installing addons, is there something more I need to do to get musiconhold to use format_mp3? I have confirmed that the module is being loaded but musiconhold seems to be complaining that it can't launch the thread |
23:21.43 | [TK]D-Fender | bryanfe : What "mode" is it using? |
23:22.13 | bryanfe | mode=quietmp3 |
23:22.37 | [TK]D-Fender | bryanfe : then you're not set for using Native MoH. That uses MPG123. You should be using "mode=files" |
23:22.46 | *** part/#asterisk smacku1 (n=smackus@c-67-169-248-217.hsd1.ut.comcast.net) |
23:23.06 | dlynes_home | mpg123 is eeeeeeeeeeeeevilllll!!!! |
23:23.09 | bryanfe | got it, will try that. (sample musiconhold.conf didn't really say that) |
23:23.28 | [TK]D-Fender | dlynes_home : not as bad as.... *coughs* telnet |
23:23.28 | Nugget | telnet is eeeeeeevil! |
23:23.46 | [TK]D-Fender | bryanfe : Yes it did ;) |
23:23.52 | dlynes_home | telnet doesn't go into a zombie process :) |
23:24.01 | bryanfe | (where? I may be dense) |
23:24.10 | [TK]D-Fender | dlynes_home : Even starting telnet means you're already a zombie ;) |
23:24.24 | dlynes_home | [TK]D-Fender: nah |
23:24.38 | dlynes_home | [TK]D-Fender: when you need to fill your craving for mud, and a mud client isn't handy |
23:24.42 | bryanfe | still getting this error, hmph... Jul 3 18:24:08 NOTICE[10985]: res_musiconhold.c:511 monmp3thread: Request to schedule in the past?!?! |
23:24.43 | dlynes_home | [TK]D-Fender: you pretty much have to use telnet |
23:25.10 | dlynes_home | bryanfe: restart when convenient |
23:25.11 | [TK]D-Fender | bryanfe : Thats because its still trying to use MPG123... shut * down completely and restart. |
23:25.18 | bryanfe | 10-4 |
23:25.28 | [TK]D-Fender | "restart with reckless abandon" |
23:25.38 | [TK]D-Fender | "restart with extreme prejudice" |
23:25.43 | dlynes_home | restart NOW |
23:25.55 | [TK]D-Fender | ":D |
23:26.02 | Qwell | [TK]D-Fender: I like that last one |
23:26.30 | [TK]D-Fender | Qwell : I figured you would :) |
23:26.36 | bryanfe | liftoff, we have MOH. |
23:26.41 | bryanfe | thank you ;) |
23:26.50 | [TK]D-Fender | bryanfe : quite welcome |
23:26.59 | Qwell | [TK]D-Fender: on a semi-related note... |
23:27.07 | Qwell | on this sunfire, you can do `poweroff -fy` |
23:27.10 | [TK]D-Fender | dlynes_home : That was well rought up... too bad you weren't :) |
23:27.19 | Qwell | I've decided that the 'fy' stands for "fuck you" |
23:27.22 | [TK]D-Fender | Qwell : cute |
23:27.27 | Qwell | force, yes :D |
23:27.32 | [TK]D-Fender | Qwell : that wa my interpretation as well |
23:27.35 | Qwell | heh |
23:27.37 | dlynes_home | Qwell: i'll have to let l-fy know that :) |
23:27.44 | Qwell | dlynes_home: like-fuckyou |
23:27.47 | Qwell | ? |
23:27.59 | dlynes_home | l-fy, the moderator in #yate |
23:28.06 | Qwell | dlynes_home: yes, I know |
23:28.08 | dlynes_home | she often comes in here |
23:28.10 | dlynes_home | ah |
23:28.14 | Qwell | oh...she...oops :D |
23:28.36 | dlynes_home | she just comes in here, when she wants to stir up shit :) |
23:28.38 | bryanfe | is SetMusicOnHold(none), where "none" is a class pointing to /dev/null, really the correct way to shut off MOH? Seems a little brutish to me ;) |
23:29.40 | [TK]D-Fender | bryanfe : Kill the class |
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23:30.14 | bryanfe | i'm looking at the example here: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+MusicOnHold |
23:31.09 | [TK]D-Fender | bryanfe : You don't normally turn off MoH.... |
23:31.24 | m_a_g_o | good evening folks, I'm trying to install *'s oh323 channel, but keep getting: H.323 listener creation failed... any idea/advice on this particular error please? =) |
23:31.29 | Juggie | bryanfe, do you want Moh for NO users? |
23:31.31 | Juggie | or just not for some. |
23:32.02 | [TK]D-Fender | ok, time for volleyball, later all. |
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23:32.16 | bryanfe | here's what I need: 1) turn on MOH, 2) Dial() with 30 sec timeout (it won't answer). 3) Wait(20), 4) Turn off MOH, and (5) Dial again (it will answer this time) |
23:32.42 | bryanfe | legacy stuff needs to be triggered with that first dial, I'm looking for MOH while the user waits |
23:33.36 | dlynes_home | m_a_g_o: turn on more logging so you can see what the real error is? |
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23:34.52 | m_a_g_o | thks, let me do that... brb |
23:36.30 | m_a_g_o | actually that is all there is to it, with debug error and warning enabled |
23:39.36 | bryanfe | I may be misunderstanding how this all works but shouldn't the Wait() command play MOH while waiting, if the MOH class was previously set with SetMusicOnHold()? |
23:41.17 | bryanfe | or maybe this here dummy should try the WaitMusicOnHold() command. |
23:42.40 | wunderkin | bryanfe, the dial is probably screwing with it.. you can dial with moh, after that yeah you will want to do waitmusiconhold |
23:43.58 | bryanfe | trying it now.. |
23:46.21 | bryanfe | Dumb question maybe, but the -addons music on hold mp3 files -- are they, like, donated to the Asterisk project royalty-free (the music, I mean), by the artists? |
23:47.14 | Qwell | bryanfe: there is a license file |
23:47.43 | Qwell | doc/README.fpm I think it is, in 1.2 |
23:48.33 | hads | doc/musiconhold-fpm.txt |
23:48.36 | bryanfe | no such file is in the current addons distry |
23:48.37 | bryanfe | distro |
23:48.42 | Qwell | bryanfe: It's in asterisk |
23:48.45 | Qwell | hads: That's trunk |
23:48.54 | hads | Ah, my bad. |
23:53.29 | RoyK[at] | <PROTECTED> |
23:54.01 | RoyK[at] | ~lart Qwell |
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