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| 00:25.56 | Get_The_Fish | Does anyone know where the help files on applications and functions that are in the console are located? |
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| 00:34.18 | *** part/#asterisk nny (~Scott@174.107.201.103) |
| 00:40.38 | thehar | frolics around |
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| 01:10.00 | ChannelZ | Get_The_Fish: they are in the modules themselves |
| 01:12.17 | ChannelZ | (the source code that is). They get pulled out and compiled into a giant XML file somewhere |
| 01:13.38 | ChannelZ | here it is in /var/lib/asterisk/documentation - in 1.6 anyway |
| 01:14.37 | *** join/#asterisk grabes (~grabes@70.15.27.211.res-cmts.sm.ptd.net) |
| 01:15.45 | grabes | I am looking to start deploying a newer version of asterisk for our new users moving forward. Should I consider 1.8, or stick with 1.6.2? |
| 01:16.23 | jdoe | I use 1.8 for my personal stuff, I'm still on 1.6.2 for work. |
| 01:16.44 | drmessano | 1.8 ROX MY SOX |
| 01:17.11 | jdoe | grabes: if the new features are worth it and you can live that it's still a bit new, *shrug* |
| 01:19.36 | carrar | grabes, use 1.8 |
| 01:19.44 | carrar | unless you find something that doesn't work |
| 01:20.00 | carrar | which of course then report it |
| 01:21.39 | carrar | You'll be glade you on 1.8 eventually |
| 01:21.45 | carrar | you're |
| 01:21.51 | carrar | err |
| 01:27.01 | grabes | I will give it a shot |
| 01:34.01 | grabes | Anyone having newt library issues on FreeBSD even though the package is installed, and it appears the configure found it |
| 01:34.23 | grabes | happens on gmake menuselect |
| 01:35.29 | Get_The_Fish | ChannelZ, yeah just found that in the source file. I am thinking about making myself some syntax helptags in asterisk |
| 01:36.34 | grabes | removing the lib for newt solved it :) |
| 01:41.02 | WIMPy | prefers the menuselect without newt. |
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| 02:02.52 | jaytee | ~itsplist-us |
| 02:02.52 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 02:10.24 | phix | hi gang |
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| 02:42.43 | newmyth | in * AMI, when I specify a channel, how do I do that - do I need the entire channel string? |
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| 04:08.42 | newmyth | anyone here familiar with the AMI ? |
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| 04:44.18 | drmessano | Any word on getting 1.8.1 out the door? |
| 04:55.59 | tyman | newmyth: I'm not but the adhearsion source would probably make great docs on the subject |
| 04:57.56 | erinspice | newmyth: https://wiki.asterisk.org/wiki/display/AST/AMI+Actions |
| 05:08.35 | *** join/#asterisk mbr661 (~mbr661@h184-60-132-106.mdsnwi.tisp.static.tds.net) |
| 05:17.55 | mbr661 | I'm using asterisk for voice chat (Openfire/Spark) but I have the following problem. My clients can only make one chat/call. To place a second call, they have to exit the program altogether and then reconnect. It looks like the call does not disconnect when they hang up. Any ideas what I can do to solve this problem? |
| 05:23.19 | kaldemar | mbr661: change the soft phone to something that works. |
| 05:25.32 | mbr661 | So you don't think it is asteris messing up, but the apps I'm using? Is there a setting in asterisk to control call disconnection time? |
| 05:26.06 | WIMPy | What's that supposed to be? |
| 05:26.07 | *** join/#asterisk [netman] (~netman@184.Red-83-52-210.dynamicIP.rima-tde.net) |
| 05:26.53 | WIMPy | Anyway, if that app can't place a 2nd call without quitting and restarting, it's pretty obvious where the problem lies. |
| 05:28.16 | mbr661 | The app does not quit and restart. I do that to get it to place another call. If I don't restart it, the call cancels |
| 05:28.59 | WIMPy | So what did I say? |
| 05:29.51 | mbr661 | That the app "crashes" after or during a call. It does not. |
| 05:30.23 | WIMPy | Where did I say that? |
| 05:31.02 | kaldemar | maybe it is responsive but it doesn't seem to work. replace it. |
| 05:32.32 | mbr661 | Ok, I'll try looking into settings on the app. I just wanted to rule out anything I might have missed with asterisk. |
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| 06:14.28 | jkroon | hmm, just had a strange idea. is it possible to 302 redirect a REGISTER request? (probably not using asterisk, but is it legal) |
| 06:15.24 | jkroon | basically the idea is to attach an anycast addy to multiple servers, clients register to that IP, but then receives a 302 redirect to the servers "real" IP address. |
| 06:21.23 | florz | what would a redirect even mean in that case? |
| 06:22.30 | florz | You are aware that SIP messages aren't addressed to IP addresses? |
| 06:22.55 | jkroon | yes. |
| 06:24.02 | jkroon | well, consider this, you register to user@voip.uls.co.za, voip.uls.co.za resolves to an anycast address, which obviously goes to the closest available server, now, if during a conversation you move over a routing boundary, or the routing change in such a way as to cause the anycast to go to a different IP you're stuffed. |
| 06:25.02 | florz | "anycast to go to a different IP"?! |
| 06:25.13 | florz | that doesn't exactly make sense |
| 06:25.16 | jkroon | so the idea is to redirect the registration to user@closestip or user@server-a.voip.uls.co.za (whichever), along with any INVITE requests going to the anycast address. |
| 06:25.29 | jkroon | anycast goes to closest server. |
| 06:25.51 | jkroon | so you litterally have 192.168.0.0/24 in multiple DCs (replace with public IP range of choice) |
| 06:26.06 | florz | I know what anycast is |
| 06:26.35 | florz | and I know that "IP" is not a synonym for "server" |
| 06:26.35 | jkroon | ok, cool. |
| 06:27.07 | jkroon | why not? (and yea, i'm looking for flaws in this) |
| 06:27.21 | florz | what "why not?"? |
| 06:33.26 | jkroon | you're saying it doesn't actually make sense, why not? |
| 06:34.03 | jkroon | mysql is starting to become too slow. |
| 06:34.16 | jkroon | 36s to lookup a single dst number... |
| 06:34.38 | jkroon | nm, dst doesn't have an index on it. |
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| 06:48.01 | kaldemar | jkroon: a registrar may redirect a register message. see sip rfc, 10.3: http://www.ietf.org/rfc/rfc3261.txt |
| 06:50.32 | jkroon | kaldemar, thank you ever so much. |
| 06:53.35 | *** join/#asterisk timahvo1 (~rogue@41.72.215.94) |
| 06:59.58 | jkroon | kaldemar, busy reading that RFC now, but it looks like a simple enough matter, the question that pops into my head is this: after a client has received a redirect (presumably 302), will it keep sending INVITEs to the anycast address, or will it send it to the registered location? Or is this left unspecified and up to the UA? |
| 07:00.21 | jkroon | (not serious, it just means I will need to redirect both INVITE and REGISTER if I do decide to try this) |
| 07:05.05 | kaldemar | jkroon: well, registrar is just a registrar. calls don't necessarily have to go to the same location where the registration was made. |
| 07:05.53 | jkroon | yea, that's what i'm thinking too, however, i've seen some pretty messed up shit on some voip phones :p |
| 07:14.52 | kaldemar | clients will probably use the address they're configured with, and follow the possible redirections they get. |
| 07:19.03 | florz | jkroon: you really should get up to speed with the abstraction layers you are dealing with |
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| 07:20.39 | florz | jkroon: as far a SIP is concerned, you are "dialing" (INVITEing, that is) a SIP URI, and thus the INVITE is sent to that SIP URI |
| 07:23.17 | florz | jkroon: and the "doesn't make sense" referred to the "anycast to go to a different IP" - routing doesn't rewrite IP addresses |
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| 07:40.12 | spenguin[work] | test |
| 07:40.51 | spenguin[work] | hey, im seeing cases where the call does not continue in the dialplan if the call was made over pri |
| 07:41.20 | spenguin[work] | after Dial() I mean |
| 07:41.24 | spenguin[work] | is this common? |
| 07:42.12 | spenguin[work] | or what could be the cause of this, im on asterisk 1.4 |
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| 07:52.03 | kaldemar | spenguin[work]: pastebin a cli output for a call |
| 07:52.53 | kaldemar | spenguin[work]: you might have priority jumping on for example, but without seeing an actual call, there's too much guessing to do. |
| 07:54.39 | spenguin[work] | kaldemar: ok |
| 07:54.48 | spenguin[work] | well I think I may have figure the cause |
| 07:55.04 | spenguin[work] | "Note that on a successful connection, in the absence of the g and G modifiers (below), the Dial command does not return to allow execution of further commands for that extension in that context." |
| 07:55.51 | jkroon | yea, that was a kicker on early 1.6 versions it would still return. |
| 07:56.29 | *** join/#asterisk lftsy (~lftsy@install.deckpoint.ch) |
| 07:57.37 | spenguin[work] | jkroon: thing is it returned on a successful test call |
| 07:58.49 | spenguin[work] | oh and I do have g as a parameter |
| 07:59.04 | spenguin[work] | kaldemar: ill post logs |
| 07:59.04 | jkroon | spenguin[work], interesting. Oh yea, now I remember, it depended on which side hung up first. |
| 07:59.22 | kaldemar | spenguin[work]: what exact version are you using? |
| 07:59.23 | jkroon | if it was the caller Dial() would not return, if it was the called party it would return. |
| 07:59.51 | jkroon | with or without that g or G parameter. |
| 08:00.20 | spenguin[work] | kaldemar: Asterisk 1.4.35 |
| 08:00.27 | jkroon | thus I used to just wack a ExecIf(["${DIALSTATUS}"="ANSWER"]?Hangup()) directly after the Dial() |
| 08:00.59 | spenguin[work] | jkroon: would it even work if Dial doesnt return? |
| 08:01.20 | spenguin[work] | and in my test, the remote party hung up |
| 08:01.23 | spenguin[work] | first |
| 08:01.29 | spenguin[work] | ill test the other way |
| 08:03.16 | spenguin[work] | jkroon: ok in my case, if the remote party hangs up first - itll return |
| 08:03.38 | spenguin[work] | but not if the caller hangs up first |
| 08:03.41 | spenguin[work] | this a bug? |
| 08:03.51 | kaldemar | no, a feature. |
| 08:04.01 | spenguin[work] | kaldemar: howso |
| 08:04.23 | spenguin[work] | how do I get dial to return if the caller hangs up first |
| 08:04.24 | kaldemar | core show application Dial: option g |
| 08:04.31 | *** join/#asterisk [netman] (~netman@81.47.145.175) |
| 08:04.35 | spenguin[work] | ok, reading up |
| 08:05.02 | kaldemar | you don't. i used to use hangup extension for that kind of stuff on ye olde versions. |
| 08:06.25 | kaldemar | F option was introduced at some point to proceed if the source channel hangs up. |
| 08:06.39 | spenguin[work] | yep just checked that out |
| 08:06.40 | spenguin[work] | ill try it |
| 08:06.54 | kaldemar | don't bother, your version doesn't have it. |
| 08:07.00 | spenguin[work] | :( |
| 08:07.15 | spenguin[work] | kaldemar: how does one figure? |
| 08:07.22 | spenguin[work] | or how did you figure |
| 08:07.32 | spenguin[work] | if the option applies to my version or not |
| 08:07.52 | kaldemar | F is a 1.6.2 option. |
| 08:07.58 | spenguin[work] | ok |
| 08:08.23 | kaldemar | spenguin[work]: the app documentation for 1.4.35 doesn't have the option. that's how i figured it out. |
| 08:08.31 | spenguin[work] | G(context^exten^pri) seems nice |
| 08:09.08 | kaldemar | that's why it's always best to read the application docs from CLI, you'll get the correct options for your version. |
| 08:09.15 | spenguin[work] | ok |
| 08:09.49 | kaldemar | or from the source code of different versions. http://svn.digium.com/svn/asterisk/tags/ is a good place for quick checks on different versions. |
| 08:10.00 | spenguin[work] | ok |
| 08:10.53 | jkroon | spenguin[work], no, but note that I use it to explicitly hangup the source if the destination answered, but hung up first - which is exactly what I needed to achieve. |
| 08:12.16 | spenguin[work] | ah ok |
| 08:12.34 | *** join/#asterisk bjornts (~BTS@2001:700:200:10:c62c:3ff:fe16:a044) |
| 08:12.46 | spenguin[work] | In my case I want to ensure it returns if the caller hangs up first |
| 08:13.18 | *** join/#asterisk tamiel (~tamiel@213.30.183.226) |
| 08:13.24 | spenguin[work] | ill try the G option, would G return back to the main contesxt once done? |
| 08:14.59 | spenguin[work] | "You cannot use any additional action post answer options in conjunction with this option." |
| 08:15.06 | tyman | Can anyone please point me to the most efficient, yet accurate, guide to server configuring polycom handsets? |
| 08:15.13 | spenguin[work] | does that mean it wont return to the main context |
| 08:15.25 | kaldemar | spenguin[work]: keep in mind that the callee is also transferred to specified exten and priority+1. |
| 08:15.36 | spenguin[work] | hrm |
| 08:15.39 | *** join/#asterisk schmidts (~schmidts@lmlo.sil.at) |
| 08:15.41 | schmidts | good morning |
| 08:15.57 | spenguin[work] | afternoon here :) |
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| 08:19.27 | spenguin[work] | kaldemar: what are my options here, im a bit confused |
| 08:24.02 | kaldemar | spenguin[work]: what exactly do you want to do after the caller hangs up? do you want to do something if it is the callee that hangs up? |
| 08:28.06 | *** join/#asterisk [netman] (~netman@83.54.227.40) |
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| 08:31.05 | spenguin[work] | kaldemar: in either case I want it to continue in the dialplan |
| 08:31.46 | kaldemar | maybe upgrade to 1.6.2 or use a hangup extension when the caller hangs up. |
| 08:34.46 | *** join/#asterisk centoslinux (~centoslin@1x-193-157-206-125.uio.no) |
| 08:35.31 | spenguin[work] | kaldemar: Im not sure what you mean by "use a hangup extension when the caller hangs up", since in my outbound dialplan and it wont return after Diall(${PRI_TRUNK}/${EXTEN:3},30,Ttg) if the caller hangs up |
| 08:35.41 | *** join/#asterisk SeTTleR (~bernd@p5DDEE103.dip.t-dialin.net) |
| 08:35.45 | SeTTleR | hi |
| 08:36.47 | spenguin[work] | how do I detect a hangup if Dial() wont let me continue in my dialplan |
| 08:38.00 | kaldemar | spenguin[work]: a hangup extension is a standard extension (http://www.voip-info.org/wiki/view/Asterisk+standard+extensions, exten => h,1,...) that gets executed when a call is hung up. |
| 08:38.47 | spenguin[work] | kaldemar: oh, my bad ill check this out thanks |
| 08:39.37 | schmidts | spenguin[work] be careful some things doesnt work in the h exten ;) |
| 08:40.17 | kaldemar | https://wiki.asterisk.org/wiki/display/AST/Handling+Special+Extensions |
| 08:43.21 | schmidts | kaldemar thx i have never heard of an 'a' or 'o' extension :D |
| 08:44.42 | spenguin[work] | schmidts: yep |
| 08:44.53 | schmidts | is this something new in 1.8? |
| 08:45.13 | spenguin[work] | what would be the scope of the h exten I define it within a contex |
| 08:45.26 | spenguin[work] | we have multiple outbound contexts |
| 08:46.16 | kaldemar | spenguin[work]: are they included or do you use Goto? |
| 08:46.17 | schmidts | spenguin[work] the h exten must be in the same context as the dial is located, so you should have one for each outbound context but you can just use a goto there |
| 08:47.07 | kaldemar | if the outbound contexts are included from another, you can put a single h exten in the including context. |
| 08:47.15 | WIMPy | schmidts: No they are at least 1.4. Only used by voicemail, I think. |
| 08:47.23 | spenguin[work] | kaldemar, schmidts ok |
| 08:47.44 | kaldemar | if you use a Goto, the "base context" changes and the previous ones no longer matter. |
| 08:48.27 | spenguin[work] | kaldemar: the different outbound contexts are in the same file just, im just using specific prefixes to use those contexts |
| 08:48.41 | schmidts | wimpy thx. atleast they are only usefull in voicemail buts still interesting it exists :D atleast it would be a nice idea for IVRs too ;) |
| 08:49.59 | spenguin[work] | kaldemar: something like exten _10.,1, is under its own context |
| 08:49.59 | WIMPy | In your IVRs you can do anything you like. I always used a to jump from voimail to voicemailmain. |
| 08:50.27 | spenguin[work] | so different prefix numbers allow me to dial out using different contexts |
| 08:50.31 | kaldemar | spenguin[work]: the used file is irrelevant, what matters is how the parent context sees them. either with include or some form of goto. |
| 08:50.42 | spenguin[work] | include |
| 08:51.16 | *** join/#asterisk bip (~bip@unaffiliated/bip) |
| 08:51.18 | kaldemar | then you can use a single h extension in the including context. |
| 08:51.28 | kaldemar | try it out, you'll see how it works. |
| 08:51.32 | bip | good morning kaldemar |
| 08:51.44 | schmidts | wimpy this reminds me to a simpson episode where homer gots to fat to press a single digit on his phone, and the voice says please press all keys with your whole hand cause your fingers are too fat to dial :D |
| 08:51.47 | kaldemar | morning. did you get your card working? |
| 08:51.57 | spenguin[work] | ok thanks kaldemar |
| 08:52.28 | bip | yes I managed to get it working ... |
| 08:52.45 | bip | but then i screwed everything up again :( |
| 08:53.05 | bip | but it s still a bold step foward ;-) |
| 08:53.31 | schmidts | wimpy http://www.youtube.com/watch?v=8DtbPOXFk00 i meant this :D |
| 08:55.41 | spenguin[work] | kaldemar: ah nice, works :) |
| 08:56.09 | WIMPy | schmidts: Did you have some traumatic experiences with voicemail? ;-) |
| 08:56.31 | schmidts | wimpy LOL |
| 08:56.31 | bip | I have a moduled misconfigured as loopstart, there is no way to talk it into becomin kewlstart using the freepbx GUI |
| 08:57.29 | kaldemar | bip: change it by hand, it's a pice of cake. |
| 08:57.32 | SeTTleR | hi, is there a way to get dahdi debug messages? i read something about dahdi debug or something in the cli, but i don't have that call. i'm using dahdi 2.3.0. can i somehow enable debugging for dahdi? i tried modprobe dahdi debug=1 but that did not change anything. |
| 08:58.15 | kaldemar | bip: just change fxsls to fxsks in /etc/dahdi/system.conf or whereever it is configured. |
| 08:58.15 | bip | I will try kaldemar |
| 08:58.40 | WIMPy | SeTTleR: That writes stuff to the kernel log. You can set it to higher values if needed. |
| 08:58.49 | bip | shall I unload reload the module afterward ? |
| 08:59.37 | kaldemar | bip: hold on a sec, is it a dahdi channel or asterisk channel that is misconfigured? |
| 08:59.43 | bip | in that file they are correctly labeled kaldemar |
| 08:59.57 | bip | dadhi I think |
| 09:00.06 | kaldemar | is it wrong in dahdi-channels.conf? |
| 09:00.31 | SeTTleR | WIMPy, ok, i'll try that. but what i want is something like sip set debug, where i can see what happens when i try to place a call via dahdi. is there any way to get this? |
| 09:01.04 | bip | http://pastebin.com/4iWxYs63 |
| 09:01.38 | WIMPy | SeTTleR: dahdi kan be very different things. |
| 09:01.59 | bip | no kaldemar appears right there as well too |
| 09:02.14 | kaldemar | bip: ok, so dahdi-channels.conf is what you need to fix. change signalling=fxs_ls to signalling=fxs_ks for channel 1. and reload the module. |
| 09:02.57 | bip | signalling=fxs_ks |
| 09:02.57 | bip | callerid=asreceived |
| 09:02.57 | bip | group=1 |
| 09:02.57 | bip | context=from-pstn |
| 09:02.57 | bip | channel => 1 |
| 09:03.04 | kaldemar | SeTTleR: if you have an analog interface, no. in case of BRI/PRI, yes. |
| 09:03.13 | SeTTleR | WIMPy, what do you mean? |
| 09:03.21 | bip | it s already fxs_ks :( |
| 09:03.22 | SeTTleR | ah ok, i have a tdm400p |
| 09:03.30 | *** join/#asterisk a1fa (~a1fa@unaffiliated/a1fa) |
| 09:03.33 | WIMPy | SeTTleR: See kaldemar |
| 09:03.37 | kaldemar | bip: and still you get that error? |
| 09:03.47 | bip | lemem check |
| 09:04.05 | SeTTleR | thx kaldemar. then i'll try to go with debug=10 when modprobing dahdi and wctdm. |
| 09:04.10 | SeTTleR | thx WIMPy |
| 09:04.14 | WIMPy | SeTTleR: If you're not digital, I guess the kernel debug is the only thing you can activate. |
| 09:04.42 | schmidts | hates this stupid customers :( yes asterisk is easy to install and getting a trunk account is even easier and i dont need to use a password for a user like 100. But why is my system calling Afghanistan Mobile??? GRRRRR |
| 09:04.59 | bip | http://pastebin.com/kDs1EyPA |
| 09:05.14 | SeTTleR | ok, i'll check that out. thx |
| 09:05.21 | kaldemar | bip: and no conflicting values exist in chan_dahdi.conf, chan_dahdi_general.conf, chan_dahdi_groups.conf or chan_dahdi_additional.conf? |
| 09:05.44 | bip | i will go check n report back sir ;-) |
| 09:07.34 | kaldemar | bip: if not, maybe the signalling change requires a restart, some configuration options do. either restart asterisk with an init script or "core restart now" in CLI. |
| 09:07.52 | bip | chan_dahdi_groups.conf is wrong, I m gonna fix that now |
| 09:09.57 | ChannelZ | damnit why won't DTMF log |
| 09:09.58 | bip | ok now just says mod 2 and 3 unplugged ... |
| 09:10.40 | bip | http://pastebin.com/vvurkfwu |
| 09:15.57 | kaldemar | bip: are they still unplugged? |
| 09:16.57 | bip | they are |
| 09:17.09 | bip | now i think i figured the problemù |
| 09:17.30 | bip | it there a document that describes those configuration files ? |
| 09:18.13 | bip | I think the chan_dadhi_group.conf file is missing a correct pinter to the group |
| 09:18.24 | bip | actually no reference to groups are made ther |
| 09:18.42 | bip | so maybe if read some doc about that file |
| 09:19.00 | bip | usually docs are in /usr/share/doc |
| 09:19.15 | bip | but I m not very familiar with centos ... |
| 09:19.51 | bip | nothing there :( |
| 09:19.56 | kaldemar | asterisk has only chan_dahdi.conf. the rest are pieces that freepbx has hacked the single file into. |
| 09:20.30 | bip | ok so i must read the freepbx docs right ? |
| 09:20.37 | kaldemar | what does "asterisk -V" say? |
| 09:20.43 | bip | one sec |
| 09:21.06 | bip | [root@fenrir ~]# asterisk -V |
| 09:21.06 | bip | Asterisk 1.6.2.7 |
| 09:21.24 | bip | it s the one that comes with asteriskNOW |
| 09:21.38 | kaldemar | freepbx docs if you want to know why they divided the configs into multiple files, but for the options you can take a look at http://svn.digium.com/svn/asterisk/tags/1.6.2.7/configs/chan_dahdi.conf.sample |
| 09:21.40 | bip | i m using that because it s supposed to be easier |
| 09:22.12 | bip | thanks kaldemar you are very nice and helpful as usual :) |
| 09:22.20 | bip | i wil go and have a good read there |
| 09:23.04 | kaldemar | well, it is easier if everything happens to work. if you want to learn how asterisk works and get better support on this channel, ditch the GUI and start using asterisk by hand. you'll gain a huge amount of flexibility and features, but there's a bunch of stuff to learn. |
| 09:23.31 | bip | if iit was my network I would do so kaldemar |
| 09:23.51 | bip | Ãbut they want a gui they can use when I m not around |
| 09:24.05 | bip | I m sure i would be an happire camper w/out a gui |
| 09:24.17 | kaldemar | there's also #asterisknow btw. don't know about the activity on the channel though. |
| 09:24.39 | bip | looks almost dead last time i went there |
| 09:25.17 | kaldemar | hand-made changes are overwritten by the GUI if done in the wrong place, be aware. |
| 09:25.18 | bip | but i m pesronally confortable w/ou a gui I have bind apache and postfix running |
| 09:25.30 | bip | and never used a gui to run em |
| 09:25.43 | bip | but end user are not supposed to access em .. |
| 09:26.10 | bip | I m painfully aware kaldemar :( |
| 09:33.47 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 09:40.12 | *** join/#asterisk ranjan (~ranjan@122.172.124.210) |
| 09:41.53 | ranjan | Hi all. |
| 09:42.10 | ranjan | how to integrate asterisk with GSM Network? |
| 09:42.44 | ranjan | Any body here tried our Celliax or GSMOpen? |
| 09:43.53 | WIMPy | In what way? |
| 09:44.38 | ranjan | WIMPy, i mean i read that GSMOpen can be used to connect mobile phones with asterisk |
| 09:44.55 | ranjan | for both SMS and Voice calls. |
| 09:44.58 | *** join/#asterisk cusco (~trilili@a79-168-179-250.cpe.netcabo.pt) |
| 09:45.08 | ranjan | WIMPy, have you tried that? |
| 09:46.01 | WIMPy | There are may ways. |
| 09:46.17 | ranjan | WIMPy, can you please guide me with that? |
| 09:46.43 | WIMPy | GSM-SIP gateways, GSM-ISDN gateways, GSM PCI cards, chan_mobile and chan_datacard. |
| 09:47.19 | ranjan | how are these things used? |
| 09:47.27 | ranjan | any documents related to this? |
| 09:47.54 | ranjan | this chan_mobile and chan_datacard are the inbuilt with asterisk? |
| 09:48.05 | WIMPy | Via SIP, BRI or PRI, PCI, Bluetoots or USB. In that order. |
| 09:48.43 | WIMPy | Not sure about cahn_mobile, but chan_datacard is not included. |
| 09:48.44 | ranjan | Just now i read that mobile phones can be connected using USB to integrate with asterisk |
| 09:49.17 | ranjan | for both messaging and voice calls, have you tried that? |
| 09:49.31 | bip | when I call inbound I get this: http://pastebin.com/sXKxMzkW |
| 09:49.46 | WIMPy | I don't have a thone that could be used for that. |
| 09:49.48 | ranjan | i have a Nokia Xpress Music , can that be used to integrate with asterisk? |
| 09:50.17 | WIMPy | I wanted to try chan_datacard, but the voice feature is disabled on my stick. |
| 09:50.58 | ranjan | the datacard you are saying is the internet device isnt it? |
| 09:51.20 | WIMPy | USB stick |
| 09:51.40 | atan | Anyone know of a chart that shows the bandwidth required for sip video codecs? |
| 09:51.42 | ranjan | ok |
| 09:52.02 | atan | I'm wondering is phones supporting h.346 will be suitable for home connections =\ |
| 09:52.04 | kaldemar | bip: that's a freepbx issue. |
| 09:52.25 | ranjan | WIMPy, so according to you, for me to integrate my mobile phone with asterisk for atleast sending messages what all things will be required? |
| 09:52.33 | bip | ok I told em kaldemar |
| 09:53.43 | spenguin[work] | <PROTECTED> |
| 09:53.48 | spenguin[work] | url? |
| 09:54.02 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 09:54.02 | WIMPy | ranjan: try to look for some documentation on chan_mobile. |
| 09:54.15 | ranjan | WIMPy, ok let me see |
| 09:54.47 | kaldemar | ranjan: http://www.voip-info.org/wiki/view/chan_mobile |
| 09:54.54 | ranjan | spenguin[work], chrome://downloads/home/ranjan/Downloads/Maruzzelli_Celliax_Interfacing_GSM_With_FreeSWITCH.ppt |
| 09:54.58 | kaldemar | echoes... |
| 09:55.08 | ranjan | kaldemar, let me see |
| 09:55.47 | spenguin[work] | ranjan: ah neat, but thats not a url I can access :p |
| 09:55.48 | atan | "there is no mechanism within SIP for lip synchronization" does this mean video over SIP might have sound/picture show up at different times? |
| 09:55.53 | spenguin[work] | anyawy will google the ppt |
| 09:56.29 | ranjan | spenguin[work], that ppt has a picture that shows that. |
| 09:56.42 | ranjan | so that we could easily understand that. |
| 09:57.01 | schmidts | atan there is no need for this, cause video and audio is transmitted in one stream not in two separete |
| 09:57.14 | atan | schmidts, thank you =) |
| 10:01.07 | *** join/#asterisk tyman (~tyler@99-28-157-10.lightspeed.frsnca.sbcglobal.net) |
| 10:01.16 | tyman | is there a way to redirect or tee output to a distinct file from the asterisk cli? |
| 10:01.24 | spenguin[work] | ok |
| 10:01.24 | spenguin[work] | thanks ranjan |
| 10:02.08 | ranjan | spenguin[work], :) please will you join me in figuring out those things as i think you are also interested in that |
| 10:02.09 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
| 10:02.48 | spenguin[work] | ranjan: ah, well I would be its for my home asterisk setup |
| 10:03.06 | spenguin[work] | but not aytime soon since I have some other projects in mind :) |
| 10:03.20 | spenguin[work] | and I need to get a phone that hooks over usb |
| 10:03.25 | ranjan | spenguin[work], i am trying out for my company. but little by little |
| 10:03.41 | ranjan | spenguin[work], you can do it via bluetooth too. |
| 10:03.47 | ranjan | spenguin[work], here is the link for that |
| 10:03.55 | ranjan | http://www.voipphreak.ca/2008/10/30/installing-and-configuring-chan_mobile-for-bluetooth-presence-support-in-asterisk-16/ |
| 10:04.14 | spenguin[work] | ok |
| 10:07.24 | *** join/#asterisk ickmund (~ickmund@cli-5b7e85e4.bcn.adamo.es) |
| 10:08.24 | *** join/#asterisk pokui (~pjo@mbp-pjo.imul.com) |
| 10:08.50 | *** part/#asterisk pokui (~pjo@mbp-pjo.imul.com) |
| 10:20.10 | *** join/#asterisk guilhermebr (~Guilherme@200.103.96.98) |
| 10:24.40 | tyman | What is the best way to determine a phone ftp boot server when only having access to a linux shell from a remote location? I thought trying to get the info from a phone by first getting the phone addr from asterisk 'sip show peer <ext>' then using elinks (cli http client) to look at the polycom phone live config. Phones refuse http connections. Any ideas? |
| 10:25.44 | kaldemar | tyman: see dhcp server config if it passes an option to the phones. |
| 10:26.21 | kaldemar | tyman: for getting asterisk CLI output to a file, use script. |
| 10:26.30 | tyman | kaldemar: don't know have access (tonight) to the dhcp server, else that would be logical choice |
| 10:28.21 | tyman | kaldemar: cutting over phones tomorrow at new location...trying to read my ass off on configuring polycom phones for first time. They have many flavors of older phones and newer ones. Tomorrow we'll deploying 21 new pcom 650s. Scrambing to prepare... |
| 10:28.54 | kaldemar | if you know the network it must be in, you could scan for ftp servers and get lucky. |
| 10:29.00 | atan | Hmmm... what's the range on one of these cheapo GSM gateways? |
| 10:29.23 | tyman | just want to pull the existing configs...if i can find the ftp server without port scanning the whole enterprise for ftp, then i could pull down the configs |
| 10:30.36 | tyman | kaldemar: lol, yeah...didn't want to port scan their broken network....they use global addrs on the inside of their network all over the place too...totally broken |
| 10:32.03 | tyman | have some crappy pptp vpn account without split tunneling (no internet access while connected) and I have to jump thru one linux client to reach the two linux servers (which are not the dhcp/ftp servers) |
| 10:39.12 | *** join/#asterisk hurdman (~ngeek@arrakis.antredugeek.fr) |
| 10:39.16 | hurdman | hi |
| 10:39.34 | hurdman | i have got that error msg : IAX2/iaxvms01-3074 (type = 6, subclass = 11, ts=760091, seqno=123) |
| 10:39.45 | spenguin[work] | tyman: having a screen session on the server would help |
| 10:39.56 | hurdman | do i understand sublcass = 0x11 or 0x0b ? |
| 10:40.10 | hurdman | subclass* |
| 10:45.25 | WIMPy | tyman: Maybe bootptest can tell you something. |
| 10:45.32 | *** join/#asterisk visik7 (~Adium@unaffiliated/visik7) |
| 10:51.11 | tyman | WIMPy: i need to google that to understand how it will help me |
| 10:52.27 | tyman | WIMPy: don't know the server ip to send to |
| 10:53.49 | tyman | spenguin[work]: my box is inside a fw, would have to open a port for us to screen I believe |
| 10:55.22 | tyman | let's try another angle, is there any "up to date" template configs that i could study and diff against the polycom defaults? |
| 11:00.18 | *** join/#asterisk PoTe (~PoTe@rev-200-40-119-222.netgate.com.uy) |
| 11:01.47 | kaldemar | well the defaults are templates with all available settings. |
| 11:03.47 | tyman | kaldemar: yes, i'm just looking for the typical user modified settings...i realize that's subjective...i normally try to figure this out on my own...just under extreme pressure and I'm googling all sorts of dated and contradictory info |
| 11:10.32 | *** join/#asterisk marl_scot (~matt.lowe@office.unk.com) |
| 11:12.57 | marl_scot | hi there, i know i came accross this a while ago, but now i cant find it :( i have asterisk 1.4 + mISDN +LCR, and my DTMF tones are VERY short, how can i increase the length? |
| 11:18.21 | *** join/#asterisk timahvo1 (~rogue@41.72.215.94) |
| 11:23.02 | *** join/#asterisk timahvo1 (~rogue@41.72.215.94) |
| 11:37.53 | *** join/#asterisk |Physis| (~|Physis|@187.60.42.210) |
| 11:46.14 | defswork | marl_scot, I get emails all day about increasing the length |
| 11:49.08 | marl_scot | lol, dont suppose you have done a howto have you? |
| 11:51.50 | hehol | http://support.mozilla.com/de/kb/Using+the+Windows+Media+Player+plugin+with+Firefox |
| 11:52.05 | hehol | Sorry, wrong window :-( |
| 11:58.25 | *** join/#asterisk timahvo1 (~rogue@41.72.215.94) |
| 12:00.45 | creativx | aggh bungo |
| 12:01.18 | creativx | is it xlite that causes my inbound audio to be choppy when I talk at the same time |
| 12:05.50 | creativx | or is it asteriskxkxkx |
| 12:06.45 | fauxalliance | creativx, probably the network environment |
| 12:06.55 | creativx | 1gbit lan |
| 12:06.59 | creativx | 1 hop |
| 12:07.10 | fauxalliance | creativx, VoIP likes packets on time, in order |
| 12:07.45 | fauxalliance | creativx, could potentially be a server load issue, codec issue or the aforementioned network issues.... |
| 12:07.50 | creativx | hmm |
| 12:07.52 | creativx | ok |
| 12:08.05 | creativx | it is reproducible |
| 12:08.39 | creativx | meh, i gotta find me someone who can debug that for me |
| 12:08.59 | *** join/#asterisk file (~file@asterisk/developer-and-muffin-lover/file) |
| 12:08.59 | *** mode/#asterisk [+o file] by ChanServ |
| 12:09.22 | fauxalliance | creativx, between you and wireshark... take a look at the wire... perhaps some VLAN tagging or a seperate subnet for VoIP is in order... |
| 12:09.47 | creativx | on a low activity gbit network, should that really matter? |
| 12:10.09 | creativx | i mean would even 2 mbit traffic across that gbit network affect the voip traffic at all |
| 12:10.26 | fauxalliance | creativx, for what it's worth... I had a old IBM (ahem) HUB.... 10baseT.... everyone sounds like a robot on this particular network device.... (under load) |
| 12:10.41 | creativx | hehe, HUB .. |
| 12:10.53 | fauxalliance | <fauxalliance> creativx, VoIP likes packets on time, in order < yea, 2 MB can plug stuff up... look into VLAN tagging |
| 12:11.46 | fauxalliance | I.E.... clear off all the traffic, and it's related overhead off the network link... then can you reproduce it? |
| 12:11.53 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 12:12.37 | creativx | it might be xlite tagging the traffic |
| 12:12.45 | creativx | im gonna do some experimenting |
| 12:13.08 | fauxalliance | creativx, lemme know how it makes out |
| 12:31.22 | *** join/#asterisk Hanumaan (~Hanumaan@dslb-088-066-128-080.pools.arcor-ip.net) |
| 12:34.35 | *** join/#asterisk michael-i (~michael-i@141.41.40.223) |
| 12:35.48 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 12:35.53 | michael-i | anyone have experience setting different echo cancellation parameters for oslec? I have runaway CPU load using it and setting "echocancel=128,param1=32,param2=0,param3=14" instead of just "echocancel=128" solves thing. No idea why though |
| 12:37.09 | michael-i | Just tried the example in the sample confs as a starting point. Now I need to roll this fix out but won't until I know exactly what I've changed with those params |
| 12:40.11 | marl_scot | defswork, dont suppose you would share your extending dtmf tones knowledge? |
| 12:50.51 | *** join/#asterisk [netman] (~netman@248.Red-79-152-117.dynamicIP.rima-tde.net) |
| 12:52.45 | *** join/#asterisk joobie (~joobie@CPE-124-180-108-138.lns10.lon.bigpond.net.au) |
| 12:53.30 | tzafrir_laptop | michael-i, AFAIK OSLEC ignores all the other params. If it doesn't: it's a bug |
| 12:53.35 | tzafrir_laptop | Is it reproducable? |
| 12:54.26 | michael-i | tzafrir_laptop: After looking at the dahdi oslec interface, it seems my problem was solved in that the module was not loaded because the parameters make it bail out! |
| 12:54.38 | michael-i | not a bug fix then :) |
| 12:55.32 | michael-i | echo cancellation on bri isdn lines using an openvox minipci card is killing the cpu....everywhere else it seems to be ok. fun friday fun |
| 12:55.43 | JerJer | man - i'm getting way to damn old to be pulling all nighters |
| 12:55.52 | russellb | well there's your problem, why you using that chinese clone crap :-) |
| 12:56.14 | russellb | and don't give me that "but it's mini PCI" excuse :-p |
| 12:56.15 | michael-i | I'd gladly use someone else's minipci bri 4-port isdn...oh wait |
| 12:56.18 | joobie | JerJer, why the all nighter? |
| 12:56.19 | michael-i | ;) |
| 12:56.20 | michael-i | crap |
| 12:56.22 | russellb | lol, i beat you |
| 12:56.39 | michael-i | *expletive* |
| 12:56.41 | russellb | i have nothing useful to offer, so i just trolled. |
| 12:56.45 | joobie | guys anyone in the US or UK and want to trunk to AU for free? |
| 12:56.47 | fauxalliance | HA! |
| 12:57.07 | *** join/#asterisk Dovid (~Dovid@173.220.127.18) |
| 12:57.12 | russellb | free? sounds like a trap |
| 12:57.19 | joobie | well |
| 12:57.26 | joobie | i gotta pass calls to u too |
| 12:57.26 | *** join/#asterisk atan2 (8eb15111@gateway/web/freenode/ip.142.177.81.17) |
| 12:57.28 | joobie | for free :P |
| 12:57.31 | michael-i | I think I'm just going to build in some more echo cancellers into my next firmware so people have a workaround |
| 12:57.56 | JerJer | joobie: was sorting out AMI details for an acd app we have been writing |
| 12:58.31 | *** join/#asterisk ickmund (~ickmund@cli-5b7e85e4.bcn.adamo.es) |
| 12:58.44 | JerJer | we need to use chan_local to set variables and what not, but we don't want them hanging around |
| 12:59.30 | JerJer | so i wrote a test to see if there was any audio blips if we redirected the result of the local chan (a dahdi channel) back into the meetme situation |
| 12:59.31 | michael-i | any consensus on the best !oslec module? |
| 12:59.47 | anonymouz666 | best oslec module? |
| 12:59.55 | michael-i | best module that is not oslec |
| 13:00.20 | anonymouz666 | best module is the one that works for you |
| 13:00.27 | anonymouz666 | in my case, oslec works |
| 13:01.15 | russellb | michael-i: whichever one is included with DAHDI and used by default I guess? |
| 13:01.17 | russellb | MG1 is it? |
| 13:01.25 | russellb | i may have made that up. |
| 13:01.40 | anonymouz666 | FYI... this list is very nice: http://lists.digium.com/mailman/listinfo/asterisk-ha-clustering |
| 13:01.54 | russellb | very low traffic, though .. |
| 13:02.03 | michael-i | russellb: sounds logical...I'm buried in my custom dahdi makefiles, let me take a peek at what the digium diets say |
| 13:02.16 | russellb | good luck to ya. |
| 13:02.28 | anonymouz666 | russellb: :( |
| 13:03.00 | russellb | but people are on there, might start a good discussion if you ask a good question :-) |
| 13:03.42 | JerJer | anonymouz666: i think everyone's idea of HA with asterisk is going to be different |
| 13:04.09 | JerJer | i have been doing serious volumes with asterisk for years (without SCF) |
| 13:05.25 | JerJer | i avoid the lists, personally |
| 13:06.16 | JerJer | ok - bedtime |
| 13:06.26 | anonymouz666 | I like the lists, at some points, the demand is the same. you got some great ideas from there - mainly from people that already does what you are planning to do |
| 13:06.49 | JerJer | i might eyeball the archives :) |
| 13:06.57 | anonymouz666 | how could be bedtime if it is 11:06 am |
| 13:07.25 | JerJer | been beating up asterisk boxes for over 30 hours now |
| 13:07.44 | anonymouz666 | wow |
| 13:08.14 | anonymouz666 | that's too much |
| 13:08.28 | JerJer | ami is fun, isn't it :) |
| 13:08.37 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-163.hsd1.nj.comcast.net) |
| 13:08.38 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 13:08.47 | *** join/#asterisk goddva_ (~glarsen@77.40.154.242) |
| 13:09.09 | anonymouz666 | AMI is a piece that I personally don't know much. |
| 13:09.19 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 13:09.34 | JerJer | i hate it and require it at the same time :( |
| 13:09.51 | anonymouz666 | yeap, there are people here that works specifically with AMI and just AMI. |
| 13:10.15 | JerJer | i hate app_queue even more though, so i'll deal with AMI |
| 13:11.09 | anonymouz666 | hehe |
| 13:11.54 | JerJer | at the volume the queues this project deals with app_queue chokes |
| 13:12.14 | JerJer | we are doing all of the 'queuing' via AMI interaction |
| 13:12.36 | *** join/#asterisk krion (~seb@unaffiliated/krion) |
| 13:12.44 | anonymouz666 | how big is your volume? |
| 13:12.52 | JerJer | one queue could be 'active' in perhaps a dozen asterisk instances :) |
| 13:14.11 | JerJer | in reality asterisk is simply a media gateway, in this situation |
| 13:14.24 | *** join/#asterisk neurosys (~neurosys@173-9-159-177-miami.txt.hfc.comcastbusiness.net) |
| 13:14.30 | JerJer | we have Perl based application servers that 'do' the queue |
| 13:15.09 | JerJer | must sleep (will rap later on) |
| 13:15.22 | anonymouz666 | ok JerJer, good night :P |
| 13:26.44 | *** join/#asterisk netrat (~netrat@2001:470:8:479::3) |
| 13:28.43 | *** join/#asterisk wesphillips (~wphillips@64-19-240-7.caprock.net) |
| 13:32.16 | netrat | i'm having trouble setting up an IAX2 trunk between two asterisk servers on the same the LAN. one server is running 1.6.2.14 and the other is 1.8.0. The phone extensions over the IAX2 trunk will ring, but voice is very choppy and the call will eventually disconnect. I've posted a few config snippits here http://pastebin.com/fv1xdJFq |
| 13:32.33 | netrat | i enabled internal_timing in the asterisk.conf file, but that didn't seem to help |
| 13:32.54 | *** join/#asterisk pabelanger (~pabelange@2607:f2c0:a000:166:218:f3ff:fe51:c71) |
| 13:32.54 | *** mode/#asterisk [+o pabelanger] by ChanServ |
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| 13:36.48 | *** mode/#asterisk [+o leif[mobile]] by ChanServ |
| 13:46.25 | *** join/#asterisk notax (~notax@128-145-16-190.fibertel.com.ar) |
| 13:46.32 | notax | hello |
| 13:47.22 | netrat | hello |
| 13:47.38 | atan2 | Hi? |
| 13:51.13 | *** join/#asterisk BMJ (~bjohns@c-24-126-158-110.hsd1.ga.comcast.net) |
| 13:51.13 | *** mode/#asterisk [+o BMJ] by ChanServ |
| 13:52.48 | *** join/#asterisk aquarat (~cowalski@mail.aquarat.za.net) |
| 13:53.07 | aquarat | Hi, I've got two Blackfin based Asterisk installations and I'm trying to link them using IAX2. One box has a static IP, the other dynamic. I set up an extension in the asterisk gui on the static box which the dynamic box connects to. The dynamic box routes calls out via the static box... but attempts to dial the dynamic box's extension results in a call drop |
| 13:53.14 | aquarat | core set verbose 9 and iax2 debug say no context found |
| 13:53.21 | aquarat | sorry about the flood |
| 13:53.23 | aquarat | :P |
| 13:53.33 | jbroome | it wasn't a flood until the last two lines |
| 13:53.49 | atan2 | snickers |
| 13:54.15 | aquarat | lol |
| 13:54.52 | aquarat | the weirdest part of this problem was it worked the first time I tried it |
| 13:55.12 | aquarat | and then I stupidly reflashed the firmware, haven't been able to get it working again |
| 13:57.16 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 13:59.47 | atan2 | Welcome back [TK]D-Fender =) |
| 14:01.38 | fauxalliance | indeed! |
| 14:05.52 | *** join/#asterisk Akiraa (~Akira@92.81.167.100) |
| 14:06.46 | *** join/#asterisk grandpapadot (~nonya@99.175.248.81) |
| 14:08.29 | grandpapadot | Hey guys, is there a way to handle a missing prompt played with BackGround() in 1.4? I see the PlayBack() has a ${PLAYBACKSTATUS} variable but don't see anything similar with BackGround() ... any suggestions appreciated! |
| 14:09.49 | *** join/#asterisk i_heart_asterisk (~chatzilla@firewall-a.buf.ny.i-evolve.net) |
| 14:11.25 | *** join/#asterisk Linuturk (~linuturk@unaffiliated/linuturk) |
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| 14:19.32 | Katty | goooooooooooooooooooooood morning!!!!!!! |
| 14:19.39 | jaytee | morning Katty |
| 14:19.41 | fauxalliance | whips out a hug for Katty |
| 14:20.02 | Katty | if anyone is interested in participating in the Asterisk Christmas Card Exchange, please /query me for details. Cut off time is around December 10th or 15th depending on which country you're shipping to. |
| 14:20.04 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 14:20.07 | Katty | hugs fauxalliance |
| 14:20.08 | Katty | hugs jaytee |
| 14:20.18 | jaytee | hugs Katty |
| 14:20.19 | fauxalliance | soooooo.... you make a sarcastic comment and you end up in a forced labour camp... not just China under totalitarian regime. http://articles.cnn.com/2010-11-18/world/china.tweet.punishment_1_labor-camp-twitter-japanese-products?_s=PM:WORLD |
| 14:20.35 | Katty | also! |
| 14:20.40 | Katty | crittercam is extra festive this morning |
| 14:20.42 | Katty | infobot: crittercam |
| 14:20.43 | infobot | crittercam is probably Katty's live broadcast of The Nut House at http://www.ustream.tv/channel-popup/squirrel-critter-cam |
| 14:23.05 | neurosys | can anyone explain why * has a different time than the system time? IE: in the voicemail? |
| 14:26.50 | jaytee | there is a timezone setting tz=<timezone> in the voicemail.conf file. As I understand it that can be set per user or groups of users. |
| 14:30.23 | neurosys | jaytee: thx! :) |
| 14:31.07 | *** join/#asterisk eject_ck (~eject_ck@haitek.cosmonova.net.ua) |
| 14:31.15 | tzanger | pfft. |
| 14:31.22 | tzanger | there *was* a bird there until y'all scared it away |
| 14:31.32 | jaytee | I'm looking at SIP trunking prices for the major US ITSPs listed and trying to get a feel for who has the best performance/pricing combination for package of say 8 SIP channels. Can anyone recommend a provider based on their own experience? |
| 14:34.46 | fauxalliance | link2voip |
| 14:34.49 | fauxalliance | ~itsp |
| 14:34.49 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 14:34.59 | fauxalliance | ^^ consult the list jaytee |
| 14:35.25 | Katty | i'll consult YOUR list in a minute. |
| 14:36.41 | fauxalliance | uh oh... /me is probably on the naughty one |
| 14:36.55 | jaytee | fauxalliance, I'm using all the info provided by itsplist-us already. I'm just not sure which of them would offer the best service for the price and some of them have very vague pricing descriptions on their websites. |
| 14:37.26 | fauxalliance | jaytee, from experience... i stick with link2voip... reasonable... excellent support |
| 14:37.46 | fauxalliance | if you should order DID's... please put my name in the coupon code |
| 14:38.48 | i_heart_asterisk | i'm trying to play a file right after a voicemail message plays....im looking in the play_message function in app_voicemail.c, i can't seem to find the line where the actual voicemail message gets played, can someone tell me the line of code i'm looking for ? |
| 14:39.02 | jaytee | fauxalliance, they aren't listed in the itsplist-us |
| 14:39.22 | fauxalliance | jaytee, they are on MY LIST |
| 14:39.33 | jaytee | ~itsplist-ca |
| 14:39.33 | infobot | Here are some popular ITSPs (Canada) starting with the more respected ones : http://www.unlimitel.ca , http://www.les.net , http://www.babytel.ca , http://www.voip.ms |
| 14:40.11 | jaytee | fauxalliance, and where exactly did you get YOUR LIST? :-) |
| 14:40.12 | *** join/#asterisk hay (~hay@tm.84.52.129.113.dc.cable.static.telemach.net) |
| 14:40.24 | jaytee | hey hay! |
| 14:40.29 | hay | hey :D |
| 14:40.43 | fauxalliance | bites his lip and seeks something blunt |
| 14:40.45 | fauxalliance | https://www.link2voip.com/rates.php |
| 14:41.45 | fauxalliance | based on my own experience jaytee ... you get what you ask for... |
| 14:43.45 | *** join/#asterisk neurosys (~neurosys@173-9-159-177-miami.txt.hfc.comcastbusiness.net) |
| 14:43.48 | tzanger | I've used both unlimitel and voip.ms. both work very well |
| 14:44.35 | hay | we have been using asterisk for quite some time and today I have found out an issue that happens when some user that has ISDN calls our number (no ISDN equipment at our place)... We can call them without any issues, but when they call and the phone rings, immediately after I pick up a phone I get "busy" tone... in logs I see: macro-hangupcall, macro-dial and macro-exten-vm all exited non-zero |
| 14:44.36 | hay | and that's all I can find out in logs... how could I find more information? We receive a number of incomming calls and they all (except those tho cases that are using ISDN) work correctly on this extension... I just need to debug this - does anyone have such an experience? thx |
| 14:44.44 | *** join/#asterisk hrhrhr (~c1@unaffiliated/hrhrhr) |
| 14:45.49 | jaytee | wonder if they work in HR? |
| 14:46.09 | tzanger | heh |
| 14:46.15 | fauxalliance | hay, wow... thats a conundrum... can you pastebin the CLI output of the call? |
| 14:46.28 | tzanger | someone contacted me via twitter regarding the XHFC driver I wrote a year or two ago |
| 14:46.34 | *** join/#asterisk [Outcast] (~anonymous@64.202.62.5) |
| 14:47.05 | tzanger | hay: sounds like a problem with your telco, not your setup |
| 14:48.25 | hay | fauxalliance, here is the pastebin log: http://pastebin.com/mRjJ2C5R |
| 14:48.46 | hay | tzanger, I see... If I won't be able to solve the issue I am going to contact them |
| 14:52.26 | i_heart_asterisk | can someone tell me what line of code in play_message function in app_voicemail.c actually plays the voicemail message ? |
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| 15:38.17 | jaytee | can a US carrier actually refuse to release a DID number if I want to port that number to another US carrier? Vitelity's website indicates that may be a possibility. |
| 15:39.13 | frigidzephyr | i think it depends on your contract with the carrier |
| 15:39.25 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 15:39.27 | jaytee | we're currently using Vonage |
| 15:40.18 | frigidzephyr | no idea. read through the sign-on agreement and contract to see if it mentions number portability |
| 15:41.17 | jaytee | frigidzephyr, thanks |
| 15:42.58 | frigidzephyr | jaytee: http://www.fcc.gov/cgb/consumerfacts/numbport.html |
| 15:43.19 | frigidzephyr | looks like with FCC rules, if you are remaining in the same geographic area, they are required to port your number |
| 15:43.20 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 15:43.33 | frigidzephyr | as long as you don't terminate service with old provider before requesting the move with the new provider |
| 15:43.59 | frigidzephyr | only for local numbers though |
| 15:44.32 | frigidzephyr | i don't see that it applies to toll numbers or numbers you get that are out of your area, etc |
| 15:53.25 | *** join/#asterisk bbeers (~bbeers@static-96-254-222-162.tampfl.fios.verizon.net) |
| 15:54.25 | _Corey_ | jaytee: The big thing to keep in mind is your paperwork must be very accurate to ensure a smooth port |
| 15:54.35 | *** join/#asterisk crowbar7 (~JKLOBUCAR@12.2.84.2) |
| 15:54.44 | _Corey_ | jaytee: Billing telephone number must be right, billing address, contact name, etc |
| 15:55.48 | _Corey_ | jaytee: A good percentage of # ports are rejected on the first attempt |
| 15:58.53 | bbeers | hello, I have a session-timers re-invite issue. Using * 1.8, my dialplan adds SIP headers before dialing, but session-timer activated re-invites do not include the added headers, so other side rejects with 420 Bad Extension. |
| 15:59.29 | goddva | if I get locks in "core show locks" - does this mean I have a problem? |
| 16:02.04 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 16:02.56 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 16:03.56 | Katty | OHAITHAR |
| 16:05.25 | drift- | how can i create a menu for when people call in |
| 16:05.31 | drift- | for like 1. for sales 2 for tech support |
| 16:05.59 | Katty | waitexten, playback, and exten. |
| 16:06.06 | Katty | oh. i guess that'd be background() |
| 16:06.08 | Katty | not playback |
| 16:06.20 | drift- | any tutorials on this some where? |
| 16:06.26 | Katty | it's in the book i'm sure. |
| 16:06.37 | frigidzephyr | in the o'reilly book, and about a million tutorials if you google |
| 16:06.37 | Katty | infobot: thebook |
| 16:06.37 | infobot | extra, extra, read all about it, thebook is Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://astbook.asteriskdocs.org, or http://ofps.oreilly.com |
| 16:10.45 | *** join/#asterisk came0 (~nick@li181-40.members.linode.com) |
| 16:11.16 | p3nguin_ | BackGround() and WaitExten() both listen for DTMF. |
| 16:11.20 | came0 | where do i find cfg files for my polycom phones to download from the ftp server? |
| 16:11.45 | came0 | i see examples here http://www.krisk.org/asterisk/pcom/ |
| 16:12.00 | came0 | but they are noted as from 2004... |
| 16:12.23 | *** join/#asterisk Defraz (~Defraz@63.226.95.147) |
| 16:18.01 | jaytee | anyone here ever use vitelity.net as an ITSP? |
| 16:19.45 | p3nguin_ | I use VoIP.ms, which uses Vitelity... does that count? |
| 16:20.12 | jaytee | I guess |
| 16:20.31 | jaytee | p3nguin_, do you like voip.ms? |
| 16:20.35 | p3nguin_ | Yes. |
| 16:21.43 | jaytee | are you using SIP or IAX2 to connect to them? |
| 16:22.04 | p3nguin_ | Currently IAX2. I've used SIP in the past, though. |
| 16:22.19 | jaytee | solid? any problems? |
| 16:22.44 | c0rnoTa | Hello all |
| 16:23.13 | c0rnoTa | may i use cable for floppy like timing cable for digium cards? |
| 16:23.15 | p3nguin_ | I don't think I've had any problems. |
| 16:24.03 | chuckf | jaytee: I have used Vitelity at home for a few years now, no issues that I've noticed with the sip connections |
| 16:24.22 | jbroome | ditto |
| 16:24.36 | *** join/#asterisk tyman (~tyler@mail.cityofmadera.com) |
| 16:27.28 | *** join/#asterisk ickmund (~ickmund@cli-5b7ee407.bcn.adamo.es) |
| 16:27.41 | jaytee | I'm shopping around for an ITSP to replace Vonage for our company. We're a small consulting firm (10 employees) and currently have capability for 4 concurrent calls. Would like to have capability for 8-10 concurrent calls. |
| 16:28.56 | *** join/#asterisk Tim_Toady (~moi@178.128.21.207) |
| 16:29.36 | *** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano) |
| 16:30.00 | chuckf | a friend of mine runs an office about that size and is the one that pointed me to Vitelity based on his experience |
| 16:30.44 | chuckf | and I think he has more concurrent calls than that, but I don't know for sure |
| 16:31.04 | jaytee | chuckf, thanks for the feedback. Based on pricing voip.ms and vitelity seem to be the most affordable |
| 16:31.14 | drmessano | What about Flowroute |
| 16:31.23 | jaytee | haven't looked at them yet |
| 16:31.27 | chuckf | I've never used them |
| 16:31.30 | drmessano | Their rates are the best I have seen |
| 16:31.35 | drmessano | Service has been awesome |
| 16:31.41 | jaytee | then why are they at the end of the list? |
| 16:32.05 | drmessano | Because that list order is stupid |
| 16:32.31 | drmessano | Vitelity sucks, IMHO.. but they still get Kudos from sheep and newbs because of the specials they run |
| 16:33.01 | jbroome | err |
| 16:33.02 | drmessano | Everyone complains about Vitelity's service, yet you ask them a week later about Vitelity and they are "the best" |
| 16:33.34 | jbroome | This *everyone* has had no problems with service in the 3-4 yrs i've had them |
| 16:34.14 | drmessano | Not sure how that's possible.. because I know for a fact they went through a period, about a year ago, that EVERY Vitelity user had some problem with the reliability.. |
| 16:34.22 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com) |
| 16:34.37 | drmessano | Since then it's not been DAILY, but I still hear complaints |
| 16:35.02 | *** join/#asterisk JonnyD_work (~Jon@12.222.63.34) |
| 16:35.25 | drmessano | voip.ms is nothing more than a vitelity reseller, and they were taking the brunt of it too |
| 16:37.14 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
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| 16:40.01 | JonnyD_work | drmessano: what was that? was their an attack on vitelity yesterdya? |
| 16:40.39 | drmessano | Why, was there a problem? |
| 16:41.11 | JonnyD_work | well i was not able to dial out thorugh one of their outbound gateways |
| 16:41.44 | drmessano | ^^^^^^ |
| 16:42.01 | drmessano | I hear that crap all the time about Vitelity |
| 16:42.26 | JonnyD_work | yeah i always have problems with them i can't stand their service it sucks |
| 16:42.27 | drmessano | But, anyway.. I am just an observer.. I suggest doing some real research. Flowroute has been consistent |
| 16:42.46 | drmessano | JonnyD_work, others would argue it's reliable as hell.. Do you disagree? |
| 16:42.59 | JonnyD_work | i don't use them i just have a bunch of clients i setup on their services |
| 16:43.04 | drmessano | Ah |
| 16:43.33 | JonnyD_work | well i dont' get nearly as many complaints |
| 16:43.55 | JonnyD_work | i used to use a different providor and they were much worse |
| 16:45.56 | JonnyD_work | does anyone have a good tool to do a brute force attack like a registration storm? i want to test if my fail2ban setup is correct |
| 16:46.13 | JonnyD_work | i was trying to find sivus but their website is goine |
| 16:46.16 | JonnyD_work | gone* |
| 16:46.17 | drmessano | ~sipp |
| 16:46.17 | infobot | i guess sipp is a free Open Source test tool / traffic generator for the SIP protocol available from http://sipp.sourceforge.net/. If you really want to know how many channels your Asterisk box can do, learn how to utilize this program. |
| 16:47.09 | JonnyD_work | yeah i used that before for performance testing i was trying to do a registration storm can that do taht too? |
| 16:48.04 | jbroome | JonnyD_work: http://code.google.com/p/sipvicious/ |
| 16:49.53 | JonnyD_work | cool thanks |
| 16:57.13 | *** join/#asterisk emcho (~emcho@2a01:e35:8a55:abc0:21e:c2ff:fe1b:2fe) |
| 16:58.10 | *** part/#asterisk emcho (~emcho@2a01:e35:8a55:abc0:21e:c2ff:fe1b:2fe) |
| 17:00.39 | i_heart_asterisk | we are having a strange issue with asterisk and our lucent TNT....someone calls to leave voicemail, recording starts, the TNT hangs up after exactly 60 seconds from start of vm recording |
| 17:01.19 | i_heart_asterisk | does anybody have any advice on this ? |
| 17:04.36 | *** join/#asterisk timholum (~chatzilla@68-117-120-138.static.eucl.wi.charter.com) |
| 17:05.32 | notax | i_heart_asterisk, are you sure the TNT is hanging up the call? |
| 17:06.27 | *** join/#asterisk fifer (~fifer@67.208.108.228) |
| 17:07.26 | timholum | hello, I am working on a program that uses that uses AMI, when I do a Redirect on an inbound call it actuly just drops the call? is there something more that is suppose to be done |
| 17:07.45 | timholum | or is redirect not the proper way to transfer a call? |
| 17:08.05 | timholum | I am running asterisk 1.4 but will probably be upgrading to 1.8 soon |
| 17:09.18 | i_heart_asterisk | notax: yes, we hve other peers going to the same asterisk box and ths problem does not occur |
| 17:09.54 | i_heart_asterisk | notax: could it be something asterisk is sending to the TNT? if so, why does this problem not occur with other peers ? |
| 17:10.21 | notax | timholum, maybe you are not redirecting correctly? what does ami says? |
| 17:11.24 | notax | i_heart_asterisk, try with a call to an extension and see if it hangs up after 60 secs |
| 17:12.04 | i_heart_asterisk | notax: we've already tried that, it does not hangup after 60 seconds...this strangely occurs only during voicemail |
| 17:12.18 | notax | i_heart_asterisk, whats your maxsecs in voicemail.conf? |
| 17:12.19 | i_heart_asterisk | notax: i should say voicemail recording |
| 17:12.37 | timholum | AMI tells me "Redirect Successfull |
| 17:12.56 | i_heart_asterisk | notax: maxsecs is set to default |
| 17:12.57 | timholum | one sec while I look through the console loggs |
| 17:13.16 | notax | timholum, check if you are redirecting to the correct exten and priority |
| 17:13.36 | *** join/#asterisk jcapelo (~jcapelo@189.140.22.241) |
| 17:13.51 | jcapelo | hello everyone |
| 17:14.14 | timholum | notax: I see the lines |
| 17:14.16 | timholum | <PROTECTED> |
| 17:14.17 | timholum | <PROTECTED> |
| 17:14.29 | timholum | right befor it runs through the hangup script |
| 17:14.34 | jcapelo | I need help with patching calls I am currently receiving in softphone through a SIP address, so that I can receive those calls (from a web "CallMe" button) in the asterisk server |
| 17:15.18 | notax | timholum, i think you are not using redirect correctly, use pastebin to show me your ami command |
| 17:15.27 | jaytee | drmessano, just setup a trial account with flowroute.com and it was totally painless |
| 17:15.42 | timholum | I am using ajam http://192.168.1.51/ajam/mxml?action=Redirect&channel=IAX2/voipms-822&exten=200 |
| 17:16.11 | came0 | to setup the boot-server option in my dhcp server ive done: option boot-server code 66 = string; option boot-server "ftp://PlcmSpIp:PlcmSpIp@192.168.1.100"; in my dhcpd.conf. Is that right? Phones still say "Cant connect to boot server" |
| 17:16.30 | notax | timholum, i don't know ajam, try to read the AMI to see what command its sending |
| 17:16.53 | drmessano | jaytee: Awesome |
| 17:17.18 | notax | timholum, use telnet to do that |
| 17:17.18 | Nugget | telnet is eeeeeeevil! |
| 17:17.31 | came0 | who uses telnet still? |
| 17:17.36 | notax | i do |
| 17:17.54 | thehar | lots of people |
| 17:17.55 | timholum | I can give it a shot from that, one sec |
| 17:18.30 | came0 | why not just ssh right into the server and interact with the ami stuff there? |
| 17:20.20 | p3nguin | Telnet has less overhead. |
| 17:20.56 | came0 | also less secure. not saying its wrong i just havent used telnet in a long time. |
| 17:22.22 | came0 | actually thats not true i telnet to mail servers sometimes. i guess it does have its place. |
| 17:22.31 | notax | came0, telnet is the best way to simulate an external program |
| 17:22.44 | came0 | yeah, i wasnt thinking about that. nm |
| 17:23.09 | jaytee | i had to read back in the chat to see if Nugget still has the auto response for "telnet" turned on :-) |
| 17:25.43 | jaytee | came0, there's a section on setting up ftp for Polycom provisioning in the Asterisk book. At my old job we had Windows servers for DHCP and I only had to put the static IP address of the ftp server, not the account and password. |
| 17:25.43 | JonnyD_work | jbroome: wow that is such a cool tool, great way to hamer my own server |
| 17:25.58 | *** join/#asterisk andyoutside (6161c46a@gateway/web/freenode/ip.97.97.196.106) |
| 17:27.00 | timholum | notax: I just ran basicly the same command using telnet with the same effect |
| 17:27.12 | notax | !pastebin |
| 17:27.32 | notax | paste the command here: pastebin.ca |
| 17:28.27 | timholum | http://pastebin.com/TAKrftpp |
| 17:29.21 | timholum | line 228 is where i typed the command in |
| 17:29.55 | notax | timholum, you are missing priority and context |
| 17:30.36 | timholum | notax: ok, so add priority: 1 and context: default I am assuming |
| 17:31.07 | notax | depends on your extensions.conf, but try with that |
| 17:32.49 | timholum | thank you :) that worked :) |
| 17:32.54 | notax | great :) |
| 17:33.01 | *** join/#asterisk imox1234 (~imox1234@p4FC5C523.dip0.t-ipconnect.de) |
| 17:33.33 | timholum | now I can add the ability to transfer calls to my crm solution :) |
| 17:34.13 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 17:34.40 | i_heart_asterisk | will "reload" affect any calls ? |
| 17:34.47 | i_heart_asterisk | current calls ? |
| 17:35.11 | i_heart_asterisk | or will "sip reload" affect current calls? |
| 17:37.02 | russellb | no |
| 17:37.08 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 17:37.08 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 17:37.38 | notax | neither |
| 17:38.21 | *** join/#asterisk Ad-Hoc (~nimbus@62.1.114.209.dsl.dyn.forthnet.gr) |
| 17:38.27 | Scorcerer | Can i use the book "Asterisk: the future of telephony 2nd edition" to learn asterisk 1.6 ? |
| 17:38.38 | Scorcerer | or difference is too big ? |
| 17:39.22 | russellb | no, it's fine |
| 17:39.34 | russellb | we also have a 3rd edition in the works based on 1.8 |
| 17:39.43 | russellb | which you can read what's done so far at http://ofps.oreilly.com |
| 17:40.17 | Scorcerer | thanks |
| 17:40.17 | came0 | trying to get provisioning work for these polycom phones i get this entry in 0004f228fdfc-boot.log: "1119172641|copy |3|00|transport res: 78 respCode 550" followed by line includeing "transport error: Error." and "transport error buffer: Error." and "Update of '/ffs0/init.mac' failed, leaving local copy intact". so the phone connects to the ftp server fine but get this error when trying to download the |
| 17:40.19 | came0 | <PROTECTED> |
| 17:40.38 | jbroome | russellb: oh sweet! |
| 17:43.21 | Scorcerer | if it's for 1.8, is it beter to install 1.8 ? or 1.6? |
| 17:44.49 | Scorcerer | ah, 1.8 isn't ready for production, so for leaning it should be fine :> |
| 17:46.03 | *** part/#asterisk ThoMe (tm@tm.muc.de) |
| 17:49.35 | *** join/#asterisk jpvoip (~petrybr@189.73.117.170) |
| 17:49.58 | *** join/#asterisk jblack (~jblack@71.181.209.104) |
| 17:50.23 | jblack | I squashed Katty with a monster truck. I hope she won't be angry |
| 17:50.44 | came0 | oh geez i was missing a 0 in my 0004f :/ |
| 17:51.04 | jpvoip | Hello, im trying to install chan-sccp-b on my asterisk (1.4) running on my ubuntu 8.14... im receiving this error when i try to ./configure: Cannot find pbx libraries - these are required... Any sugessstion what i can do to fix this? |
| 17:51.39 | anonymouz666 | haha nice quote from chapter 13... "something" -- somebody |
| 17:52.25 | *** join/#asterisk luckman212 (~quassel@pool-96-246-172-198.nwrknj.fios.verizon.net) |
| 17:54.12 | luckman212 | guys, I am looking for a SIP client (softphone) for Windows that can register itself to the sip:// protocol (i.e. accepts command line parameters to dial a call) and also supports SIP URI (xyz@somewhere.com) dialing. X-Lite v3 used to do this but they removed it in v4. Blink doesn't seem to, nor does the 3CX softphone. Any suggestions? |
| 17:55.08 | drmessano | So use Xlite V3 |
| 17:55.27 | jblack | Nice truck, drmessano |
| 17:55.27 | luckman212 | haha |
| 17:55.34 | russellb | anonymouz666: haven't chosen a quote yet :-) |
| 17:55.38 | luckman212 | well Xlite v3 can't dial SIP URIs |
| 17:56.31 | drmessano | luckman212, you said it did? |
| 17:56.51 | jblack | https://secure.wikimedia.org/wikipedia/en/wiki/List_of_SIP_software |
| 17:56.58 | luckman212 | I meant, it accepted a "-dial:xxxxx" commandline parameter |
| 17:57.18 | luckman212 | but it is not able to dial something like 1234@loligo.com for example |
| 17:57.24 | *** join/#asterisk voxter (~voxter@macpro.daytonhome.voxter.net) |
| 17:57.56 | *** part/#asterisk jpvoip (~petrybr@189.73.117.170) |
| 17:58.38 | drmessano | X-lite V3 can dial SIP URIs |
| 17:58.42 | drmessano | I know that for a fact |
| 17:58.48 | luckman212 | jblack: thanks, I was on that page earlier today. I tested Blink, Ekiga, SIP Communicator, OfficeSIP messenger, X-Lite v4 and the 3CX softphone. None of them would accept commandline parameters so none were suitable for protocol handlers. (for click to dial from a browser) thats why I was wondering if anyone in here had another suggestion |
| 17:59.09 | drmessano | X-Lite V3 will do what you're asking.. So why not just use it? |
| 17:59.18 | luckman212 | drmessano: yes you're right, it can- but it seems to want to send them to the PBX that you are registered to. In my case, my PBX doesn't understand how to dial SIP URIs so the dial fails. |
| 17:59.32 | luckman212 | V4 allows you to register an account on the pbx, but still "direct dial" a URI |
| 17:59.34 | jblack | wow. back and forth. |
| 17:59.50 | luckman212 | but sadly they removed the commandline feature |
| 17:59.52 | drmessano | luckman212, how does your PBX not understand how to dial SIP URIs? |
| 17:59.54 | jblack | so set up your pbx to route the call. |
| 18:00.02 | drmessano | luckman212, DNS? |
| 18:00.32 | leifmadsen | http://www.youtube.com/watch?v=eDU0CTDMk2g&feature=related |
| 18:00.35 | leifmadsen | you're welcome |
| 18:00.36 | drmessano | luckman212, and who cares what they did in V4.. if V3 works, and has worked fine for a number of years.. you have your answer. Now to fix the URI issue |
| 18:00.42 | luckman212 | drmessano: I am using PiaF and well, it doesn't... I tried but its based on Freepbx and I think their dialplans/GUI doesnt support [a-z] type matching for outgoing |
| 18:01.03 | jblack | leifmadsen: I'm a diabetic. :( |
| 18:01.09 | drmessano | luckman212, that's not the case.. I use FreePBX and I have always been able to dial SIP URI's |
| 18:01.12 | leifmadsen | jblack: sorry to hear that :( |
| 18:01.30 | jblack | That's like waving meth in front of a recovering addict. :'( |
| 18:01.34 | drmessano | luckman212, check the CLI and see what is really happening there.. |
| 18:01.40 | leifmadsen | jblack: I'd probably do that too though |
| 18:02.05 | jblack | Oh, well as long as you're consistent. :) |
| 18:02.22 | luckman212 | drmessano: what happens is, e.g If I try to dial 1234@loligo.com, it instead dials "1234" --- it chops everything after the @ so instead it rings x1234 on my local pbx |
| 18:02.46 | jblack | Now I'm watching stupid music videos on youtube. |
| 18:03.04 | jblack | Right now the hippo in the jungle one... that I think I've already seen a dozen times. |
| 18:03.40 | drmessano | luckman212, sip:1234@server.com |
| 18:04.33 | voxter | leifmadsen: i see you ack'ed an issue of mine earlier. does ack just mean we're looking to see if its a bug, or that its confirmed |
| 18:05.31 | came0 | whats the extension that speaks the extension number youre calling from? |
| 18:06.37 | *** join/#asterisk mpe (~mpe@85.24.87.252.generic-hostname.arrownet.dk) |
| 18:06.42 | *** join/#asterisk jpvoip (~petrybr@189.73.117.170) |
| 18:06.42 | luckman212 | drmessano: 1234@loligo.com or sip:1234@loligo.com -- both give me the same result, I get: Executing [1234@from-internal:1] Playback("SIP/705-00000025", "demo-congrats") in new stack |
| 18:07.03 | luckman212 | it routes to my from-internal context |
| 18:07.12 | jpvoip | what is the pbx libs path? |
| 18:07.37 | drmessano | luckman212, perhaps you have something setup incorrectly in X-Lite. I know this works, because I have dialed other SIP URI's with X-Lite v3 before.. even going back to older FreePBX/Trixbox releases |
| 18:08.00 | *** join/#asterisk [netman] (~netman@83.54.219.152) |
| 18:08.28 | *** join/#asterisk mercutioviz (~michaelco@freeswitch/developer/msc) |
| 18:08.33 | luckman212 | the stupid thing is, I deleted my xlite v3 when I installed v4 earlier today. Now I don't have a v3 installer and they don't seem to offer the v3 download anymore. feeling very silly right now. |
| 18:08.43 | luckman212 | I'm going to try to restore v3 from a backup |
| 18:09.01 | mercutioviz | hey, does anyone here know what happened to voip abuse project/infiltrated.net? the domain name expired |
| 18:09.08 | luckman212 | jpvoip: not sure quite what you're asking? /var/lib/asterisk? |
| 18:09.22 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 18:09.59 | drmessano | luckman212, were you just doing all this on V4? |
| 18:10.35 | luckman212 | drmessano: no I did test with v3, but perhaps I missed some option somewhere |
| 18:11.03 | *** join/#asterisk gabdrach (~Adium@190.177.79.190) |
| 18:11.36 | drmessano | Ok, but you just told me you tried it 2 different ways and i didn't work.. You tried sip:number@host with v3 earlier? |
| 18:11.38 | *** join/#asterisk bmg505 (~leon@196-209-120-122.dynamic.isadsl.co.za) |
| 18:11.49 | luckman212 | the interesting thing is in v4, there is an 'advanced' setting that says "send outgoing request directly to target" -- once I enabled that, I was able to place calls to SIP URIs directly, bypassing my local pbx. I dont remember seeing such an option in v3 but maybe i missed it. |
| 18:11.55 | luckman212 | http://i.imgur.com/MlMcJ.png |
| 18:12.02 | luckman212 | that was the key |
| 18:12.05 | gabdrach | hi, im using asterisk 1.8 and set verbose 1 on cli, but still get meesage like '-- Locally bridging SIP/voxbone_be2-0000493f and SIP/jcsip.free411.com-00004948' how can i disable those messages? |
| 18:12.11 | *** part/#asterisk jpvoip (~petrybr@189.73.117.170) |
| 18:12.22 | notax | gabdrach, core set verbose 0 |
| 18:13.12 | gabdrach | notax, thanks |
| 18:13.13 | drmessano | luckman212, not sure what to tell you.. You already stated v4 wouldn't handle part of your need. The entire context of what I was discussing was in regard to V3. You seem to be pretty stuck on using V4, which is silly. |
| 18:13.46 | leifmadsen | voxter: just means I don't see any particular reason to close it and that I don't need any more information from you. It'll be assessed and then assigned to a developer at some point, or we'll ask you more questions (or someone will comment on the issue at some point) |
| 18:13.55 | luckman212 | not at all, I dont care about v4 at all. in fact I prefer v3 but as I said I wasn't able to get v3 to dial SIP URIs directly |
| 18:14.04 | gabdrach | notax, no way. still showing :( |
| 18:14.09 | voxter | leifmadsen: cool thanks :) |
| 18:14.15 | drmessano | http://support.xten.net/ <--- Search the forums to download V3.. they always have links |
| 18:14.39 | notax | gabdrach, weird, maybe it's a bug or they changed it in 1.8 |
| 18:16.17 | luckman212 | drmessano: ok thanks, i do appreciate the help |
| 18:16.21 | *** join/#asterisk tris (~tristan@CAMEL.ETHEREAL.NET) |
| 18:17.11 | *** join/#asterisk dwery (~dwery@nslu2-linux/dwery) |
| 18:17.41 | dwery | hello. I'd need a simple SIP keypad for outdoor operations at a gate. Anyone has suggestions? |
| 18:17.52 | *** join/#asterisk bmint (~bmint@h174.92.190.173.static.ip.windstream.net) |
| 18:22.28 | voxter | hey polycom people, for ip550/650 should i be using this new UC firmware ? |
| 18:29.32 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 18:29.43 | notax | dwery, use a cheap ip phone? |
| 18:29.53 | dwery | notax: needs to be weatherproof |
| 18:30.19 | bmint | If I place a call on hold Asterisk drops that call after 61 seconds. Disconnecting call 'SIP/XXXX' for lack of RTP activity in 61 seconds. Anyone have an idea as to what is going on? |
| 18:31.14 | notax | bmint, you could try with qualify |
| 18:34.51 | *** join/#asterisk [netman] (~netman@83.54.223.63) |
| 18:40.18 | *** join/#asterisk Carlos_PHX_ (~Carlos@ip68-99-199-10.ph.ph.cox.net) |
| 18:43.07 | *** join/#asterisk [netman] (~netman@72.Red-88-19-41.staticIP.rima-tde.net) |
| 18:46.26 | luckman212 | drmessano: I restored my X-lite v3 and I can click on sip:// links again. But it routes all calls thru my pbx. I couldn't find that "send outgoing request directly to target" feature anywhere in v3. you said you were able to make sip uri calls via xlite v3-- do you mind telling me a bit more about your setup? are you running your own asterisk server or do you just register with an ITSP |
| 18:55.20 | *** join/#asterisk tris (~tristan@CAMEL.ETHEREAL.NET) |
| 18:59.26 | *** join/#asterisk Defraz (~Defraz@63.226.95.147) |
| 19:02.22 | *** join/#asterisk bbryant (~brett@c-76-26-221-79.hsd1.sc.comcast.net) |
| 19:03.35 | *** topic/#asterisk by leifmadsen -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.0 (2010/10/21), 1.6.2.14 (2010/11/11), 1.4.37 (2010/11/11), *-Addons 1.6.2.2, 1.4.12 (2010/09/22), dahdi-linux 2.4.0 (2010/09/01), dahdi-tools 2.4.0 (2010/09/01), libpri 1.4.11.4 (2010/09/01) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
| 19:06.20 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-163.hsd1.nj.comcast.net) |
| 19:11.52 | bbeers | Can I disable sending RTCP when SRTP is on? (srtpcapable=yes, encryption=yes) |
| 19:12.36 | *** part/#asterisk dwery (~dwery@nslu2-linux/dwery) |
| 19:15.05 | *** join/#asterisk pgarcia (~chatzilla@yoda.kanatek.com) |
| 19:16.45 | pgarcia | When enabling g729 on Asterisk (encoding/decoding), is there any check before completing a SIP call? Actually, I'd like to know what is the behavior when you have, let's say 5 g729 licenses and tries to make the 6th call, considering all end-points can handle g729 AND uLaw, for example. |
| 19:23.27 | pgarcia | I'm wondering if is there a way to fallback to the next available codec in case I've run out of g729 licenses, or if the call is rejected. |
| 19:23.36 | *** part/#asterisk mercutioviz (~michaelco@freeswitch/developer/msc) |
| 19:27.13 | *** join/#asterisk myster (~myster@207.148.172.210) |
| 19:28.35 | *** part/#asterisk myster (~myster@207.148.172.210) |
| 19:48.10 | leifmadsen | with Set(FORWARD_CONTEXT=some_context), if I do a Dial() to a device that sends back a 302, the Local channel should be trying to dial the requested forwarding number/location in the context defined by that FORWARD_CONTEXT channel variable right? |
| 19:48.11 | leifmadsen | pabelanger: ? |
| 19:48.39 | pabelanger | leifmadsen: zomgpong |
| 19:48.50 | leifmadsen | see my question above? am I off my rocker? |
| 19:49.05 | jaytee | drmessano ping |
| 19:49.29 | pabelanger | leifmadsen: promiscredir = yes ? |
| 19:50.45 | leifmadsen | hmmm |
| 19:50.58 | leifmadsen | already enabled |
| 19:56.56 | pabelanger | leifmadsen: I never set FORWARD_CONTEXT when dealing with 302 redirects. Asterisk would dial the number properly. That was under 1.6.2 |
| 19:57.33 | leifmadsen | ya this is 1.6.2.11 |
| 19:57.45 | leifmadsen | right now it tries to use the Local channel to dial in the default context |
| 19:57.52 | leifmadsen | regardless of how I set FORWARD_CONTEXT |
| 19:58.09 | leifmadsen | No such extension/context 4165551212@default while calling Local channel |
| 19:59.34 | pabelanger | leifmadsen: Hmm, I never used it with local channels. Just chan_sip directly |
| 19:59.47 | leifmadsen | no that's how it gets done automatically |
| 19:59.56 | leifmadsen | when a 302 comes back, Asterisk tries to forward it with the Local channel |
| 20:00.07 | leifmadsen | when calling a SIP end point |
| 20:00.33 | pabelanger | Hmm, ok. Its been awhile, don't remember setting it up for my client/ |
| 20:01.28 | leifmadsen | ya it's not anything you need to setup or enable since that's the default way it works. I remember before just setting FORWARD_CONTEXT Channel variable though |
| 20:01.36 | leifmadsen | I wonder if this is done via a function now... hmm |
| 20:02.35 | bbeers | anyone using SRTP? |
| 20:03.03 | leifmadsen | what is your real question? |
| 20:03.48 | bbeers | I see RTCP messages going out while using SRTP. I want to disable that. |
| 20:04.39 | bbeers | Or send SRTCP. |
| 20:23.31 | *** join/#asterisk Linuturk (~linuturk@unaffiliated/linuturk) |
| 20:31.37 | mintee | how would one export and import the asterisk database |
| 20:35.29 | *** join/#asterisk asteriskmonkey (~darlek@74-51-38-204.telnetcommunications.com) |
| 20:36.18 | asteriskmonkey | anyone know how to make a polycom immediatly if the primary address in the line address fails? |
| 20:36.24 | asteriskmonkey | dial that is :P |
| 20:42.38 | thehar | win 13 |
| 20:42.40 | thehar | fail |
| 20:49.23 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 20:50.05 | keith4 | if I just have wav and gsm versions for MOH, will asterisk transcode to ulaw/g729/whatever as necessary? |
| 20:52.13 | wdoekes2 | yes |
| 20:53.15 | lpmusic | it'll only transcode to g729 if you have the codec licensed though (so if you have it pre-encoded to g729 you'll be fine) |
| 20:54.39 | keith4 | i had some pre-encoded in a handful of formats, but customer wanted volume reduced by 30%. i used sox to lower the level, but it seems to not like g729 input files. or perhaps i'm just missing some sox libs |
| 20:55.15 | keith4 | in fact, it seems to have ignored the alaw and ulaw formats, too. hmm. |
| 20:55.31 | frigidzephyr | "core show translation" to see what can be transcoded |
| 20:56.03 | frigidzephyr | intersections with "-" cannot be transcoded |
| 20:57.05 | keith4 | ah, brilliant |
| 20:58.02 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 21:10.31 | *** join/#asterisk fofware (~Fabian@host186.190-225-12.telecom.net.ar) |
| 21:23.12 | *** join/#asterisk joesuffceren (~chatzilla@ip68-104-167-226.ph.ph.cox.net) |
| 21:26.01 | joesuffceren | I made a boneheaded mistake and bought a digium analog interface (4 FXO with echo cancellation module) to put in a box with *drum roll* no PCI slots. I need to return it to get a PCI-e one, but digium's PCI-e cards are more expensive and beyond my budget. Any recommendations for good quality but cheaper brands I can replace with? |
| 21:27.44 | *** join/#asterisk GNUtoo|laptop (~gnutoo@host31-144-dynamic.24-79-r.retail.telecomitalia.it) |
| 21:27.54 | Qwell | joesuffceren: the difference in price should be fairly minimal |
| 21:28.16 | *** part/#asterisk wesphillips (~wphillips@64-19-240-7.caprock.net) |
| 21:29.08 | GNUtoo|laptop | hi, how do you block attacker from trying all the possible passwords and using 100% of the CPU? |
| 21:29.14 | Qwell | firewall them |
| 21:29.28 | WIMPy | GNUtoo|laptop: Cut your internet connection. |
| 21:29.49 | GNUtoo|laptop | is there something like sshblock for asterisk? |
| 21:29.55 | *** join/#asterisk a_bug (~abhinav@122.176.160.115) |
| 21:30.49 | a_bug | hi guys, a question please - I'm using pika with asterisk, and on originating a call, I get the following error on releasing the channel : -- <PIKA/digital/1> Playing 'vm-sorry' (language 'default') |
| 21:30.53 | GNUtoo|laptop | obviously asterisk dooesn't use pam |
| 21:30.54 | a_bug | [Nov 19 23:53:17] ERROR[2858]: chan_pika.c:4872 do_GP_event: PKX_CALL_Release(0x01000022) failed, status = [-0x5103] PKX_ERROR_CALL_IN_WRONG_STATE |
| 21:31.00 | GNUtoo|laptop | so can't be configured from there |
| 21:31.02 | Qwell | a_bug: ask PIKA |
| 21:31.07 | a_bug | oh |
| 21:31.12 | joesuffceren | GNUtoo|laptop: you can use fail2ban |
| 21:31.15 | a_bug | ok thanks Qwell |
| 21:31.21 | GNUtoo|laptop | fail2ban works with asterisk? |
| 21:31.25 | GNUtoo|laptop | if so thanks a lot |
| 21:31.49 | joesuffceren | GNUtoo|laptop: may not natively, but can be configured to do so. I think I had to install a regex specifically for it. it's been a while |
| 21:31.53 | GNUtoo|laptop | ok |
| 21:32.04 | joesuffceren | Qwell: differnce is like 25% |
| 21:32.06 | GNUtoo|laptop | I bet you have to give it the logs locations too |
| 21:32.22 | Qwell | joesuffceren: for just the baseboard, maybe. it uses the same modules. |
| 21:32.26 | joesuffceren | GNUtoo|laptop: correct tell it where the log is and what error messages it should look for |
| 21:32.41 | GNUtoo|laptop | ok |
| 21:33.02 | joesuffceren | qwell: oh, that's an interesting thought. wonder if the vendor would let me return just the base PCI board and keep the modules. :-) |
| 21:33.04 | joesuffceren | probably not |
| 21:33.11 | Qwell | joesuffceren: maybe! ask |
| 21:33.32 | Qwell | maybe they'd ship you the pci-e one (if you send the pci back) for just the difference.. who knows |
| 21:33.52 | Qwell | I'm sure for some resellers, that wouldn't be completely out of the realm of possibility |
| 21:33.56 | joesuffceren | Qwell: thanks for the info about their sharing modules. didn't realize that. I'll definitely ask! |
| 21:34.07 | joesuffceren | ipphone-warehouse. familiar with them? |
| 21:34.18 | joesuffceren | if that idea doesn't work, any other brands you'd recommend, or stay away? |
| 21:34.24 | Qwell | I'm not 100% sure on the echocan module though. my comment was wrt the fxo/fxs modules. |
| 21:34.35 | Qwell | stay away from other brands.</obvious bias> |
| 21:35.08 | Qwell | but really, even if you have to replace all of it, the cost shouldn't be any more than what the base board would be - since it does use the same modules, they'd be the same price obviously (hopefully..) |
| 21:35.11 | joesuffceren | I've never used anything else but digium, so I'm a bit biased as well... grrr, probably just have to eat my mistake |
| 21:38.43 | joesuffceren | qwell: The VPMADT032 may be used with all of Digium's analog cards (TDM410, AEX410, TDM800, AEX800, TDM2400, AEX2400) as well as Digium's single-port T1/E1 cards (TE122, TE121). |
| 21:38.49 | joesuffceren | yay |
| 21:41.05 | WIMPy | BTW: Does anyone know what happens when you put a VPM400M on a TE405P? Will on only work for the first two ports? |
| 21:41.36 | *** part/#asterisk asteriskmonkey (~darlek@74-51-38-204.telnetcommunications.com) |
| 21:46.39 | p3nguin | In the worst case, it'll catch on fire and burn to the ground. |
| 21:51.51 | WIMPy | It doesn't |
| 21:52.15 | WIMPy | In fact it looks like it works on all channels, but I guess, it can't. |
| 21:53.03 | carrar | It explodes |
| 21:53.21 | *** join/#asterisk mpe (~mpe@85.24.87.252.generic-hostname.arrownet.dk) |
| 21:53.23 | carrar | and then you get rickrolled |
| 21:55.46 | *** join/#asterisk ccesario (~ccesario@201-42-156-44.dsl.telesp.net.br) |
| 22:09.39 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 22:09.52 | jaytee | be back later |
| 22:09.55 | *** part/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 22:11.30 | *** join/#asterisk wikii (~wiki.mir@host-172-net-105-160-119.mobilinkinfinity.net.pk) |
| 22:13.20 | wikii | i have installed hylafax. i can send faxes on lacal numbers but i cannot sent faxes to another STATE(area code) number. please find my logs at http://ja.pastebin.ca/1996603 |
| 22:14.11 | wikii | anyone please help |
| 22:16.03 | WIMPy | wikii: Does that look like some GUI stuff? Otherwise Line 103 makes it clear. |
| 22:16.43 | wikii | yeah it is cli log |
| 22:17.51 | *** join/#asterisk diemos (~diemos@173-13-138-49-sfba.hfc.comcastbusiness.net) |
| 22:17.57 | diemos | herrrrroooooo |
| 22:18.04 | wikii | when i dial number with area code it shows hangup cause 28.. and without area code it sends fax |
| 22:18.25 | WIMPy | wikii: Fix your dialplan. |
| 22:18.37 | wikii | there are three different logs in that post |
| 22:18.58 | wikii | last log show successful fax sending |
| 22:19.24 | wikii | in my dialplan i have use sendfax command |
| 22:19.31 | wikii | nothing else |
| 22:19.36 | *** join/#asterisk luckman212_phone (~luckman21@rrcs-184-75-3-130.nyc.biz.rr.com) |
| 22:20.05 | WIMPy | You are sending an invalid number. Use a valid one. |
| 22:20.48 | wikii | acutally i want to send fax to another state in US therefore i am dialling number with area code |
| 22:21.15 | wikii | i called that number ..its a valid number |
| 22:21.26 | *** join/#asterisk kbr (~kbr@2a01:e35:2e31:f940:221:85ff:fee6:2e95) |
| 22:21.51 | WIMPy | Then your dialplan screwed it. |
| 22:22.26 | wikii | i tried to edit dialrules for hylafax... but no success |
| 22:23.54 | *** join/#asterisk jblack (~jblack@71.181.209.104) |
| 22:25.05 | wikii | WIPMY : please give me example of dialplan .. how to fix my dialplan |
| 22:25.29 | citywok | why don't you post YOUR dialplan |
| 22:26.27 | wikii | line 5 to line 8 is my dialplan |
| 22:26.39 | WIMPy | wikii: I don't know what number you're trying to reach on on what format your telco expects it. I can only see that they sya it's not a valid number. |
| 22:27.21 | wikii | i have used sendfax -n -d 732124567 /var/ftp/[filename] |
| 22:27.25 | citywok | why is it SIP when you dial 1732, and IAX2 if you don't dial with the 1732 |
| 22:27.30 | *** join/#asterisk SparFux (~raoul@e182017048.adsl.alicedsl.de) |
| 22:28.03 | SparFux | Hi. In DAHDI, can I snoop q931 / q921 on the fly? |
| 22:28.03 | citywok | not to mention your syntax is wrong for the filename, why are there quotes mid name |
| 22:28.18 | wikii | acutally i am pasing values for C# code therefore i have usedsipfor orignate method |
| 22:28.27 | WIMPy | SparFux: pri set debug... |
| 22:28.40 | SparFux | WIMPy: in * console? |
| 22:28.51 | WIMPy | SparFux: yes |
| 22:29.43 | wikii | acutally i am pasing values[fax number and file name] from C# code therefore i have usedsipfor orignate method |
| 22:29.53 | WIMPy | wikii: You are actually dialling 17328... Guess that's the problem. |
| 22:30.18 | citywok | yes, but the ' is in the middle of the file name |
| 22:30.30 | SparFux | WIMPy: thx. |
| 22:30.32 | citywok | -- /var/ftp/'1110_58_30_0331166.pdf' doesn't make sense |
| 22:30.37 | wikii | i have tried without dialing 1 in the begining..but no success |
| 22:30.58 | citywok | wikii then you need to figure out what you are supposed to do to dial long distance |
| 22:31.13 | *** join/#asterisk jblack (~jblack@71.181.209.104) |
| 22:31.13 | wikii | city : yeah i have removed those lines but no success |
| 22:31.35 | wikii | long dialing prefix is 1 |
| 22:32.29 | wikii | it can send faxes on local number...without are code |
| 22:32.33 | *** part/#asterisk SparFux (~raoul@e182017048.adsl.alicedsl.de) |
| 22:32.34 | wikii | area* |
| 22:37.30 | *** join/#asterisk myster (~myster@74.198.148.18) |
| 22:37.54 | wikii | WIMPY i have tried without long distance dialing prefix.. but it shows the same result |
| 22:38.41 | *** join/#asterisk luckman212_phone (~luckman21@rrcs-184-75-3-130.nyc.biz.rr.com) |
| 22:43.53 | wikii | Wimpy: city : thankyou for your time. |
| 22:44.25 | *** part/#asterisk wikii (~wiki.mir@host-172-net-105-160-119.mobilinkinfinity.net.pk) |
| 22:55.04 | *** part/#asterisk myster (~myster@74.198.148.18) |
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