| 00:02.50 | seb- | [TK]D-Fender: still there? |
| 00:17.26 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 00:18.19 | seb- | [TK]D-Fender: ping? sorry i missed you by 30min! |
| 00:18.21 | seb- | aw |
| 00:25.09 | *** join/#asterisk qdk (n=qdk@0x573d8dd1.bynqu1.dynamic.dsl.tele.dk) |
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| 02:07.01 | luckyaba | is there a way to log a user into a queue via the AMI ? |
| 02:08.53 | *** join/#asterisk siera08 (n=chatzill@218.207.141.90) |
| 02:10.45 | siera08 | hi, does anyone tell me about any IPPBX supporting bugging? |
| 02:21.46 | BeeBuu | luckyaba: maybe use "command" for that |
| 02:27.21 | luckyaba | Looking through the command info BeeBuu but not having a lot of lucj |
| 02:27.23 | luckyaba | luck** |
| 02:27.41 | luckyaba | Did however come across some other useful things while reading through this! |
| 02:33.54 | ManxPower | there is no such thing as a user, you should be able to add a device to a queue using AMI. |
| 02:34.21 | *** join/#asterisk heison (n=heison@204.29.161.34) |
| 02:35.26 | heison | hello... |
| 02:35.35 | heison | does anyone know if asterlink is still in business? |
| 02:36.06 | luckyaba | Maximo, Sorry.. I meant log an extension into a queue. |
| 02:36.15 | luckyaba | err ManxPower |
| 02:36.54 | luckyaba | I have a queue with 2 people in it and I want to auto login a couple other users if the queue hits say 5 users in it |
| 02:37.07 | luckyaba | or 5 customers calling in |
| 02:37.23 | luckyaba | queue with 2 extensions logged into it |
| 02:37.49 | luckyaba | and log a couple other extensions into it |
| 02:47.17 | *** join/#asterisk chendy (n=chatzill@59.40.164.130) |
| 02:49.08 | *** join/#asterisk yacc_ (n=andreas@91-115-19-30.adsl.highway.telekom.at) |
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| 03:19.18 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
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| 03:22.45 | carrar | woops |
| 03:22.47 | carrar | ~pb |
| 03:22.48 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/ |
| 03:22.49 | carrar | #$!@# |
| 03:25.36 | *** join/#asterisk pawz (n=pawz@124-254-81-250-static-dsl.ispone.net.au) |
| 03:30.56 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 03:42.54 | rue_mohr | can anyone tell me what happens when you put too many analog sets on a pots line at once (off hook) |
| 03:42.54 | BeeBuu | luckyaba: i think the "redirect" in AMI is what you need |
| 03:53.10 | luckyaba | BeeBuu, Yeah I think that may be what I am looking for... I could redirect the call to an overflow queue |
| 03:53.32 | luckyaba | that would allow the folks who normally don't take the queue calls to only get them when its busy..... |
| 03:55.37 | *** join/#asterisk chendy (n=alex@59.40.164.130) |
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| 04:01.08 | BeeBuu | luckyaba: i can help people is happy |
| 04:01.47 | BeeBuu | in fact ,that command is someone told me in here~~~~ :-P |
| 04:03.28 | *** join/#asterisk MrNaz (n=mrnaz@203.214.68.222) |
| 04:03.48 | luckyaba | haha, right on |
| 04:09.04 | drmessano | ~~~~~~ |
| 04:09.05 | infobot | ACTION takes out a revolver and shoots drmessano in the head three times. |
| 04:09.18 | drmessano | ~~~~~~~ |
| 04:09.19 | infobot | ACTION lets a freakishly huge killer whale named Hugh eat drmessano. |
| 04:09.26 | drmessano | ~~~~~~~~ |
| 04:09.27 | infobot | You know, this got old a long time ago. |
| 04:09.34 | drmessano | ~~~~~~~~~ |
| 04:09.35 | infobot | I'm ignoring you now. |
| 04:09.35 | jaytee | heheehe |
| 04:09.42 | drmessano | ~~~~~~~~~~ |
| 04:09.53 | drmessano | ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ |
| 04:10.00 | drmessano | I killed him |
| 04:10.02 | drmessano | :( |
| 04:10.11 | jaytee | ~botsnack |
| 04:10.11 | infobot | aw, gee, jaytee |
| 04:11.03 | *** join/#asterisk elitecoder (n=liq@apollo.bullethost.com) |
| 04:12.22 | elitecoder | Hi all. I'm trying to pass variables to an AGI script: http://pastebin.com/d69aca33e |
| 04:12.29 | elitecoder | I passed it but it's not showing up in the variable list |
| 04:12.55 | elitecoder | this is what I get: http://pastebin.com/d7e795429 |
| 04:13.52 | elitecoder | I'm guessing I should use some other application like set maybe |
| 04:19.28 | Micc | anyone know why a linux firewall would only allow one ip phone to register but the second always fails? |
| 04:19.49 | Micc | Its like it doesn't know where to send the udp packets to for the second registration. |
| 04:20.18 | elitecoder | Micc: I'm going to take a guess here and say that because it's trying to use a different (random?) port that you're not allowing on your firewall. |
| 04:20.49 | Micc | elitecoder, both phones work fine if they are the first one. |
| 04:21.12 | Micc | elitecoder, all phones have to register on the 5060 port first, right? |
| 04:21.22 | elitecoder | oh register |
| 04:21.23 | Micc | then they negotiate a port after that. |
| 04:21.25 | elitecoder | ok nevermind |
| 04:21.26 | drmessano | Not necessarily |
| 04:21.38 | Micc | I'm only listening on 5060 |
| 04:21.47 | Micc | as far as I know asterisk can only listen to one port. |
| 04:21.48 | drmessano | They can use whatever SOURCE port they want |
| 04:22.06 | drmessano | They all register TO 5060 on asterisk |
| 04:22.50 | drmessano | Sounds like the firewall is misconfigured |
| 04:25.54 | rue_mohr | no |
| 04:26.07 | rue_mohr | RTP = "Really?! that port!?" |
| 04:26.16 | rue_mohr | which is 10000 |
| 04:26.18 | rue_mohr | iirc |
| 04:26.38 | rue_mohr | maybe I missed the point, by |
| 04:26.57 | drmessano | RTP isnt just 10000 |
| 04:27.06 | Micc | drmessano, I'm having them open up ports below 5060 from 1024 to 5061 |
| 04:27.09 | drmessano | and its registration |
| 04:27.17 | rue_mohr | yea |
| 04:27.21 | rue_mohr | goes up from there |
| 04:27.26 | drmessano | micc: Opening ports is not the issue |
| 04:28.04 | Micc | drmessano, it seems like a solution. It might not be the issue. |
| 04:28.17 | drmessano | Its NOT a solution |
| 04:28.22 | drmessano | It will NOT work |
| 04:28.41 | Micc | drmessano, then what will? |
| 04:28.56 | drmessano | Your firewall does not need ports forwarded to the inside for SIP CLIENTS |
| 04:29.03 | drmessano | It needs to handle SIP properly |
| 04:29.29 | Micc | drmessano, right, but if it doesn't know how to handle sip properly, wouldn't it help to open some ports? |
| 04:30.13 | Micc | drmessano, is there some iptables document that shows how to setup iptables to handle SIP properly? |
| 04:30.35 | drmessano | What are you gonna open ports to? |
| 04:30.57 | drmessano | Just open a bunch of ports, hope for the best? |
| 04:31.13 | Micc | from our asterisk server to the phone. |
| 04:32.03 | Micc | drmessano, If you watch the udp traffic hitting the firewall it tries a bunch of ports. But this phone in particular has a pattern and it usually will register the next phone down around 1027. |
| 04:34.06 | *** join/#asterisk TiPsC (n=TiPsC@adsl-074-186-104-155.sip.mia.bellsouth.net) |
| 04:37.10 | Micc | why does it show Nat N for all my clients when I do sip show peers? |
| 04:37.15 | Micc | In my sip.conf all of them have nat=yes |
| 04:37.35 | Micc | does N mean nat=ys? |
| 04:37.41 | Micc | and blank means nat=no? |
| 04:39.43 | carrar | heh |
| 04:40.09 | carrar | GOOD TIMES |
| 04:40.38 | carrar | Micc |
| 04:40.49 | carrar | Whast the FIRST line you see when you do a sip show peers |
| 04:41.18 | carrar | well I guess you said that above |
| 04:41.41 | drmessano | This is where the whole "If you had all the answers, then why don't you have it working" comes in. I hate people that tell you how to fix something they just asked for help with. |
| 04:41.54 | carrar | http://www.voip-info.org/wiki/view/asterisk+cli+command+sip+show+peers |
| 04:43.50 | drmessano | I would open ports 1 to 79, and 81 to 65534 |
| 04:44.01 | drmessano | Leave 80 closed, dont want to get hacked |
| 04:44.08 | carrar | TRUE DAT |
| 04:45.23 | Micc | I still think under Nat it should show Y instead of N |
| 04:45.36 | Micc | that page still doesn't make it clear. |
| 04:46.00 | carrar | asterisk -rx "sip show nat" | sed "s/ N / Y /" |
| 04:46.06 | carrar | DONE |
| 04:46.09 | carrar | err |
| 04:46.14 | carrar | asterisk -rx "sip show peers" | sed "s/ N / Y /" |
| 04:46.16 | drmessano | [00:46] <Micc> I still think under Nat it should show Y instead of N <-- FAIL |
| 04:46.35 | carrar | all your N's are now Y's!! |
| 04:47.13 | carrar | Please submit that code change to digium |
| 04:47.15 | rue_mohr | ~pb |
| 04:47.16 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/ |
| 04:47.17 | drmessano | ROFL |
| 04:47.20 | rue_mohr | http://rafb.net/paste/ |
| 04:47.29 | rue_mohr | better take a closer look and remove it from the list |
| 04:49.32 | Micc | maybe the other letters are reserved for other things, so N = Nat=yes, R = nat=route, blank = nat=no, hmm what would nat=never be then? |
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| 06:02.31 | kn0x | t |
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| 06:16.31 | *** join/#asterisk b14ck (n=blacky@cpe-98-151-210-28.socal.res.rr.com) |
| 06:16.44 | b14ck | hi all |
| 06:18.42 | *** join/#asterisk dshap (n=IceChat7@ip70-181-91-110.oc.oc.cox.net) |
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| 06:23.07 | dshap | can someone here help me out with the originate CLI command? |
| 06:23.18 | dshap | i'm not sure what to put for the channel argument |
| 06:24.21 | b14ck | for the channel, you specify a channel like: SIP/1111 |
| 06:24.21 | b14ck | eg: Channel: SIP/1111 |
| 06:24.21 | dshap | and what is the 1111? |
| 06:24.21 | dshap | is that the number i want to call? |
| 06:24.21 | b14ck | in this case, 1111 is an extension |
| 06:24.25 | b14ck | ya |
| 06:24.30 | dshap | okay so if i want to dial a PSTN number |
| 06:24.31 | dshap | i just do |
| 06:24.35 | dshap | Channel: SIP/thatnumber |
| 06:24.38 | b14ck | SIP/1111@trunkname |
| 06:24.50 | dshap | okay and "trunkname" |
| 06:24.53 | [TK]D-Fender | dshap: Same way you do for Dial() |
| 06:24.53 | dshap | that's defined in my sip.conf? |
| 06:25.05 | dshap | [TK]D-Fender: never got Dial() to work |
| 06:25.15 | [TK]D-Fender | .... |
| 06:25.16 | b14ck | if you look in your sip.conf for your trunk name eg: [flowroute] or something |
| 06:25.18 | b14ck | then you would do |
| 06:25.23 | [TK]D-Fender | dshap: then what HAVE you done? |
| 06:25.30 | b14ck | Channel: SIP/8882221111@flowroute |
| 06:25.46 | dshap | [TK]D-Fender: receive incoming calls, basic dial plan menu & applications |
| 06:25.55 | dshap | thanks b14ck i'm going to try it now |
| 06:25.59 | b14ck | k |
| 06:26.06 | [TK]D-Fender | dshap: Thats a start I guess.. |
| 06:26.16 | dshap | [TK]D-Fender: u gotta start somewhere hah |
| 06:26.27 | [TK]D-Fender | dshap: rare to take a call in and never call out to anything else... |
| 06:26.30 | KyleK | I've got SIP/trunk/xxxxxx and SIP/xxxxxxx@trunk in my dialplan |
| 06:26.32 | dshap | true |
| 06:26.55 | dshap | do i NEED to set callerID? |
| 06:27.02 | b14ck | anyone in here interested in working on a project with me? i am writing a new web interface for asterisk based pbx systems. a freepbx alternative |
| 06:27.07 | dshap | or could i just make a .call file with 3 lines: channel, application, and data |
| 06:27.08 | [TK]D-Fender | dshap: Usually no |
| 06:27.22 | KyleK | dshap: only if you care what the caller id says |
| 06:27.26 | dshap | got it |
| 06:27.37 | dshap | also this book i'm reading says that i should worry about how the call file is moved into the directory |
| 06:27.42 | dshap | it says there is a difference between cp and mv |
| 06:27.47 | KyleK | yup |
| 06:27.47 | dshap | what if i'm uploading via FTP |
| 06:27.48 | dshap | is that okay? |
| 06:27.54 | b14ck | dshap, do a mv only |
| 06:27.55 | [TK]D-Fender | dshap: Horribly worse |
| 06:27.57 | KyleK | you can move from ftp |
| 06:27.58 | b14ck | do not ftp it to the directory |
| 06:28.14 | dshap | ok so i can FTP it to the parent directory |
| 06:28.17 | dshap | then mv into outgoing |
| 06:28.24 | b14ck | ftp it to like /tmp, then just mv it to outgoing, ya |
| 06:28.28 | [TK]D-Fender | dshap: AMI Originate <-- |
| 06:28.28 | dshap | word. |
| 06:28.45 | b14ck | dont forget to: chmod 755 && chown asterisk:asterisk it before the mv! |
| 06:29.00 | b14ck | well i guess the chmod isnt important, actually |
| 06:29.01 | KyleK | why does it need execute permissions? |
| 06:29.04 | b14ck | but the chown is |
| 06:29.12 | dshap | not sure what chown is |
| 06:29.19 | b14ck | chown changes the file's permissions |
| 06:29.28 | b14ck | so that asterisk can read/write to the file and own it |
| 06:29.28 | KyleK | dshap: just needs to be readable and deletable by asterisk |
| 06:29.31 | dshap | o i thought that was chmod |
| 06:30.52 | dshap | "chown asterisk:asterisk test.call" ? |
| 06:31.07 | KyleK | yup |
| 06:31.28 | dshap | it says invalid user |
| 06:31.53 | [TK]D-Fender | dshap: Did you create that user? |
| 06:32.03 | [TK]D-Fender | dshap: Is that the user you believe * runs as? |
| 06:32.10 | dshap | omg |
| 06:32.12 | dshap | hahaha |
| 06:32.13 | dshap | i tried it anyways |
| 06:32.16 | dshap | and it worked! |
| 06:32.28 | dshap | my asterisk server just called my cell phone |
| 06:32.30 | dshap | this is awesome |
| 06:32.33 | [TK]D-Fender | dshap: probably because * is running a ROOT. |
| 06:32.37 | dshap | yes |
| 06:32.38 | dshap | that is true |
| 06:32.40 | dshap | im running as root |
| 06:32.48 | dshap | so chown asterisk:root test.call? |
| 06:32.51 | dshap | or the other way around |
| 06:33.00 | dshap | or it's not even necessary because it just worked |
| 06:33.01 | [TK]D-Fender | dshap: No need |
| 06:33.02 | drmessano | [02:25] <dshap> [TK]D-Fender: never got Dial() to work <-- WIN |
| 06:33.05 | KyleK | root doesn't care about permissions |
| 06:33.23 | dshap | gotcha |
| 06:33.44 | dshap | so what's so bad about FTP'ing the call file? |
| 06:33.47 | b14ck | dshap, it is fun when you get your first call working :) |
| 06:33.57 | dshap | it gets part of it before it gets all of it? |
| 06:34.00 | KyleK | dshap: wrong question |
| 06:34.15 | KyleK | "whats wrong about copying the file?" cos ftping and them moving is fine :) |
| 06:34.21 | dshap | got it |
| 06:34.27 | dshap | so copying the file it has to make a whole new file |
| 06:34.34 | dshap | so it writes it a little bit at a time |
| 06:34.36 | dshap | and that'sthe problem |
| 06:34.37 | dshap | yea? |
| 06:34.44 | [TK]D-Fender | dshap: File locking <- |
| 06:34.45 | drmessano | .... |
| 06:34.51 | dshap | oh |
| 06:34.51 | b14ck | bascailly, dshap, asterisk will look for the files you put into outgoing at an extremely aggressive rate |
| 06:34.58 | dshap | understood |
| 06:35.01 | [TK]D-Fender | dshap: The very instant * sees it, * gets "grabby" and FUBAR's |
| 06:35.06 | b14ck | so if you are slowly copying a file into the directory, asterisk will see what is there and try to use the partial file to make the call |
| 06:35.17 | dshap | got it |
| 06:35.18 | b14ck | and when asterisk does that, nobody wins |
| 06:35.21 | b14ck | =) |
| 06:35.36 | dshap | what if i want to have dynamically generated call files though |
| 06:35.44 | dshap | like from a PHP script |
| 06:36.02 | dshap | i have to do the PHP commands for creating a file and then moving it? |
| 06:37.17 | b14ck | yep |
| 06:37.34 | b14ck | but there's an alternative: use the asterisk ami! |
| 06:37.45 | b14ck | using the asterisk AMI is like doing call files through sockets (like networking) |
| 06:38.13 | b14ck | you send asterisk commands in the form of strings (just like a call file), and asterisk reads them in through a socket, and then executes them when you are finished writing |
| 06:38.21 | [TK]D-Fender | b14ck: Funny I said that 10 minutes ago... |
| 06:38.35 | b14ck | [TK]D-Fender, I was just answering his question. |
| 06:38.41 | dshap | thank you both |
| 06:38.50 | dshap | i am adding Asterisk AMI to my list of stuff to read and learn about |
| 06:40.44 | drmessano | Try Dial() first |
| 06:41.00 | drmessano | Just sayin.. |
| 06:41.02 | dshap | and the correct syntax for caller ID in a call file is "CallerID: WhateverName 111-111-1111" |
| 06:41.10 | dshap | ? |
| 06:41.24 | KyleK | Name <Number> i thought |
| 06:42.05 | [TK]D-Fender | dshap: CallerID: "John Doe" <999> |
| 06:42.23 | drmessano | CallerID: Name <number> Caller ID, please note that it may not work if you do not respect the format: CallerID: Some Name <1234> |
| 06:42.48 | drmessano | I found that on Google |
| 06:42.51 | dshap | k got confused since the book im reading had a different format than on voip-info |
| 06:43.02 | dshap | i should probably trust voip-info over anything |
| 06:43.10 | drmessano | Which book? |
| 06:43.13 | KyleK | is it |
| 06:43.17 | KyleK | ~book |
| 06:43.17 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 06:43.23 | dshap | that one |
| 06:43.33 | dshap | Next, we set the Caller ID of the outgoing call: |
| 06:43.33 | dshap | CallerID: Asterisk 800-555-1212 |
| 06:43.37 | dshap | that's what it says in the book |
| 06:45.11 | KyleK | oic |
| 06:45.14 | dshap | but it doesn't seem like the provider i'm using right now allows me to set caller id anyways |
| 06:45.57 | drmessano | Most don't, except for the ones that do |
| 06:46.06 | dshap | lol |
| 06:46.15 | *** join/#asterisk stijnbe (n=stijnbe@d54C16246.access.telenet.be) |
| 06:46.22 | KyleK | what provider? |
| 06:46.29 | dshap | voip.ms |
| 06:46.36 | *** join/#asterisk ghenry (n=ghenry@pdpc/supporter/monthlybyte/ghenry) |
| 06:46.39 | dshap | they have a "value" and a "premium" route |
| 06:46.49 | dshap | i just changed my route to premium which they say caller ID passes better on |
| 06:46.54 | dshap | and they charge u a little more per minute |
| 06:47.02 | dshap | apparently i can change back and forth through their web app |
| 06:47.02 | [TK]D-Fender | dshap: "better"? Crock of shit |
| 06:47.08 | dshap | lol |
| 06:47.39 | dshap | haha |
| 06:47.42 | dshap | but it actually worked |
| 06:47.52 | KyleK | [TK]D-Fender: somehow it happens in practice, skype is occasionally iffy on caller id i thought |
| 06:48.12 | [TK]D-Fender | dshap: It either does, or it doesn't, or should but doesn't and are incompetant to fix. The 3rd is good reason to look elsewhere |
| 06:48.47 | [TK]D-Fender | KyleK: Just for saying "Skype" should get you a whallop with a ClueBat (tm) :) |
| 06:48.56 | KyleK | :o |
| 06:49.02 | dshap | "On our premium route, most if not all destinations will receive proper callerid. On the value route, we can not guarantee callerid will pass but it should on a certain number of destinations" |
| 06:49.15 | drmessano | [TK]D-Fender: Better meaning "We throw your calls at a different group of wholesale routes, and rather than NONE of the routes allowid CID, 20% of the "premium" do.." |
| 06:49.57 | dshap | although "most if not all" is not really good enough for putting something into production |
| 06:50.12 | drmessano | Awesome |
| 06:50.38 | [TK]D-Fender | KyleK: Skype is not PSTN, and to be somewhat less that gentle "That POS bastard-child of 'telephony' has precisely NO respect except from tech wannabe's and idiot kiddies" |
| 06:50.50 | drmessano | Q: Do you allow CallerID info passed on your network? A: Probably |
| 06:50.55 | dshap | lol |
| 06:50.56 | KyleK | hahahahahahaha |
| 06:51.14 | [TK]D-Fender | reaches for his ClueBat (tm) |
| 06:51.18 | *** part/#asterisk elitecoder (n=liq@apollo.bullethost.com) |
| 06:51.29 | [TK]D-Fender | ~clubat SkypeUser |
| 06:51.30 | dshap | drmessano: in case you were wondering, i pretty much gave up on my RDNIS search after some providers e-mailed me back and said "i've looked for this myself, no one supports it" |
| 06:51.33 | [TK]D-Fender | ~cluebat SkypeUser |
| 06:51.34 | infobot | ACTION pulls out a ClueBat (tm) and thwaps SkypeUser. |
| 06:51.42 | dshap | i'm looking into a DID-per-user business model |
| 06:51.53 | drmessano | DID per business? |
| 06:52.01 | dshap | like |
| 06:52.06 | dshap | every user gets their own DID |
| 06:52.16 | SkypeUser | dshap: even les.net? |
| 06:52.16 | dshap | this is what messagesling.com does |
| 06:52.21 | *** join/#asterisk vi390 (n=fc@unaffiliated/vi390) |
| 06:52.24 | SkypeUser | hrm |
| 06:52.30 | dshap | les.net is the guy who told me to give it up |
| 06:52.37 | dshap | i was e-mailing Les himself back and forth |
| 06:52.54 | dshap | he told me RDNIS exists but no VoIP providers support it |
| 06:53.31 | drmessano | dshap: Every user getting their own DID is the OLD model of telephony.. You should google for "PBX" |
| 06:54.04 | dshap | right right, but we've already been over why i need a DID per user |
| 06:54.07 | [TK]D-Fender | drmessano: Not entirely.. its good for direct-access numbers, fax, etc |
| 06:54.49 | dshap | it just only seems affordable if you buy enough of them |
| 06:54.51 | vi390 | does someone have a sample of an outbound extension, where no predial number is necessary (every number dialed is used as outbound) |
| 06:54.54 | drmessano | [TK]D-Fender: He didn't say "DID's for users" He said EVERY |
| 06:55.23 | SkypeUser | dshap: maybe theres a market for RDNIS and VoIP? |
| 06:55.41 | dshap | i doubt a very big one |
| 06:55.51 | dshap | so few search results for RDNIS |
| 06:55.51 | drmessano | Seriously, why did we ever come down from the trees and give up key systems |
| 06:55.58 | dshap | a lot of providers i e-mailed didn't even know what it was |
| 06:56.00 | drmessano | Lets go back to 50 line ATT phones |
| 06:56.23 | drmessano | "Joe, call on Line 30" |
| 06:57.20 | [TK]D-Fender | drmessano: So you can't have every user have their own DID? I did for my company.... 2 in fact :) |
| 06:57.24 | SkypeUser | dshap: well how badly do you want rdnis? :) |
| 06:57.56 | dshap | SkypeUser: for my application, i can acheive the same functionality by assigning each of my users their own DID |
| 06:57.59 | SkypeUser | dances around Fender |
| 06:58.13 | dshap | which would be cheaper than getting a PRI |
| 06:58.18 | dshap | or many PRI's |
| 06:58.20 | KyleK | its more fun being WindowsUser and answering ubuntu questions |
| 06:58.24 | dshap | that would support actual RDNIS |
| 06:58.47 | [TK]D-Fender | dshap: What is you application? |
| 06:59.00 | dshap | same thing as Youmail.com & Messagesling.com |
| 06:59.11 | dshap | mobile voicemail alternative |
| 06:59.27 | dshap | although it's clear that Youmail uses a PRI or something fancy |
| 06:59.31 | dshap | since they have RDNIS |
| 06:59.32 | drmessano | Now you found another service youre trying to be the same as? |
| 06:59.38 | drmessano | Why dont you just SIGN UP |
| 06:59.53 | drmessano | ~nextvonage |
| 06:59.56 | dshap | obviously i have plans to distinguish my service |
| 06:59.56 | drmessano | :( |
| 07:00.05 | dshap | but they'll need the same functionality as those 2 |
| 07:00.14 | dshap | it will need* |
| 07:00.21 | drmessano | dshap: Big plans, and you cant even Dial().. You have messed up priorities |
| 07:00.35 | dshap | my goals are long-term |
| 07:00.48 | dshap | 1st priority is learning how to use Asterisk |
| 07:00.49 | drmessano | Yes, very long term |
| 07:01.05 | dshap | although i'm not going to be doing any rocket science |
| 07:01.27 | KyleK | dshap: the lack of RDNIS might be a software problem, you could pick a voip provider, ask them what equipment they use, add the necessary support and wait for it to be in production :) |
| 07:01.28 | drmessano | You're a Little Prince sort of Asterisk guy |
| 07:01.42 | dshap | i could probably pay an expert to have a functional system up and running in days |
| 07:01.52 | dshap | but then i won't have the satisfaction of having made it myself |
| 07:02.03 | [TK]D-Fender | dshap: And actually have a viable competitive business? |
| 07:02.06 | KyleK | also you'd be dependant on that guy :) |
| 07:02.49 | dshap | [TK]D-Fender: i've started to think about the business model but my first priority is developing the product |
| 07:02.56 | dshap | obviously without a product i can have no business |
| 07:03.20 | KyleK | dshap: make a note of my RDNIS support idea :) |
| 07:03.26 | [TK]D-Fender | dshap: Step one : Spend life working on Project X. Step 2 : Consider if it'll actually be worth it. |
| 07:03.43 | dshap | KyleK: from what i understand, the RDNIS information is provided in normal SS7 signaling. when the upstream carriers convert from PSTN to SIP, they can choose to include a special header with RDNIS info or they can choose not to |
| 07:04.08 | dshap | evidntly they choose not to :-\ |
| 07:04.34 | dshap | [TK]D-Fender: at the very least, I know *I* will get a lot of use out of my product. so i know it will be worth it |
| 07:04.46 | KyleK | dshap: most people are lazy, if asterisk doesn't support it on thier hardware, and thats the PSTN to SIP software they use, they wont bother |
| 07:05.18 | dshap | these major SIP providers use Asterisk to convert from PSTN to SIP? |
| 07:05.23 | drmessano | ~littleprince |
| 07:05.24 | infobot | A Little Prince is a newb who flies in every 6 months, as if brought in by a comet, who asks the same basic "first day" questions, gets the same answers, and flies off again. (Borrowed from the book "The Little Prince" by Antoine de Saint-Exupery) |
| 07:05.26 | KyleK | well i dont know |
| 07:05.40 | dshap | lol |
| 07:06.01 | drmessano | [TK]D-Fender ^^^^^^^^^^^^^^ |
| 07:06.18 | dshap | i managed to find the CTO of YouMail on LinkedIn |
| 07:06.21 | dshap | i sent him a friend request |
| 07:06.34 | dshap | along with a little message about how i want to ask him some questions about his business |
| 07:06.36 | KyleK | What kind of equipment do you use? |
| 07:06.38 | KyleK | Clustered SER/Asterisk |
| 07:07.14 | dshap | so i just need to get the asterisk dev community to make SIP diversion headers a default and once everyone upgrades i'm good |
| 07:07.16 | dshap | to go |
| 07:07.19 | [TK]D-Fender | dshap: What about their service implies any use of VoIP? |
| 07:07.43 | [TK]D-Fender | dshap: What does * have to do with that? |
| 07:07.44 | dshap | [TK]D-Fender: nothing at all. |
| 07:08.02 | drmessano | <dshap> so i just need to get the asterisk dev community to make SIP diversion headers a default and once everyone upgrades i'm good <----- WHAT???? |
| 07:08.05 | dshap | but im curious how it works |
| 07:08.24 | dshap | well that was in response to what KyleK said |
| 07:08.25 | [TK]D-Fender | dshap: You can't make THEM send headers they don't want to. Doesn't matter in the slightest what * supports whent he other guy decides not to both |
| 07:08.29 | [TK]D-Fender | +er |
| 07:08.58 | dshap | KyleK made it sound like they aren't going to go out of their way to send the headers |
| 07:09.08 | dshap | but if it's built into Asterisk by default, then why wouldn't they? |
| 07:09.08 | [TK]D-Fender | dshap: And * can pull whatever retarded header you want anyways. |
| 07:09.24 | [TK]D-Fender | dshap: Because they don't give a flying fuck about Asterisk? |
| 07:09.42 | [TK]D-Fender | dshap: You don'[t seem to understand the order of things. |
| 07:09.53 | carrar | watches out for flying stuff |
| 07:09.53 | dshap | he said they USE asterisk to convert PSTN to SIP |
| 07:10.27 | dshap | the major VoIP providers |
| 07:10.33 | dshap | nevermind |
| 07:10.34 | carrar | heh |
| 07:10.35 | KyleK | well, voip.ms says they do, dunno about anything beyond that |
| 07:10.37 | dshap | it's a moot point really |
| 07:10.46 | carrar | w00t! |
| 07:10.55 | dshap | pretty sure voip.ms is a reseller, KyleK |
| 07:11.01 | KyleK | ah |
| 07:11.14 | dshap | they can't send me headers if their upstream carrier doesn't send THEM the headers |
| 07:11.14 | drmessano | So youre expecting everyone implementing Asterisk to configure things the way YOU want? |
| 07:11.20 | dshap | nooooo |
| 07:11.21 | dshap | jeez |
| 07:11.22 | dshap | alright |
| 07:11.23 | dshap | forgetit |
| 07:11.25 | dshap | sorry i mentioned it |
| 07:11.26 | KyleK | god fucking damnit |
| 07:11.26 | dshap | haha |
| 07:12.02 | dshap | i'm stoked about my call files working, i think im gonna call it a night and quit while i'm ahead |
| 07:12.06 | drmessano | <dshap> so i just need to get the asterisk dev community to make SIP diversion headers a default and once everyone upgrades i'm good <-- Fuck you, I dont WANT Diversion set.. but if I choose to, then good for me |
| 07:12.07 | [TK]D-Fender | dshap: if they use * to actually convert to PSTN that implies they use Zaptel/DAHDI and have access to RDNIS and are also capable of setting custom headers |
| 07:12.09 | jql | I love me some proprietary headers |
| 07:12.29 | dshap | i get it |
| 07:12.33 | dshap | they don't want to send the headers |
| 07:12.35 | dshap | they can |
| 07:12.37 | dshap | but they dont want to |
| 07:12.41 | dshap | but i want them to |
| 07:12.41 | drmessano | I think all of you should be using DNS SRV records |
| 07:12.44 | dshap | and therefore i am a douchebag |
| 07:12.49 | drmessano | Matter of fact, I demand it |
| 07:12.57 | [TK]D-Fender | dshap: dspOr they are lying to you to tell you what you want to hear. |
| 07:12.58 | dshap | im not demanding anything haha |
| 07:13.08 | dshap | i'm just saying that it would be nice to have for my application |
| 07:13.17 | dshap | since it would prevent me from having to get a ton of DID's |
| 07:13.18 | KyleK | [TK]D-Fender, drmessano: I threw out the idea that maybe if people are using asterisk to go from PSTN to VoIP, checking up on RDNIS support for different hardware would be an idea |
| 07:13.20 | jql | most everyone is getting their PSTN access via Sonus, and I *know* Sonus offers the awesomeness of Diversion: |
| 07:13.27 | [TK]D-Fender | dshap: No, your being a douchebag is a functionally independent trait ;) |
| 07:13.39 | dshap | hahah |
| 07:13.51 | KyleK | i hate it how you both take anything i say and iterate until you find a problem |
| 07:14.09 | dshap | this channel rocks |
| 07:14.12 | dshap | i'll be back another time |
| 07:14.14 | dshap | good night |
| 07:14.21 | *** part/#asterisk dshap (n=IceChat7@ip70-181-91-110.oc.oc.cox.net) |
| 07:14.33 | [TK]D-Fender | KyleK: Yes, but its a considerably quicker path than "6 Degrees Of Separation" :p |
| 07:14.43 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 07:40.10 | *** join/#asterisk Defraz (n=T0tal@24-117-236-174.cpe.cableone.net) |
| 07:42.12 | pawz | i have my asterisk box on my lan but in a DMZ. i can successfully register to an extension from outside on the internet, but when i make calls from outside i get no audio in either direction. what's likely to be wrong ? |
| 07:42.58 | [TK]D-Fender | pawz: the complete lack of all of the actually settings you need to do. |
| 07:43.02 | [TK]D-Fender | ~sipnat |
| 07:43.02 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 07:43.04 | [TK]D-Fender | ^^^^ |
| 07:43.11 | pawz | cool thanks i'll read that |
| 07:48.40 | pawz | wow, that was easy :") it works great now |
| 07:49.10 | ectospasm | /ectospasm |
| 07:53.35 | [TK]D-Fender | pawz: You're welcome |
| 07:54.32 | vi390 | can there maybe someone help. I think I did not quite get the conzept of outbound connection. Here is what I have => http://pastebin.com/d4c9558a4 |
| 07:55.22 | vi390 | I want that any outbound connection , hmm the best would be, any connection which can not be catched localy, should be dialed outbound. How can I do this? |
| 07:56.29 | ectospasm | heheh |
| 07:57.10 | *** join/#asterisk war9407 (i=war@liquidswords.org) |
| 07:57.55 | ectospasm | vi390: use dialplan logic to determine if this call can be caught locally. If not, send it to wherever with a Dial() string. Of course, Asterisk will still be in the middle of a call. |
| 07:58.57 | vi390 | ectospasm: you mean in the internal_call part of extensions.conf after the local handling ? |
| 07:59.27 | ectospasm | vi390: that's how it generally should work. I haven't looked at your dialplan |
| 07:59.28 | [TK]D-Fender | vi390: because your [200] peer says "context=main" and [main] only includes [internal_call] and that only contains exten => _[1-5]XX,1,Dial(SIP/${EXTEN},60) |
| 08:00.37 | vi390 | [TK]D-Fender: aah, okey this gives me the hint.. thanks |
| 08:02.13 | vi390 | so I just put something after the main internal. But how can I determin if the dail can be done internally or not. Does it automtically execute things after exten => _[1-5]XX,1,Dial(SIP/${EXTEN},60) if they can not be caught by that ? |
| 08:02.50 | ectospasm | vi390: you'll need a catchall, like _X. or i |
| 08:02.53 | [TK]D-Fender | vi390: there is no such thing as "dial internally" Your device can only dial extens you point it to |
| 08:03.28 | [TK]D-Fender | ectospasm: No. |
| 08:03.47 | [TK]D-Fender | vi390: If you wanted them to be able to dial more things, then INCLUDE THEM |
| 08:04.43 | vi390 | [TK]D-Fender: hmm, yes. Ok shure. What I meant is. If something can not be dialed, because there is no attached number 300, will it automticaly jump to the next entry in the line? |
| 08:05.03 | ectospasm | vi390: it will jump to the i extension, if one exists |
| 08:05.10 | [TK]D-Fender | vi390: No. |
| 08:05.21 | [TK]D-Fender | ectospasm: SIP calls will not hit "i" |
| 08:05.42 | ectospasm | news to me |
| 08:05.53 | vi390 | hmm I think I do not get the part with the "i" extension |
| 08:06.15 | vi390 | shure is , that I need sort of a "catchall" |
| 08:06.33 | vi390 | where it can Fallback, if numbers are not dailable internaly |
| 08:06.49 | ectospasm | _X., will catch any number not caught by what's above |
| 08:07.16 | vi390 | ectospasm: ok, doing a test with that :) |
| 08:07.22 | ectospasm | And you can take whatever they dialed, and send it out any trunk with Dial() |
| 08:07.51 | vi390 | if so, it would be easy to do what I need :) |
| 08:08.12 | ectospasm | [TK]D-Fender: I'd still like to hear your explanation for why SIP calls don't hit "i" |
| 08:08.36 | [TK]D-Fender | ectospasm: 404 <- |
| 08:08.44 | vi390 | hmm, And I would be intrested what "i" means |
| 08:08.48 | [TK]D-Fender | ectospasm: Feel free to show me otherwise |
| 08:09.05 | [TK]D-Fender | vi390: Go read *'s list of Standard Extensions. This is dialplan 101 |
| 08:09.26 | ectospasm | vi390: "invalid" |
| 08:09.37 | [TK]D-Fender | ectospasm: "assumed functioning" |
| 08:09.52 | vi390 | [TK]D-Fender: where do I find that, sounds I really want to read that |
| 08:10.00 | ectospasm | !thebook |
| 08:10.01 | [TK]D-Fender | ~wikis |
| 08:10.02 | infobot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 08:10.07 | vi390 | k |
| 08:10.20 | ectospasm | ~thebook |
| 08:10.20 | infobot | [thebook] 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 |
| 08:11.04 | vi390 | thanks |
| 08:26.32 | *** join/#asterisk canburak (n=can@85.100.120.41) |
| 08:26.47 | *** join/#asterisk ttl- (n=patrick@81.83.174.130) |
| 08:28.10 | canburak | hi, I am trying to setup audiocodes such that I would be able to route to landlines based on extensions, which route should I follow? I can do roundrobin routing, ignoring the extensions but I want to route outgoing calls based on extensons. I would appreciate any pointers |
| 08:31.12 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 08:33.29 | *** join/#asterisk l2trace99 (n=jr@rrcs-71-43-104-238.se.biz.rr.com) |
| 08:41.20 | *** join/#asterisk jgoo (n=r3rman@ppp15-9.adsl.forthnet.gr) |
| 08:43.34 | jgoo | hi, using asterisk 1.4.22-2 and call pickup doesn't work, is this a known version? pickup groups are set to 1, and *8 is enabled in the application features page (freepbx) |
| 08:45.15 | sfire | jgoo, wrong time of day to ask... hhehehhe |
| 08:53.29 | *** join/#asterisk plq (n=plq@85.96.253.182) |
| 08:53.44 | jgoo | Yeah, I figured. Damn you all .GMT is the most awesome timezone |
| 08:53.54 | jgoo | you should all adhere to it, or perish. |
| 08:53.55 | jgoo | lol |
| 08:54.09 | pawz | swatch internet time ftw ! |
| 08:54.10 | jplank | technically we all do |
| 08:54.13 | pawz | j/k not really |
| 08:54.30 | jplank | I live by GMT -5 :) |
| 08:55.12 | plq | quoting http://en.wikipedia.org/wiki/Double_dispatch: "while virtual functions are dispatched dynamically in C++, function overloading is done statically." my question is: why? |
| 08:55.30 | plq | aargh |
| 08:55.31 | plq | sorry |
| 08:55.31 | *** part/#asterisk plq (n=plq@85.96.253.182) |
| 09:00.30 | *** join/#asterisk stijnbe (n=stijnbe@d54C16246.access.telenet.be) |
| 09:02.42 | pawz | i want to create a dial rule that matches all numbers except an 04 prefix. I'm trying [0][1-3,5-9]. but it says it's not valid. how should i write this instead ? |
| 09:13.00 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 09:15.01 | *** join/#asterisk tokozedg (n=CCNA@89.232.24.53) |
| 09:25.10 | Kevin` | has anyone used a sound card as an fxo port? is there an easy (software) interface I can use for ringing, hangup, hook control? |
| 09:27.29 | *** join/#asterisk twanny796 (n=chatzill@85.232.206.65) |
| 09:31.31 | drmessano | You cant use a sound card for an FXO port |
| 09:31.44 | Kevin` | says whowhy? |
| 09:32.09 | *** join/#asterisk Milad (n=milad@unaffiliated/slackark) |
| 09:32.27 | *** join/#asterisk farkus_ (i=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 09:32.32 | drmessano | There really needs to be a wiki page on hardware and why shit wont work |
| 09:32.54 | Kevin` | humor me? |
| 09:32.54 | drmessano | Rather than the weekly three hour "Why doesnt my video card make a good FXS" convo |
| 09:33.39 | Milad | is any way queue_member_count just count available member in queue ? |
| 09:34.43 | Kevin` | drmessano: http://www.epanorama.net/circuits/teleinterface.html - some reference, THEN humor me? |
| 09:37.10 | Kevin` | my time isn't worth $200/hour, and I don't have any more money left to blow on interface hardware this month =p |
| 09:37.45 | tokozedg | fxo is used to plug analog rj-11 line in it comming from PSTN |
| 09:38.09 | tokozedg | and can you plug RJ-11 in your sound card? |
| 09:38.32 | Kevin` | maybe I should ask in ##electronics, but I doubt many people there know the asterisk software interface :( |
| 09:40.10 | drmessano | $90 for a single port FXO card that works without software being written |
| 09:40.22 | *** join/#asterisk farkus_ (i=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 09:40.44 | Kevin` | $80 for fxs AND fxo via ip that works without software bieng written |
| 09:40.52 | Kevin` | as I said, no money left this month =p |
| 09:41.13 | drmessano | $80 for what? |
| 09:41.58 | drmessano | An SPA-3102? |
| 09:42.12 | Kevin` | ah yeah, that |
| 09:42.42 | drmessano | You sure do youve your arguments well planned |
| 09:42.46 | drmessano | gah |
| 09:42.54 | drmessano | Brain melt.. |
| 09:43.05 | Kevin` | haha |
| 09:44.54 | Kevin` | how is the quality of that vs the cheapish supported fxo cards btw? |
| 09:45.20 | drmessano | The cheapish cards are complete and total crap |
| 09:45.30 | drmessano | Not even worth using or mentioning |
| 09:48.09 | *** join/#asterisk farkus_ (i=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 09:48.14 | Kevin` | http://cgi.ebay.com/Openvox-A400P-1FXS-FXO-Digium-Asterisk-Trixbox-TDM400_W0QQitemZ300318630949QQihZ020QQcategoryZ61841QQssPageNameZWDVWQQrdZ1QQcmdZViewItem - this kind |
| 09:48.36 | drmessano | I just told you about it 10 mins ago |
| 09:48.45 | drmessano | [05:40] <drmessano> $90 for a single port FXO card that works without software being written |
| 09:49.07 | Kevin` | so those type of cards you would recommend, because you did mention them? |
| 09:49.14 | Kevin` | ;) |
| 09:49.34 | drmessano | They work.. Not sure about heavy load production use |
| 09:49.44 | jgoo | using asterisk 1.4.22-2 and call pickup doesn't work, is this a known problem? pickup groups are set to 1, and *8 is enabled in the application features page (freepbx) |
| 09:50.24 | Kevin` | how do they perform vs one of those linksys devices |
| 09:51.12 | drmessano | I would say better |
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| 09:51.49 | Kevin` | hmm. |
| 09:51.50 | Kevin` | # /etc/init.d/dahdi restart |
| 09:51.50 | Kevin` | /etc/init.d/dahdi: line 50: /etc/init.d/functions: No such file or directory |
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| 10:09.31 | IPGHOST | hi |
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| 10:12.16 | jgoo | hey - so, call pickup - what is the problem with it? I swear, you google it and you find just pages of people with problems and no solutions |
| 10:12.29 | jgoo | asterisk 1.4.22-2 |
| 10:13.31 | jgoo | which asterisk live cd is best, trix, now or that other one, what is it called? |
| 10:13.35 | jgoo | elastix? |
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| 10:15.43 | jgoo | anyone using call pickup? *8 or something similar? what is it working on? |
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| 11:26.16 | master_of_master | hi, how can I dial from the CLI to a certain number (via ISDN@capi)? |
| 11:27.49 | Kevin` | channel originate proto/channel application dial whatever |
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| 11:42.38 | Simon- | is anyone use res_jabber with a large roster? I'm getting XML parse errors reading the roster with 249 contacts |
| 11:44.19 | Kevin` | http://pastebin.ca/1440920 - asterisk doesn't like me doing this, although it appears to work so far. why is it discouraged? what's an equivalent? (X doesn't match letters) |
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| 12:14.23 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 12:36.21 | Yurik | hi, anybody uses asterisk 1.6.2 beta? |
| 12:36.39 | leifmadsen | Yurik: I was just using the 1.6.2 branch yesterday in development |
| 12:36.49 | Yurik | how stable is it? |
| 12:36.55 | leifmadsen | Yurik: it's in beta... |
| 12:37.01 | Yurik | true.. |
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| 12:37.23 | Yurik | just needs siren14 very much, for some reason |
| 12:37.25 | leifmadsen | expect to find at least one issue, and the need to report it and help debug and find the issue |
| 12:37.46 | leifmadsen | Yurik: all you can do is give it a shot and test the heck out of all the things you need it to do |
| 12:37.59 | Yurik | not in production, though :) |
| 12:38.15 | leifmadsen | well ya... you need to test prior to production |
| 12:38.21 | leifmadsen | just like all applications |
| 12:38.23 | Yurik | it's scary as hell :) |
| 12:38.41 | Yurik | wishes there was a backport of siren14, but that's only dreams :D |
| 12:38.43 | leifmadsen | it shouldn't be IF YOU TEST |
| 12:39.01 | leifmadsen | and disable everything you don't need |
| 12:40.07 | Yurik | is thinking, whether there is any other way to get siren14... hmm |
| 12:40.31 | *** join/#asterisk hellc2 (n=hellc2@85.137.126.138.dyn.user.ono.com) |
| 12:44.22 | vi390 | Iam trying since hours now to configure outbound Sip to Sip. I allways get "Failed to authenticate on INVITE" My Provider is "sipgate" is there any weird setting, Or handling, that I dont know? I tried with so many possible configs now, and it seems Iam sort of stucked. Can Someone help out? |
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| 12:45.32 | vi390 | I send the CallerId before the Dial with: exten => _X.,1,SET(CALLERID(all)=sipID) |
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| 12:47.36 | *** part/#asterisk Simon- (i=simon@proxima.lp0.eu) |
| 12:47.51 | IPGHOST | hi |
| 12:48.05 | IPGHOST | any one can help me on database CFIM ? |
| 12:48.23 | IPGHOST | how i can forward IAX directly to next asterisk box |
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| 13:12.13 | *** join/#asterisk prxtien (n=pro@ppp121-45-110-243.lns10.adl6.internode.on.net) |
| 13:12.18 | prxtien | hey all, hope everyone is well |
| 13:12.37 | prxtien | in 1.6.0.X how do i identify what codec both parties negociate to use? |
| 13:14.08 | kaldemar | either from the call setup debug or by core show channel <channel> |
| 13:17.28 | prxtien | okay thanks |
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| 13:21.00 | prxtien | when i dial into my system from a mobile, i am getting alot of bad static 50% of the time, the other 50% i am getting perfect quality... can maybe anyone suggest where to start troubleshooting |
| 13:21.41 | prxtien | people ringing me report this bad quality aswell, but i never hear it on my end, its only if i dial into my system myself... my incoming calls are coming via my isps pstn>sip gateway and then into my asterisk box.. |
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| 13:22.05 | Yurik | is there any way to switch telephone-event payload from a specific call? |
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| 13:51.13 | Yurik | ok, even more generic question — is there anyway to control media attributes in rtp in asterisk? |
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| 13:58.28 | DelphiWorld | hello |
| 13:59.03 | leifmadsen | Yurik: what you're looking to do is more a function of a proxy server, and not asterisk |
| 13:59.22 | Yurik | or some hacking in chan_sip.. :) |
| 14:00.44 | oej | yurik: Media attributes - which one do you mean? |
| 14:00.55 | oej | You can control codecs and packetization |
| 14:01.03 | Yurik | I need to inject custom media attribtues into INVITE |
| 14:01.09 | oej | Like? |
| 14:01.16 | oej | Just curious |
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| 14:01.38 | Yurik | there is some sip proxy that seemingly does not work without them |
| 14:01.39 | oej | We have no setting for telephone-event currently, but that should be easy |
| 14:01.47 | oej | And no, leif, a proxy is not involved in media :-) |
| 14:02.06 | oej | A Sip proxy that bothers with SDP? That's weird. |
| 14:02.10 | leifmadsen | oej: unless your proxy can handle it through a media proxy :) |
| 14:02.12 | Yurik | right |
| 14:02.21 | *** join/#asterisk codefreeze-lap (n=murf@72.21.67.40) |
| 14:02.27 | oej | leifmadsen: That's a media proxy then, not a SIP proxy... |
| 14:02.33 | Yurik | so I am hacking chan_sip to inject those attrs |
| 14:02.47 | oej | Ok, so you're solving your problem. The power of Open Source |
| 14:02.48 | leifmadsen | ugh.... TFoT needs a complete re-write... |
| 14:02.53 | Yurik | and as far as I can understand there is no easy way to do it in another way |
| 14:03.03 | oej | leifmadsen: WHy? Spanish? |
| 14:03.13 | leifmadsen | oej: too many things have changed since 1.4.0 |
| 14:03.16 | oej | yurik: No, there has been no demands for it up to now |
| 14:03.26 | Yurik | I am creating a person demand :-P |
| 14:03.26 | leifmadsen | I'm reading through chapter 3 to try and update it, and this is really quite fruitless |
| 14:03.37 | oej | leifmadsen: yes, and which version will you cover now? 1.6.0, 1.6.1 and 1.6.2 are all very different |
| 14:03.59 | oej | This release plan creates a mess for training material, support and books |
| 14:04.27 | leifmadsen | what is the alternative? wait several years for new features? |
| 14:04.36 | leifmadsen | I'm in favour of it |
| 14:04.47 | leifmadsen | even if it will cause books to be constantly updated |
| 14:05.13 | oej | The alternative is locking the core and merging new functions, but not changing stuff between releases |
| 14:05.35 | oej | And allowing changing stuff between "major" releases |
| 14:05.48 | oej | Asterisk depends more and more on a third party market that we need to acknowledge |
| 14:06.06 | oej | But I've been fighting this fruitlessly for far too long ;-) |
| 14:06.48 | oej | The Asterisk project lacks a product manager... |
| 14:06.54 | leifmadsen | I'm not getting into this |
| 14:07.01 | oej | ;-) |
| 14:07.18 | leifmadsen | decides to put down the book and work on dialplan for a customer instead |
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| 14:07.36 | oej | I thought you where working fully for Digium now, Leif |
| 14:09.12 | [TK]D-Fender | oej: Doesn't need one really :) |
| 14:09.35 | oej | need what? A Leif? |
| 14:09.37 | leifmadsen | oej: no, I always have been, and always will be a consultant |
| 14:09.44 | [TK]D-Fender | oej: See Digium really sells hardware and Asterisk is more like a marketing strategy :) |
| 14:10.02 | oej | leifmadsen: Sorry, my misunderstanding then. Good to know. I subcontract more and more. |
| 14:10.24 | leifmadsen | oej: I've been subcontracting out to other consultants lately because they keep coming to me to try and get work |
| 14:10.35 | oej | good! |
| 14:10.37 | [TK]D-Fender | oej: So what they're really doing is "marketing to agile (bleeding-edge) users"! |
| 14:10.57 | oej | I can't agree with what you're saying about Digium. |
| 14:11.09 | oej | There's more and more services and software, look at Switchvox. |
| 14:11.22 | *** part/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:11.27 | oej | And they're investing heavily in Asterisk |
| 14:13.00 | [TK]D-Fender | oej: Yes, Switchvox also uses * and lets them sell servers & cards... all around their software :) |
| 14:13.53 | [TK]D-Fender | oej: GUI users often don't know or care so much about whats underneath since the GUI tells them what they can and can't do. |
| 14:14.21 | [TK]D-Fender | oej: "I don't wannt code, why can't I mjust make it do XYZ?!" |
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| 14:18.24 | *** join/#asterisk rjune_ (n=rjune@38.103.117.250) |
| 14:18.31 | [TK]D-Fender | oej: IIRC Switchvox systems lock you out of just about everything else on the box. |
| 14:18.49 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 14:18.59 | oej | I see that you have a lot of opinions... :-) |
| 14:19.20 | oej | I still don't think that Digium is only targeting the hardware market |
| 14:19.32 | [TK]D-Fender | oej: Yup.. everybody is entitled to my own opinion ;) |
| 14:19.41 | oej | he he |
| 14:20.15 | [TK]D-Fender | oej: Consider how much the ABE division factors into their net profits. |
| 14:21.20 | [TK]D-Fender | oej: They do seem to be a common OSS business model. Giving away free milk to sell milk pitchers. |
| 14:21.22 | *** join/#asterisk farkus_ (i=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 14:21.45 | oej | And I earn money on selling their free milk :-) |
| 14:23.15 | rjune_ | isn't it more you earn money delivering their free milk? |
| 14:23.44 | [TK]D-Fender | rjune_: I'd say more like for "pasteurizing" ;) |
| 14:26.17 | rjune_ | I'm having voicemail issues, I have them configured, but nothing happens. http://pastebin.ca/1441060 I *THINK* it's voicemail has not been setup. |
| 14:27.23 | *** join/#asterisk farkus_ (i=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 14:28.34 | [TK]D-Fender | rjune_: pastebin your voicemail.conf |
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| 14:31.02 | rjune_ | http://pastebin.ca/1441067 |
| 14:31.24 | [TK]D-Fender | rjune_: .... AND the files it INCLUDES.. |
| 14:32.22 | [TK]D-Fender | rjune_: nvm.. whats that [] before your actual entry? that KILLS [default] |
| 14:32.36 | [TK]D-Fender | rjune_: Which is the origin of your failure |
| 14:32.51 | rjune_ | Ahhh |
| 14:33.00 | rjune_ | I pasted them all. :-) |
| 14:33.06 | rjune_ | thanks, I'll check into that |
| 14:33.22 | [TK]D-Fender | rjune_: Took me a second to notice the order was reversed |
| 14:33.23 | *** join/#asterisk farkus_ (i=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 14:34.33 | rjune_ | Odd. |
| 14:35.47 | rjune_ | iusing freepbx, if I putdefault into the voicemail context, it generates [], otherwise, it generates the proper context |
| 14:38.53 | rjune_ | what does [] actually indicate? |
| 14:39.07 | [TK]D-Fender | rjune_: another vm context |
| 14:39.18 | [TK]D-Fender | rjune_: If you actually put something in it. |
| 14:40.07 | rjune_ | voicmeail.conf is where all of that is located, correct? |
| 14:40.41 | [TK]D-Fender | rjune_: Normally. As you see with FreePBX, they try to break it into separate files and "include" them |
| 14:41.05 | rjune_ | which is fine, I'm just trying to understand why what was done got done |
| 14:41.16 | rjune_ | and #freepbx is silent |
| 14:41.23 | *** join/#asterisk farkus_ (i=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 14:42.10 | rjune_ | but when I change the config file, it works properly |
| 14:42.59 | [TK]D-Fender | rjune_: Feel free to complain to them :) |
| 14:44.57 | *** join/#asterisk smultron (n=smultron@cpe-67-9-150-163.austin.res.rr.com) |
| 14:47.27 | rjune_ | [TK]D-Fender, You make it sound futiel |
| 14:47.30 | rjune_ | futile even |
| 14:48.32 | [TK]D-Fender | rjune_: Well pretty much the rest of FreePBX users don't seem to have your problem so I'll give a fair bet that you've done something wrong with it |
| 14:48.56 | rjune_ | I'm quite sure I have. |
| 14:49.05 | rjune_ | don't know how it happened, but at least I know how to fix it |
| 14:49.13 | rjune_ | I removed [] manually, and presto chango, it works |
| 14:49.23 | *** join/#asterisk farkus_ (i=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 14:52.55 | Yurik | okay, not meda attributes.. |
| 14:53.09 | Yurik | can I change headers without hacking? I need to add Subject: |
| 14:53.10 | Yurik | :) |
| 14:53.40 | l2trace99 | does anyone have issues with calls created via an originate not showing up in the logs ? |
| 14:54.12 | [TK]D-Fender | Yurik: "core show functions like SIP" <- |
| 14:54.26 | Yurik | thanks! |
| 14:55.00 | Yurik | Set(SIP_HEADER()=..) is supposed to work, I guess? |
| 14:56.58 | [TK]D-Fender | Yurik: Function is there for a reason... |
| 14:57.23 | *** join/#asterisk farkus_ (i=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 14:58.47 | l2trace99 | or even when where is just a hi volume of calls not hitting the logs |
| 14:58.48 | Yurik | Contrary to previously claimed,SIP_HEADER is read-only. |
| 14:58.49 | Yurik | This example does "not" work! |
| 14:58.49 | Yurik | <PROTECTED> |
| 14:58.52 | Yurik | *sigh* |
| 14:59.16 | Yurik | http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+SIPAddHeader |
| 14:59.18 | Yurik | here we go |
| 14:59.19 | [TK]D-Fender | Yurik: Show me a complete attempt |
| 14:59.34 | [TK]D-Fender | Yurik: Yup, that app too |
| 15:00.02 | *** join/#asterisk chandoo (n=chandoo@ool-4353b978.dyn.optonline.net) |
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| 15:06.03 | Yurik | sigh, playing with custom blackbox sip proxy is exhausting |
| 15:13.22 | *** join/#asterisk farkus_ (i=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 15:13.25 | Yurik | I will dream about 400 Bad SDP -- unsupported payload type tonight |
| 15:13.30 | Yurik | err, this morning |
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| 15:28.25 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 15:30.13 | leifmadsen | oej: quick question -- are 'urgent' messages kept in an 'urgent' folder? (i.e. I get their count with ${VMCOUNT(urgent@context)} |
| 15:31.23 | *** join/#asterisk farkus_ (i=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 15:34.18 | rjune_ | How do I assign an extension to dial an external number such as a cell phone? |
| 15:35.04 | *** join/#asterisk saftsack (n=saftsack@p57924F36.dip.t-dialin.net) |
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| 15:41.17 | leifmadsen | rjune_: exten => 9999,1,Dial(SIP/myProvider/18002223334) |
| 15:41.30 | leifmadsen | ~book |
| 15:41.31 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 15:42.08 | rjune_ | I have it, haven't read it all the way. |
| 15:42.17 | leifmadsen | you need to read about dialplans |
| 15:42.47 | leifmadsen | your question was one of the most basic of functionalities in the dialplan |
| 15:44.08 | [TK]D-Fender | leifmadsen: And he has to get his ass out of GUI-Land :) |
| 15:44.17 | [TK]D-Fender | (FreePBX) |
| 15:44.25 | leifmadsen | there is nothing wrong with GUI land |
| 15:44.27 | leifmadsen | they have their place |
| 15:44.53 | leifmadsen | however, the dialplan is ridiculously more powerful without the GUI :) |
| 15:45.07 | *** join/#asterisk kaptengu (n=kaptengu@unaffiliated/kaptengu) |
| 15:45.16 | [TK]D-Fender | leifmadsen: Yes, I believe Dante describe their circle in Hell as being just below that of the one reserved for child molesterers and people who talk in theaters ;) |
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| 15:49.37 | DelphiWorld | hello |
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| 16:49.51 | leifmadsen | farkus: you need to fix your IRC client |
| 16:49.55 | leifmadsen | farkus: or network connection |
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| 17:00.27 | srf21c | I'm looking for some inexpensive DID and custom toll free number termination on a pay as you go basis. So far I've identified Vitelity and Teliax as front runners. |
| 17:00.45 | srf21c | Teliax has DIDs for $5/mo. Vitelity $1.50/mo. |
| 17:00.46 | ManxPower | srf21c: Those would be my picks |
| 17:01.00 | srf21c | Vitelity seems to have lower in and outbound rates as well. |
| 17:01.18 | srf21c | ManxPower: ok, thx. Hopefully I'm on the right track then. |
| 17:01.44 | srf21c | I did check out ipcomms.net, but they have a $10/mo minimum. |
| 17:02.23 | *** join/#asterisk farkus_ (i=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 17:02.52 | srf21c | Vitelity however has a $35 minimun, whereas $10 gets you "in the door" at Teliax. |
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| 17:12.50 | oej | leifmadsen: Sorry, was out in the garden |
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| 17:45.39 | FSB_1 | Anyone who knows how to reset Cisco ATA 188 if you don't hear the IVR-prompt? |
| 17:46.35 | tzafrir_laptop | there are simple ways to do that. But do you want to it to still be functional afterwards? |
| 17:47.02 | FSB_1 | Offcourse |
| 17:47.22 | tzafrir_laptop | I guess that application of the hammer method will not do, then |
| 17:47.29 | FSB_1 | Probably not |
| 17:48.08 | mattbUK | Anyone on here use voiptalk and can tell me what IP ranges their IAX2 service operates on - their website doesn't seem to say and I want to add rules to my firewall |
| 17:48.10 | tzafrir_laptop | do you get a web interface? |
| 17:48.24 | tzafrir_laptop | that was for FSB_1 |
| 17:48.34 | FSB_1 | tzafrir_laptop: Yup, locked down though. |
| 17:49.23 | *** join/#asterisk farkus_ (i=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 17:49.49 | mattbUK | Does anyone here use a UK IAX2 service that allows connections over openVPN or similar? |
| 17:49.51 | tzafrir_laptop | suspecst farkus isn't really here |
| 17:50.13 | tzafrir_laptop | why would you need openvpn with iax2? |
| 17:50.27 | tzafrir_laptop | are there places that block it? |
| 17:51.04 | ManxPower | tzafrir_laptop: Maybe his conversations are so important that someone would spend the required time to eavesdrop |
| 17:51.32 | mattbUK | tzafrir_laptop: as ManxPower says |
| 17:51.57 | tzafrir_laptop | there's actually encrypted iax2. Not sure if it leaves the call setup unencrypted |
| 17:52.14 | ManxPower | mattbUK: Use burner cell phones, it's much easier and cheaper |
| 17:52.41 | ManxPower | tzafrir_laptop: I don't think I'd trust any encryption Digium wrote 8-| |
| 17:52.49 | mattbUK | tzafrir_laptop: I was looking at that just now - I couldn't decide if it was up to the task |
| 17:53.01 | tzafrir_laptop | ManxPower, it's an IETF standard |
| 17:53.31 | tzafrir_laptop | (which, IIRC, originally differed slightly from what was implemented in Asterisk...) |
| 17:53.39 | ManxPower | tzafrir_laptop: Look at the history of security bugs in Asterisk |
| 17:53.46 | mattbUK | ManxPower: we're looking (prototype) outbound payment collection calls so ideally needed to IVR it but I'm not convinced asterisk is properly up to the job) |
| 17:54.08 | leifmadsen | oej: no problem |
| 17:54.50 | tzafrir_laptop | btw: which other programs implement IAX2 encryption? IAX2 rsa key authentication? |
| 17:55.22 | mattbUK | ManxPower: the history is exactly the reason I'm a bit reluctant - but if it were VPN'd and the firewall locked down to a specific inbound service provider IP range I would be (slightly) more confident that we're not wasting out time even looking at this |
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| 17:59.14 | ManxPower | mattbUK: if you are trying to encrypt collection calls you are already wasting your time. |
| 18:00.00 | ManxPower | People worry so much about VoIP security and then totally ignore the box outside the building where anyone could tap into the lines. |
| 18:00.33 | ManxPower | ("the lines" == traditional telephony lines) |
| 18:00.48 | coppice | nobody worries about VoIP security. that's why TLS for SIP and SRTP have taken so long to start moving |
| 18:01.01 | ManxPower | coppice: nobody but mattbUK 8-) |
| 18:02.02 | mattbUK | ManxPower: very true, but the payment card industry requires we have to at least make sure our gear is secure - to be honest I wish we'd never even got started looking into this - it's a total waste of resource and we should just out source it to someone who knows better - but management is management |
| 18:02.24 | *** join/#asterisk farkus_ (i=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 18:03.29 | jql | meh, PCI compliance |
| 18:04.05 | drmessano | We're not even ISA compliant |
| 18:04.41 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 18:05.36 | srf21c | these guys offer encrypted voip termination http://www.rayservers.com/privacy |
| 18:06.02 | srf21c | mattbUK: maybe you could outsource it to them. |
| 18:06.17 | mattbUK | srf21c: cheers I'll take a look |
| 18:06.35 | srf21c | I believe they are using OpenVPN and IPSec to secure the calls. |
| 18:06.43 | srf21c | not IAX. If that matters. |
| 18:07.22 | mattbUK | srd21c: that's all good - thanks for digging them up |
| 18:08.42 | srf21c | mattbUK: np, hope it helps. |
| 18:10.30 | *** join/#asterisk farkus_ (i=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 18:11.30 | srf21c | btw, anyone here happen to know what the three most popular asterisk/voip web forums are? I'm new to the scene trying to find out where the action is. |
| 18:12.13 | srf21c | This channel is a nice resource, but I'm also looking for a less ephemeral source of asterisk info. |
| 18:13.04 | KyleK | asterisk-users mailing list? |
| 18:13.08 | KyleK | ~mailinglist |
| 18:13.09 | infobot | [~mailinglist] The Asterisk mailing lists can be found at http://lists.digium.com , http://www.asteriskguru.com/archives Search the archives by adding "site:lists.digium.com" to your Google search. |
| 18:13.21 | jql | voip-info wiki, digium forums, asteriskguru |
| 18:13.46 | KyleK | asking for top three just sounds like we're helping you write a blog post |
| 18:14.07 | jql | at least offer attribution in your post. :) |
| 18:14.17 | srf21c | KyleK: thanks, not a big fan of the mailing format myself, much prefer web forums |
| 18:14.31 | srf21c | jql: thanks |
| 18:14.44 | leifmadsen | I hate web forums |
| 18:14.58 | srf21c | leifmadsen: why is that? |
| 18:15.00 | *** part/#asterisk mattbUK (n=mattbUK@82-46-93-20.cable.ubr16.stav.blueyonder.co.uk) |
| 18:15.07 | leifmadsen | so little control of what I'm seeing, and web forums are SLOW |
| 18:16.23 | *** join/#asterisk farkus_ (i=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 18:17.10 | jaytee | notfarkus_ is now known as notfarkus |
| 18:18.21 | KyleK | srf21c: If the mailing lists are more read than the web forums (if there is any) you may just have to suck it up :) |
| 18:18.50 | srf21c | KyleK: ok, well I did ask what was most popular, and if the mailing lists are, then so be it. |
| 18:19.01 | leifmadsen | ya, the web forums are barely read at all -- and even more rarely by people who actually know something |
| 18:19.14 | srf21c | leifmadsen: good to know, thanks. |
| 18:19.29 | leifmadsen | asterisk-users is pretty much the defacto location for help |
| 18:20.43 | KyleK | I probably should subscribe to it |
| 18:21.13 | *** join/#asterisk egypcio (n=egypcio@unaffiliated/egypcio) |
| 18:21.35 | srf21c | on another topic, does anybody have experience with voxalot? |
| 18:22.49 | srf21c | I'm researching methods for having an asterisk server automatically select a free or least cost route for outgoing calls. |
| 18:23.36 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 18:23.43 | carrar | thats a fun task |
| 18:24.23 | *** join/#asterisk farkus_ (i=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 18:24.55 | carrar | srf21c, writting that will sharpen your programing and dba skills |
| 18:25.01 | *** join/#asterisk lirakis (n=etamme@198.105.46.21) |
| 18:25.48 | carrar | assuming they aren't already sharp! |
| 18:26.16 | lirakis | hey d/ |
| 18:26.18 | *** part/#asterisk lirakis (n=etamme@198.105.46.21) |
| 18:26.28 | carrar | heh |
| 18:26.37 | *** join/#asterisk lirakis (n=etamme@198.105.46.21) |
| 18:26.41 | lirakis | hey guys |
| 18:26.49 | lirakis | ive setup asteris realtime for peers |
| 18:26.54 | lirakis | and it looks like its loaded |
| 18:26.59 | lirakis | but when i do sip show peers |
| 18:27.05 | lirakis | it doesnt show the peer i have in my database |
| 18:27.34 | lirakis | when i do a reload, i see "Binding sippeers to mysql/asterisk/sip" |
| 18:28.17 | lirakis | do i have to do some thing to make extconfig understand that it is sip peer stuff? or does asterisk figure that out automatically from the table? |
| 18:28.28 | srf21c | carrar: Aye, sounds like I might be biting off more than I can chew, so to speak. |
| 18:28.28 | carrar | enable realtime caching in sip.conf? |
| 18:28.52 | lirakis | carrar i didnt do that ... |
| 18:29.08 | carrar | srf21c, if you aren't biting off more, then you're nor learning! |
| 18:29.14 | carrar | r=t |
| 18:29.21 | carrar | lirakis, enable it |
| 18:29.41 | srf21c | lirakis? |
| 18:29.56 | carrar | I'm speaking to 2 poeple |
| 18:30.01 | carrar | common in here |
| 18:30.15 | srf21c | hah, sorry, missed the username. |
| 18:30.38 | lirakis | carrar: is there only a rtcachefriends or is there are rtcachepeers as well? |
| 18:30.45 | srf21c | didn't see the colon after the name |
| 18:30.51 | carrar | friends |
| 18:31.30 | lirakis | carrar: hmm so it has to be of type friend |
| 18:31.37 | carrar | jsut add it |
| 18:31.48 | carrar | then ask questions |
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| 18:32.55 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 18:33.03 | lirakis | carrar: i added it and did a reload - i still dont see my peer |
| 18:33.18 | carrar | where did you add it |
| 18:33.37 | carrar | I don't see it |
| 18:34.22 | lirakis | carrar: in the general section of my sip.conf |
| 18:34.49 | lirakis | <PROTECTED> |
| 18:34.53 | carrar | and you're peers re-resgistered? |
| 18:35.05 | lirakis | its not a registering peer - it is static |
| 18:35.47 | carrar | sip prune realtime PEERNAME |
| 18:35.51 | carrar | sip show peer PEERNAME load |
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| 18:40.38 | *** join/#asterisk Mw3 (i=mw3@mw3.hu) |
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| 18:47.07 | *** join/#asterisk grey (n=grey@vs1.svartalfheim.net) |
| 18:47.08 | CrazyTux[m] | Hey guys; is there a command in asterisk to only show the "current active channels / active calls" and not each individual channel being listed? i.e. "core show channels" |
| 18:47.09 | grey | heya |
| 18:48.00 | grey | Anyone using Link2VoIP with Asterisk? I want to setup a system so that my family can make calls to Zambia for fairly cheap, and according to l2v's rates here https://www.link2voip.com/international_rates.php?a=Z it would be a good price, |
| 18:48.04 | carrar | parse it out |
| 18:48.18 | grey | but the minimum payment they allow is $50USD, which kind of sucks, |
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| 18:51.32 | Maliuta | grey: you looked at pennytel (http://www.pennytel.com) |
| 18:52.06 | Maliuta | grey: not sure if zambia is on the $0.08AU untimed list ... but it's a good bet |
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| 19:00.05 | grey | Maliuta: I'll take a look, Thanks |
| 19:01.31 | *** part/#asterisk srf21c (n=seth@ip98-165-60-42.ph.ph.cox.net) |
| 19:03.06 | *** join/#asterisk the_5th_wheel (n=edd@webster.cybertek.co.za) |
| 19:05.42 | grey | Maliuta: where the heck can I get an actual list of countries their plan lets me call to? |
| 19:06.23 | *** join/#asterisk farkus_ (i=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 19:06.40 | grey | Maliuta: ah found it, and it does look good, Thanks very much for the pointer :) |
| 19:07.50 | Maliuta | grey: it's not the $0.08 untimed I get to canadia,us,uk .... |
| 19:08.00 | Maliuta | grey: remember the prices are $AU |
| 19:08.14 | Maliuta | which if you're in the US makes them cheaper |
| 19:09.18 | grey | well, I'm in Canada |
| 19:09.36 | grey | but I think I can figure it out from here |
| 19:11.13 | Maliuta | still cheaper |
| 19:11.34 | carrar | Why not set your family up with sip devices |
| 19:11.42 | carrar | then it's FREE |
| 19:11.46 | Maliuta | my 'rents are in Ft MacMurray (AB) ... they make a killing transferrring monies home |
| 19:12.17 | Maliuta | carrar: I think the point is they want to call PSTN numbers in zambia |
| 19:12.42 | Maliuta | carrar: and it's not "free", it still costs bandwidth |
| 19:13.05 | carrar | I consider bw a necessity |
| 19:13.09 | carrar | so calls are free! |
| 19:13.57 | Maliuta | yeah, go somewhere that bw is charged at stupid rates (like Telstra do here) |
| 19:14.06 | grey | I have no one technical over there who could set up a sip phone anyways, |
| 19:14.34 | *** join/#asterisk srf21c (n=seth@ip98-165-60-42.ph.ph.cox.net) |
| 19:14.49 | Maliuta | I have similar issues in .ca ... if I had been able to get my hands on the right router it wouldn't have been an issue |
| 19:14.50 | grey | so yes, calling PSTN is the best bet, and it's only got to be setup for about a month anyways |
| 19:15.25 | Maliuta | hmm, should call mum before I go to beds |
| 19:16.03 | carrar | Sounds like you only have 1 choice |
| 19:16.08 | carrar | everyone is going to have to move |
| 19:16.11 | carrar | :) |
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| 19:25.38 | srf21c | I'm having problem with a Snom SIP IP phone dropping into "Unreachable" SIP peer status. Asterisk server is a colo box, IP phone is at home behind a firewall running PAT. |
| 19:26.04 | srf21c | If I force the phone to reregister, that will temporarily "solve" the problem. |
| 19:26.29 | srf21c | What I'm thinking about doing is setting up another asterisk server on my home LAN, then using IAX to trunk to my colo asterisk server. |
| 19:26.31 | srf21c | thought? |
| 19:26.46 | srf21c | (s)? |
| 19:26.48 | Maliuta | sounds like something on the firewall is screwey ... but you provide no evidence |
| 19:26.59 | srf21c | Running pf on OpenBSD 4.4 |
| 19:27.11 | srf21c | as well as the colo box. |
| 19:27.14 | Maliuta | goes for cigarmarette before calling 'rents over his working SIP |
| 19:27.38 | srf21c | I was getting some blocked udp packets from IP phone to colo'd ast* server. |
| 19:28.02 | srf21c | But opening up inbound udp ports above 8000 on asterisk server seems to solve that. |
| 19:29.14 | drmessano | Did you open ANY ports for RTP prior? |
| 19:30.16 | srf21c | oh yeah. |
| 19:30.28 | srf21c | I opened everything above 10000 |
| 19:30.30 | *** join/#asterisk farkus_ (i=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 19:30.43 | srf21c | but apparently the Snom was trying to hit the server with upd in the 8000 range. |
| 19:30.44 | drmessano | Did you match whats defined in rtp.conf? |
| 19:31.04 | srf21c | haven't even looked at my rtp.conf yet, hold on and I'll check. |
| 19:31.05 | drmessano | The SNOM is only going to hit whats negotiated |
| 19:31.58 | srf21c | Hmph. I don't even have an rtp.conf file in /etc/asterisk. |
| 19:32.06 | drmessano | um yeah |
| 19:32.13 | drmessano | ~sipnat |
| 19:32.13 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 19:32.17 | drmessano | That may help |
| 19:32.30 | srf21c | drmessano: gracias. |
| 19:33.08 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:33.10 | srf21c | You wouldn't happen to know what ast* 1.4 defaults to for udp ports, would you? |
| 19:34.27 | srf21c | Btw, case #9 in the 2nd link you send me, sez it's solved with nat=yes and qualify=yes in sip.conf, Which is what I have configured. |
| 19:34.44 | drmessano | udp is a transport.. youre asking about about specific protocols |
| 19:34.54 | drmessano | Define SOMETHING in rtp.conf |
| 19:34.58 | drmessano | 10000 to 20000 perhaps |
| 19:35.50 | drmessano | rtpstart=10001 |
| 19:35.50 | drmessano | rtpend=11000 |
| 19:35.52 | drmessano | There |
| 19:36.08 | *** join/#asterisk lirakis (n=lirakis@198.105.46.21) |
| 19:36.20 | srf21c | got. Btw rtp.conf sez that default range is ; Defaults are rtpstart=5000 and rtpend=31000 |
| 19:36.21 | lirakis | hey guys - sorry my computer totally froze before |
| 19:36.41 | srf21c | I think I'll reign it in to 10000-200000 |
| 19:36.52 | drmessano | Default is 10000 to 20000.. You dont need 20000 ports |
| 19:37.00 | drmessano | Asterisk ONLY uses what you DEFINE |
| 19:37.08 | lirakis | so ... ive got realtime peers setup, mysql realtime status shows it is connected and i have realtime caching enabled in sip.conf |
| 19:37.09 | lirakis | however |
| 19:37.11 | FSB_1 | lirakis: TOTALLY? |
| 19:37.13 | lirakis | when i do sip show peers |
| 19:37.13 | drmessano | There is no WRONG answer unless you run out of ports |
| 19:37.14 | FSB_1 | Like ice? |
| 19:37.34 | drmessano | You are telling Asterisk what ports it is allowed to use |
| 19:37.36 | *** join/#asterisk farkus_ (i=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 19:37.38 | lirakis | FSB_1: yeah like my cpu overheated and just stopped processing anything |
| 19:37.39 | srf21c | er, I don't believe the statement that "Asterisk ONLY uses what you DEFINE" is entirely correct |
| 19:37.45 | drmessano | Yes, it is |
| 19:37.56 | lirakis | any how ... sip show peers it does not show the peer i have defined in the mysql table |
| 19:37.59 | drmessano | Thats why you SET it |
| 19:38.05 | srf21c | Because it has been using rtp ports on it's own, without any /etc/asterisk/rtp.conf file present. |
| 19:38.33 | srf21c | furthermore, the example rtp.conf file itself states that the default range is rtpstart=5000 and rtpend=31000 |
| 19:38.33 | drmessano | .... |
| 19:38.36 | drmessano | Christ |
| 19:39.05 | srf21c | I can understand why it would be beneficial to define a range. |
| 19:39.22 | drmessano | If you define NOTHING, then YES, it will use some internal default.. But when it comes to what SHOULD I DEFINE, the answer is ASTERISK ONLY USES WHAT YOU DEFINE and therefore there is NO WRONG ANSWER.. |
| 19:39.48 | srf21c | hey, no need to shout, I hear ya screaming. ;) |
| 19:40.17 | drmessano | No need to nitpick through something when someone is trying to help you with something.. |
| 19:40.29 | srf21c | I appreciate the help. |
| 19:40.40 | srf21c | Just striving for accuracy, that's all. |
| 19:40.52 | srf21c | complete understanding. |
| 19:41.08 | lirakis | so - is there anything i need to do to "map" the extconfig binding to sip peers ? or is it automatically figured out by the fields in the table? |
| 19:41.19 | srf21c | So I'm curious how small I can shrink that rtp range down to. |
| 19:41.28 | srf21c | a hundred ports maybe? |
| 19:41.31 | drmessano | 2 ports per call |
| 19:41.39 | Maliuta | I use 10 ports |
| 19:41.40 | drmessano | So work out how many calls you plan concurrently |
| 19:41.48 | Maliuta | don't expect that many concurrent calls here |
| 19:42.00 | KyleK | calls Maliuta 6 times in a row |
| 19:42.06 | srf21c | drmessano: no more than 5 calls at a time. |
| 19:42.12 | KyleK | surround sound voicemail ;) |
| 19:42.16 | drmessano | That would be 10.. make it 20 |
| 19:42.24 | srf21c | aye. |
| 19:42.31 | Maliuta | KyleK: I think your host is on the firewall blacklist ;P |
| 19:42.35 | KyleK | actually tbat'd be pretty neat |
| 19:42.38 | drmessano | That way you can be lazy and forget about only having 10 ports open |
| 19:43.23 | Maliuta | I even only allow SIP/RTP from the hosts at my VTSP's |
| 19:43.40 | srf21c | is is possible to have asterisk load the rtp.conf file from the asterisk console? |
| 19:43.45 | grey | What kind of pattern would I use, to match any numeric number dialed that doesn't start with a *? |
| 19:43.52 | drmessano | Maliuta: Just because youre paranoid doesnt mean we're not out to get you |
| 19:43.56 | grey | something like [0-9]+ in perl |
| 19:44.14 | drmessano | srf21c: asterisk -rx reload from the cli |
| 19:44.15 | Maliuta | drmessano: I'm aware everone is out to get me, I have the logs to prove it |
| 19:44.34 | Maliuta | drmessano: they can't get past my tin foil hat though ;P |
| 19:44.36 | leifmadsen | grey: uhhh..... _X. ? |
| 19:44.48 | grey | that's probably it |
| 19:44.56 | leifmadsen | just don't match for *.... |
| 19:45.13 | grey | I'm reading the asterisk docs and stuff, but am still pretty out of touch with everything it... |
| 19:45.22 | srf21c | Maliuta: are you running encryption between your IP handsets and your asterisk server? That's next on my todo list. |
| 19:45.30 | *** join/#asterisk farkus_ (i=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 19:45.46 | drmessano | Youre gonna use app_unicorn? |
| 19:45.51 | Juggie | hmmm [TK]D-Fender, you there? |
| 19:46.22 | Maliuta | srf21c: not sure my cisco can handle the encryption ... in anycase being on the same LAN segment makes it bit of a waste of effort |
| 19:47.01 | [TK]D-Fender | Juggie: Just got in |
| 19:48.35 | Juggie | [TK]D-Fender, anything special i have to do to get cdr userfield to work? i have it enabled in cdr_mysql.conf and i'm doing a set(CDR(userfield)=blah) |
| 19:48.59 | Juggie | but its not making it into the Master.csv or mysql and yes userfield=1 is set in the mysql config |
| 19:49.15 | [TK]D-Fender | Juggie: Never played with that... |
| 19:49.30 | Maliuta | right, bed for me |
| 19:49.50 | seanbright | Juggie: what version of asterisk? |
| 19:52.23 | *** join/#asterisk farkus_ (i=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 19:53.35 | lirakis | arrgh!!! |
| 19:53.43 | leifmadsen | matey |
| 19:53.46 | lirakis | wtf doest atfot mention switch => Realtime/@extensions |
| 19:53.52 | lirakis | *doesnt |
| 19:54.00 | leifmadsen | lirakis: because we didn't do a lot of realtime stuff? |
| 19:54.12 | leifmadsen | lirakis: because it's not "Asterisk: The Definitive Guide"? |
| 19:54.32 | lirakis | lol |
| 19:54.37 | leifmadsen | isn't laughing. |
| 19:54.37 | seanbright | cuz leifmadsen is the freakin' man, maybe? |
| 19:54.44 | leifmadsen | seanbright: well fuckin' duh |
| 19:54.55 | [TK]D-Fender | lirakis: more like "OMG my highschool physics book doesn't ahve advanced string theory?!?! WHAT A RIP-OFF!!!!!" |
| 19:55.25 | Juggie | seanbright, 1.4 |
| 19:55.43 | lirakis | leifmadsen: i know, but it does go through a basic setup - its just incomplete. broken examples arent fun for the reader, just like ungrateful annoying readers are annoying for you |
| 19:56.14 | leifmadsen | lirakis: you're always welcome to file an errata |
| 19:56.40 | lirakis | leifmadsen: thanks for the work on the book though - its a great reference. i will put in errata req. is there another edition in planning/works? |
| 19:56.48 | seanbright | Juggie: pastebin your cdr_mysql.conf |
| 19:56.51 | leifmadsen | lirakis: in planning, yes |
| 19:56.56 | seanbright | Juggie: unless you want to figure it out yourself |
| 19:57.04 | leifmadsen | lirakis: needs a major rewrite though, so will be about a year from now |
| 19:57.13 | seanbright | leifmadsen: want some help? |
| 19:57.19 | Juggie | seanbright, ok but its pretty simple. |
| 19:57.20 | seanbright | i'll except wads of cash, of course. |
| 19:57.35 | leifmadsen | seanbright: perhaps! First thing I want to do is figure out a good outline |
| 19:57.38 | seanbright | Juggie: you can paste to PM if you want |
| 19:57.52 | leifmadsen | one of these days I'll setup a conference call, and anyone who wants to participate can |
| 19:57.56 | seanbright | awesome |
| 19:58.02 | lirakis | argh. i still cant get my dynamic realtime peer to show up in sip show peers |
| 19:58.11 | Juggie | seanbright, http://pastebin.ca/1441458 |
| 19:58.20 | Juggie | not much to see really, it works fine i've setup mysql a 100 times |
| 19:58.24 | Juggie | just not writing the user field |
| 19:58.26 | leifmadsen | lirakis: it doesn't show up until it is used |
| 19:58.32 | leifmadsen | lirakis: and only when it is cached |
| 19:58.37 | leifmadsen | lirakis: it won't just "show up" |
| 19:58.44 | leifmadsen | there is no pre-caching in asterisk realtime |
| 19:58.45 | seanbright | Juggie: yeah, that looks good. |
| 19:58.47 | lirakis | leifmadsen: okay - even if its static and not a registering host |
| 19:58.53 | leifmadsen | lirakis: yes |
| 19:59.05 | leifmadsen | and it won't ever show up without caching |
| 19:59.14 | lirakis | leifmadsen: great thanks. i do have caching setup |
| 19:59.27 | seanbright | Juggie: asterisk 1.4 svn? or a release? |
| 20:00.03 | *** join/#asterisk BadHAL (n=nn@cpe-72-179-194-139.stx.res.rr.com) |
| 20:00.13 | Juggie | latest release |
| 20:00.21 | Juggie | the dialplan looks like this when executing: Set("DAHDI/1-1", "CDR(userfield)=type=rentalproductpickup#facility=324#distributor=uhs#po=#machine=0#status=hungup") in new stack |
| 20:00.28 | Juggie | not sure maybe my formatting messed it up |
| 20:00.41 | srf21c | ok cool, have the new rtp port range setup. Hopefully this will solve my unreachable sip extension issue. |
| 20:01.06 | seanbright | Juggie: what does the Set look like in extensions.conf? |
| 20:01.23 | *** join/#asterisk farkus_ (i=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 20:01.26 | seanbright | ohh |
| 20:01.31 | seanbright | nevermind, misread you. |
| 20:01.38 | seanbright | how big is the userfield column in your table? |
| 20:01.50 | Juggie | big enough |
| 20:01.54 | seanbright | hrm. |
| 20:01.55 | Juggie | mysql would trunecate it |
| 20:02.02 | Juggie | but its 100% empty |
| 20:02.06 | Juggie | its also empty in Master.csv |
| 20:02.11 | Juggie | and its enabled in cdr.conf |
| 20:03.14 | *** join/#asterisk InfoMomo (n=infomomo@206.167.94.12) |
| 20:03.45 | Juggie | seanbright, here is what the DP says, i'm nooping it back out and it looks ok. |
| 20:03.46 | Juggie | http://pastebin.ca/1441465 |
| 20:04.35 | seanbright | Juggie: pb cdr.conf por favor? |
| 20:09.23 | *** join/#asterisk farkus_ (i=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 20:09.35 | Juggie | k |
| 20:10.11 | lirakis | thanks leif, ill have to mess around more at home later |
| 20:11.47 | Juggie | seanbright, http://pastebin.ca/1441473 |
| 20:12.11 | seanbright | Juggie: that is weird. |
| 20:12.22 | seanbright | that is my technical assessment. |
| 20:12.22 | seanbright | heh |
| 20:12.39 | Juggie | great, i dont think i can tell that to my client :) |
| 20:12.40 | seanbright | i would test a very simple dialplan |
| 20:12.49 | seanbright | answer(), set userfield, hangup() |
| 20:13.31 | Juggie | i will try a restart |
| 20:13.38 | Juggie | since i compiled cdr_Addon and loaded it in |
| 20:14.33 | seanbright | well, cdr_csv should still work. |
| 20:15.11 | *** join/#asterisk farkus_ (i=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 20:16.11 | *** join/#asterisk botox93 (n=botox93@213.221.82.242) |
| 20:17.00 | Juggie | seanbright, alas it does not |
| 20:17.12 | Juggie | it has "" in place of the user field |
| 20:17.13 | Juggie | quite odd eh |
| 20:18.03 | Juggie | it may possibly be because of the action:originate i suppose |
| 20:19.15 | seanbright | don't want to try the simple test dialplan idea? |
| 20:21.13 | Juggie | well i dont know the # to dial into the system to be honest |
| 20:21.20 | Juggie | all the calls are originated |
| 20:21.26 | Juggie | (using ami) |
| 20:21.38 | Juggie | i changed the set to set(CDR(userfield)=test) |
| 20:21.43 | Juggie | so we'll see if it makes any difference |
| 20:22.22 | Juggie | just have to wait for a call to happen now |
| 20:22.31 | *** join/#asterisk farkus_ (i=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 20:26.22 | The_TiK | i am trying to get an agi-script to call a number, right now, i call in, it executes the script but i get a congestion error.. any ideas? |
| 20:26.46 | *** join/#asterisk SparFux (n=raoul@e182031114.adsl.alicedsl.de) |
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| 20:27.52 | SparFux | Hello all. I wan tto connect a sip client to my asterisk and it is set up and can be pinged. I installed the authentication stuff and used host=<ip> annd asterisk shows me Zyxel1/Zyxel1 192.168.118.55 5062 UNREACHABLE Why is it unreachable, I can ping it! ? |
| 20:28.28 | The_TiK | the script just has print exec dial iax2 |
| 20:30.04 | SparFux | And there is no firewall issue. |
| 20:30.30 | *** join/#asterisk farkus_ (i=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 20:30.55 | SparFux | I use this configuration in sip.conf: http://pastebin.com/d76c41c11 |
| 20:31.34 | seanbright | The_TiK: pastebin your script |
| 20:31.36 | seanbright | ~pb |
| 20:31.37 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/ |
| 20:31.56 | seanbright | but someone else will have to look at it since i am about to shower |
| 20:31.58 | seanbright | kbye |
| 20:32.00 | The_TiK | its just the one line |
| 20:32.23 | The_TiK | print "EXEC DIAL IAX2/user@service/111111111\n" |
| 20:34.08 | The_TiK | it works if I set the dial line in the extensions.conf...but doesn't work when i doing it through the script |
| 20:36.01 | Juggie | seanbright: ping |
| 20:36.29 | Juggie | w/ Set(CDR(userfield)=test) it still logs nothing.. |
| 20:36.30 | Juggie | oddd... |
| 20:37.12 | Juggie | hmmm |
| 20:37.13 | Juggie | wait a minute |
| 20:37.17 | Juggie | its not generating a cdr at all |
| 20:37.36 | Juggie | its only generating a CDR for one side of the originate |
| 20:37.38 | Juggie | not both |
| 20:37.49 | *** join/#asterisk Simon- (i=simon@proxima.lp0.eu) |
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| 20:39.30 | Simon- | for some reason I'm getting extremely long delays in Playback of a gsm file with 1.6.1.0, but only with IAX2... http://s85.org/sbSUoZBQ is a recording of it playing you-have-these-options |
| 20:39.34 | Yurik | anybody know a voice changer for asterisk except http://lobstertech.com/2005/oct/31/asterisk_voice_changer/ ? |
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| 20:55.16 | seanbright | The_TiK: just that one line? that's not even an executable under linux... |
| 20:59.09 | The_TiK | its perl |
| 21:02.30 | *** join/#asterisk farkus_ (i=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 21:03.19 | seanbright | The_TiK: i just put that one line in a file, chmod 755 file, and ./file and got an error |
| 21:03.31 | seanbright | so do me a favor |
| 21:03.35 | seanbright | pastebin your script |
| 21:06.25 | *** join/#asterisk Defraz (n=T0tal@24-117-236-174.cpe.cableone.net) |
| 21:07.17 | The_TiK | http://pastebin.com/m56500bfc |
| 21:07.30 | The_TiK | i also put the asterisk output |
| 21:08.34 | seanbright | try: |
| 21:08.40 | seanbright | print "EXEC DIAL IAX2/user\@service/111111111\n" |
| 21:08.57 | seanbright | or \@outbound |
| 21:09.05 | seanbright | or whatever your actual script is since you refuse to pastebin it |
| 21:09.41 | seanbright | (the relavent addition is the \ in front of the @) |
| 21:09.59 | The_TiK | i did pastbin it...thats all i have in the file...is the #!/usr/bin/perl and the print line |
| 21:10.13 | seanbright | ok |
| 21:10.25 | *** join/#asterisk farkus_ (i=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
| 21:10.31 | seanbright | farkus_: you're KILLING me |
| 21:11.15 | seanbright | The_TiK: have you added the \ and tested? |
| 21:11.25 | The_TiK | heh, thanks seanbright, i don't know why i was overlooking escaping the @ sign |
| 21:11.33 | The_TiK | yeah, it works, ty |
| 21:12.13 | seanbright | np |
| 21:12.35 | srf21c | so the rtp.conf changes did not solve my intermittment sip peer unreachable problem. :( |
| 21:13.06 | srf21c | Snom 370 still waffles back and forth betwixt OK status and unreachable. |
| 21:13.28 | srf21c | bugger. |
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| 21:34.56 | xphree | Hello, anyone can help me with a problem in a2billing? |
| 21:35.05 | xphree | or there is a support channel about a2billing? |
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| 21:49.08 | *** mode/#asterisk [+b farkhus_!*@*] by leifmadsen |
| 21:49.28 | *** mode/#asterisk [+b farkas!*@*] by leifmadsen |
| 21:50.39 | [TK]D-Fender | leifmadsen: jsut do it against that specific host. Odds are its dynamic and his resetting it might correct the problem AND let him back in. |
| 21:51.23 | leifmadsen | [TK]D-Fender: he can just change his name when he gets back |
| 21:51.39 | [TK]D-Fender | leifmadsen: Except he needs to be ID'd for the chan to let him in |
| 21:51.51 | leifmadsen | [TK]D-Fender: if you know how to do it better, go nuts |
| 21:52.02 | leifmadsen | you have ops access |
| 21:52.20 | [TK]D-Fender | leifmadsen: True, but I reserve them for real trouble-makers :) |
| 21:52.34 | [TK]D-Fender | leifmadsen: With great power comes.... awesomeness ;) |
| 21:52.37 | leifmadsen | he can register farkas2 or something |
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| 22:58.44 | rjune_ | I'm trying to get all page working with some polycom phones. I'm not sure what feature I need to look for in the manual |
| 22:59.07 | rjune_ | I have the all page working, but the phones don't all pickup automatically |
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| 23:16.15 | sprite-- | Does anyone have a good query for matching a number to most specific prefix for rates? |
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| 23:36.00 | click | evening - i've got asterisk-1.6.1.0 with asterisk-gui (SVN-branch-2.0-r4809) installed, though creation of users doesn't seem to be accepted, ie. new extensions are placed in users.conf as it should but when trying to log in the user, it doesn't accept it (not logging him/her in) - what is the proper method to get asterisk to use users.conf for new users and creating extensions? (i might be doing something wrong here, obviously) :) |
| 23:36.10 | click | (linux) |
| 23:36.38 | click | (and yes, the user is made through the gui) |
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| 23:55.04 | ectospasm | click: what type of users are you adding? SIP, IAX, analog, etc? If you add them in AsteriskGUI, do they show up in "sip show peers" or whatever? |
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| 23:59.52 | click | ectospasm: sip, and not showing when doing sip show peers |