| 00:00.06 | [TK]D-Fender | jeev: NO |
| 00:00.10 | dinominant | Qwell: Asterisk says it is, but ok. next is this: ERROR[3516]: chan_iax2.c:8895 set_config: Unable to load config iax.conf |
| 00:00.11 | [TK]D-Fender | jeev: 4 are DATA, 4 are POWER |
| 00:00.11 | jeev | ok, do we call every 2 wires a pair? obviously |
| 00:00.14 | jeev | ahhh |
| 00:00.23 | jeev | Fender, some of the standards i read, data was shared on the same power line |
| 00:00.28 | Qwell | dinominant: Next? How about you tell us what actually happens. |
| 00:00.37 | jeev | http://www.altair.org/labnotes_POE.html |
| 00:00.38 | [TK]D-Fender | jeev: You don't have a choice on this. Get a PoE switch up there or brick one. |
| 00:00.38 | jeev | first option |
| 00:01.17 | [TK]D-Fender | jeev: FORGET IT. |
| 00:01.20 | *** join/#asterisk moy (n=moy@189.169.71.102) |
| 00:01.26 | jeev | okie |
| 00:01.26 | jeev | :D |
| 00:01.31 | [TK]D-Fender | jeev: It also takes signalling to trigger, its not dumb voltage |
| 00:01.56 | jeev | oklee doklee |
| 00:01.59 | jeev | i will have to buy another brick |
| 00:02.15 | jeev | i need to find one that i dont have to buy, locallyt |
| 00:03.53 | dinominant | Qwell: well when I try to start asterisk (/etc/init.d/asterisk start) it fails. |
| 00:05.21 | dinominant | Qwell: The logs only contain warnings and notices, no errors |
| 00:06.12 | tzafrir | dinominant, how do you see it fails? |
| 00:06.34 | tzafrir | you can't connect to it with asterisk -r? |
| 00:08.15 | dinominant | tzafrir: /etc/init.d/asterisk start gives me a [ !! ] instead of an [ OK ] |
| 00:08.34 | dinominant | tzafrir: and asterisk -r gives: Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
| 00:08.50 | tzafrir | dinominant, so aparantly there is no asterisk process running after that |
| 00:09.09 | tzafrir | you see nothing in the logs: are the logs actually newer? |
| 00:09.20 | tzafrir | maybe asterisk fails to write to the logs? |
| 00:10.17 | *** join/#asterisk mattwj2002 (n=matt@c-76-17-132-205.hsd1.mn.comcast.net) |
| 00:10.24 | mattwj2002 | hey everyone |
| 00:10.45 | dinominant | tzafrir: hmm, no. I just deleted /var/log/asterisk/messages and tried to start it again. The log file wasn't created again |
| 00:10.46 | [TK]D-Fender | dinominant: run it directly yourself as asterisk -gvvvvvvvc |
| 00:10.55 | [TK]D-Fender | dinominant: And look at what comes out in CLI |
| 00:11.00 | [TK]D-Fender | dinominant: PASTEBIN it all. |
| 00:11.02 | [TK]D-Fender | ~pb |
| 00:11.02 | jbot | [~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 |
| 00:11.03 | [TK]D-Fender | ^^^^^^^^^^^^^^ |
| 00:11.11 | mattwj2002 | hey everyone |
| 00:11.12 | tzafrir | dinominant, so there's an issue of permissions |
| 00:11.37 | mattwj2002 | I am going to try this Teliax setup again |
| 00:11.41 | tzafrir | [TK]D-Fender, that would actually work. But create log files that the asterisk user cannot write on later on |
| 00:11.45 | mattwj2002 | wish me luck |
| 00:12.54 | dinominant | [TK]D-Fender: http://pastebin.com/m77587795 |
| 00:12.58 | tzafrir_laptop | dinominant, you need /var/log/asterisk with write permissions to the user you run asterisk as |
| 00:13.29 | dinominant | tzafrir_laptop: currently I'm running it as root |
| 00:13.41 | [TK]D-Fender | dinominant: Ok, you don't seem to have NAY sample configs. |
| 00:13.57 | [TK]D-Fender | dinominant: go find them for Gentoo or compile from source as per the directions. |
| 00:13.58 | mattwj2002 | dinominant can you use sudo? |
| 00:14.09 | [TK]D-Fender | ANY* |
| 00:14.17 | [TK]D-Fender | mattwj2002: no need |
| 00:14.30 | mattwj2002 | okay....I thought sudo would be safer than full root |
| 00:14.51 | [TK]D-Fender | dinominant: GO GET THE SAMPLE CONFIGS |
| 00:14.53 | tzafrir_laptop | hmmm... he does have the autoload enabled in modules.conf, which is a no-default value |
| 00:15.02 | dinominant | [TK]D-Fender: will do, thanks |
| 00:15.07 | tzafrir_laptop | default without config file is not to load any module |
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| 00:23.22 | mattwj2002 | so anything cool in the new beta? |
| 00:23.33 | mattwj2002 | as compared to the stable version |
| 00:24.23 | [TK]D-Fender | mattwj2002: Go read the changelog |
| 00:24.55 | mattwj2002 | sorry I just thought people might want to talk about it |
| 00:25.00 | mattwj2002 | I did anyways |
| 00:26.35 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 00:28.01 | [TK]D-Fender | mattwj2002: They make changelogs and post articles and all of these things so we don't have to answer that question 50 million times everytime someone comes in here asking the same. |
| 00:28.17 | mattwj2002 | sorry |
| 00:28.32 | mattwj2002 | I was just making conversation |
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| 00:40.28 | C4colo | is there a way to have the CDR record another number in the database than the extension from which the Dial() command was called? |
| 00:41.23 | C4colo | for example, someone calls 8001234567 and that extension does a Goto(3031234567,1) where the call is actually originated by a Dial command |
| 00:41.31 | C4colo | I want the original 800 number to be in the CDR |
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| 02:02.51 | jaytee | it's quiet |
| 02:02.56 | mattwj2002 | hi everyone |
| 02:03.02 | jaytee | hi |
| 02:03.06 | mattwj2002 | I need some help if you would be so kind |
| 02:03.30 | jaytee | I'm not kind, I'm crusty and mean :-) |
| 02:03.53 | mattwj2002 | runs away :P |
| 02:04.36 | jaytee | so what's your problem/question? |
| 02:04.49 | mattwj2002 | I have a new Asterisk box setup |
| 02:04.59 | mattwj2002 | I am just trying to get Teliax to work |
| 02:05.09 | mattwj2002 | I can make outgoing calls but can't get incoming |
| 02:05.23 | jaytee | I've never used Teliax |
| 02:05.23 | mattwj2002 | I turned on debug and got this error |
| 02:05.46 | mattwj2002 | <PROTECTED> |
| 02:05.47 | mattwj2002 | <PROTECTED> |
| 02:06.02 | mattwj2002 | there is more too it but I would have to use the paste bin |
| 02:06.07 | mattwj2002 | what is that address again |
| 02:06.18 | jaytee | address? |
| 02:06.26 | jaytee | ~pastebin |
| 02:06.26 | jbot | [~pastebin] 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 |
| 02:06.28 | mattwj2002 | for pastebing |
| 02:06.30 | mattwj2002 | thanks |
| 02:07.30 | jaytee | also, you might want to look on www.voip-info.org in the Asterisk section. They have alot of links to howtos and I'm pretty sure I saw one about setting up Teliax. |
| 02:08.36 | mattwj2002 | http://paste.debian.net/13027/ |
| 02:11.09 | joat | that looks like your inbound is not set |
| 02:11.23 | jaytee | yep, looks that way |
| 02:12.25 | mattwj2002 | please explain |
| 02:12.45 | joat | you have your register line set, right? |
| 02:12.55 | mattwj2002 | yup |
| 02:13.20 | joat | did you set up a container for inbound? |
| 02:13.27 | joat | something that starts with |
| 02:13.35 | joat | [teliax or somesuch] |
| 02:13.39 | joat | username= |
| 02:13.46 | mattwj2002 | yup |
| 02:13.47 | joat | secret= |
| 02:13.57 | joat | hmmm... |
| 02:14.07 | mattwj2002 | what do I need in the extensions.conf |
| 02:14.15 | joat | check username, password, and the context |
| 02:14.15 | *** join/#asterisk korihor (n=korihor@190.39.163.45) |
| 02:14.53 | joat | you have a "s" extension in the context you declared in the iax.conf file? |
| 02:14.54 | mattwj2002 | do I need anything that starts with an s? |
| 02:15.03 | joat | (easiest way out of this) |
| 02:15.05 | joat | yes |
| 02:15.05 | mattwj2002 | no I don't :| |
| 02:15.26 | *** join/#asterisk angom_w (n=angom@201.170.65.143) |
| 02:15.28 | mattwj2002 | how do I do that? |
| 02:15.40 | *** join/#asterisk PepOSX (n=angeldav@200.93.28.168) |
| 02:15.51 | joat | if you declared the context to be "default" then |
| 02:15.54 | [TK]D-Fender | mattwj2002 "[Jul 25 21:07:14] NOTICE[19905]: chan_iax2.c:8365 socket_process: Rejected connect attempt from IP ADDRESS, who was trying to reach 'TELEPHONE NUMBER@'" My guess is you need something like "TELEPHONE NUMBER". But them maybe I'm jsut crazy |
| 02:15.56 | joat | [default] |
| 02:16.04 | joat | exten => s,1,Answer |
| 02:16.11 | joat | exten => s,2,Wait(2) |
| 02:16.12 | [TK]D-Fender | no, NOT "s" |
| 02:16.19 | jaytee | you'll need to point to a context in extensions.conf in the define for the Teliax account in your iax.conf that you want to handle incoming calls. |
| 02:16.47 | mattwj2002 | okay |
| 02:17.34 | joat | still the same theory... set up a context for inbound, use "goto" to forward to whatever you're using for a phone |
| 02:17.48 | joat | fancy stuff later |
| 02:19.02 | mattwj2002 | okay |
| 02:19.05 | mattwj2002 | ummm |
| 02:19.42 | mattwj2002 | joat what do I need for the s extension |
| 02:19.47 | mattwj2002 | I think that is what I am missing |
| 02:20.16 | joat | treat it like any other extension... it's the one that gets used by default |
| 02:20.20 | mattwj2002 | I all ready have a exten => telephonenumber,1,Answer() |
| 02:20.21 | ManxPower | mattwj2002: "s" means "don't know the destination number" You almost always know the destination number, except for FXO ports, and a few ITSPs. |
| 02:20.33 | mattwj2002 | and then |
| 02:20.53 | mattwj2002 | exten => telephonenumber,2,Dial(SIP/2000,20,tr) |
| 02:20.57 | mattwj2002 | should that work |
| 02:21.03 | mattwj2002 | that is in the correct context |
| 02:21.15 | *** part/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 02:21.39 | joat | telephonenumber is the username you used in iax.conf |
| 02:21.50 | joat | and 2000 is the line you have set up for you hard phone |
| 02:21.52 | mattwj2002 | correct |
| 02:21.56 | mattwj2002 | correct |
| 02:22.03 | joat | err... s/you/your/ |
| 02:22.47 | mattwj2002 | this is all I have in my extensions.conf |
| 02:22.51 | mattwj2002 | [general] |
| 02:22.51 | mattwj2002 | static=yes |
| 02:22.51 | mattwj2002 | writeprotect=yes |
| 02:22.51 | mattwj2002 | [default] |
| 02:23.04 | mattwj2002 | and then the stuff for my extensions |
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| 02:25.34 | mattwj2002 | anyone have any ideas? |
| 02:25.40 | mattwj2002 | sorry about the flood |
| 02:28.50 | joat | recheck the phonenumber (username) and secret? |
| 02:29.28 | mattwj2002 | okay |
| 02:29.35 | joat | watch the log file too |
| 02:31.08 | mattwj2002 | questions for you |
| 02:31.18 | mattwj2002 | in the iax.conf what does the [name] set |
| 02:31.25 | mattwj2002 | in this case what is the name? |
| 02:31.48 | joat | look here too: https://www.teliax.com/pages/asteriskconfig |
| 02:32.17 | mattwj2002 | :O |
| 02:32.22 | mattwj2002 | I might have found something! |
| 02:33.12 | TJNII | mattwj2002: [name] is a context. It is a identifier for you, and all the settings that follow it are in that context and applied to that device. |
| 02:33.31 | mattwj2002 | I fixed it!!! |
| 02:33.43 | mattwj2002 | buys beer for all those who helped! |
| 02:34.52 | mattwj2002 | it looks like the username and context need to be the same in iax.conf |
| 02:35.04 | mattwj2002 | I changed that and I was able to receive an incoming call! |
| 02:35.05 | mattwj2002 | :D |
| 02:35.24 | mattwj2002 | I also had to update my extensions.conf to the context |
| 02:35.32 | mattwj2002 | passes out some cold ones |
| 02:35.40 | TJNII | The username and context don't need to be the same. |
| 02:36.39 | TJNII | IIRC the username and secret need to be the same on both ends, though. |
| 02:36.54 | mattwj2002 | I think for teliax is has to be |
| 02:36.57 | mattwj2002 | look at this |
| 02:37.03 | mattwj2002 | https://www.teliax.com/pages/asteriskconfig |
| 02:37.16 | mattwj2002 | [voipuser] |
| 02:37.23 | mattwj2002 | username=voipuser |
| 02:37.26 | TJNII | Stop |
| 02:37.29 | TJNII | Use a pastebin |
| 02:37.31 | TJNII | ~pb |
| 02:37.32 | jbot | [~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 |
| 02:37.34 | mattwj2002 | sorry |
| 02:38.12 | mattwj2002 | http://pastebin.com/m20c588d3 |
| 02:38.25 | mattwj2002 | notice the context and the username are the same |
| 02:38.52 | mattwj2002 | all I know is it works :) |
| 02:39.26 | TJNII | Well, they can be the same, but they don't have to be. That's all. |
| 02:39.47 | TJNII | But it works, so what's the difference |
| 02:40.10 | mattwj2002 | exactly :) |
| 02:40.21 | jaytee | the username and context DON'T have to be the same, but the context in iax.conf must match the context in extensions.conf |
| 02:41.25 | TJNII | jaytee: Are you sure about that? I have entries in my iax.conf that don't match anything in extensions.conf. |
| 02:41.39 | TJNII | except fot the context= of course. |
| 02:43.06 | joat | TJNII: i think you're agreeing with jaytee |
| 02:43.22 | jaytee | it's the contexts that have to match i.e.; context=teliax in iax.conf and [teliax] in extensions.conf or context=incoming in iax.conf and [incoming] in extensions.conf |
| 02:43.38 | TJNII | Okay. I misunderstood. |
| 02:44.10 | TJNII | We're confusing [context] and context= I think. |
| 02:44.25 | joat | think so too |
| 02:45.32 | jaytee | the context= line in your iax.conf or your sip.conf are "pointers" to the context in your extensions.conf file where the call will "enter" the dialplan. |
| 02:46.14 | jaytee | if you don't have it set then the call goes to the default context (if there is one) and if there's no match the call fails. |
| 03:00.51 | TJNII | needs a roll of packing tape |
| 03:00.59 | TJNII | Off to Wal-mart, I guess |
| 03:06.10 | ta^3 | Just wonder, what does 'overlap dialling' stands for? (allowoverlap=yes/no) |
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| 03:14.28 | nny_1 | so wait catching up on a night of conversation |
| 03:15.37 | nny_1 | i have quite a few contexts that use s |
| 03:15.50 | nny_1 | may pb my normal configs this weekend and let someone laugh at them |
| 03:16.45 | nny_1 | probably be rise to re read context and extension e |
| 03:16.47 | nny_1 | ideals* |
| 03:17.27 | nny_1 | ok not so bad, my only "s" used is for macros etc |
| 03:17.47 | nny_1 | all my normal stuff does exten => 999,1 999,2 etc |
| 03:18.25 | nny_1 | still may pb my config at one point.. maybe pay someone to give me some kind of 10 minute peer review.. dunno.. always trying to improve |
| 03:18.30 | jaytee | if you're not supposed to use s then 1) why does it exist and 2) what would you use in place of it in a macro? |
| 03:18.51 | nny_1 | i think the macro is supposed to use it |
| 03:19.00 | nny_1 | because you are calling the macro from another function |
| 03:19.06 | nny_1 | er context |
| 03:19.27 | jaytee | I was thinking of hiring [TK]D-Fender to do that with my dialplan but then I figured it would be wrong to deprive the channel of his expertise by giving him an aneurysm. |
| 03:19.43 | TJNII | Heh |
| 03:20.22 | nny_1 | here's my example |
| 03:20.23 | nny_1 | http://pastebin.com/m6bbe75e8 |
| 03:20.28 | nny_1 | thats just snippets |
| 03:20.47 | nny_1 | i have made some other macros to dial a cell phone after X rings.. or ring 1 extension for X seconds than another etc |
| 03:21.03 | jaytee | pfffft, mine makes that look like a Dick and Jane reader |
| 03:21.30 | nny_1 | ew those are old |
| 03:21.39 | nny_1 | i have u before voicemail on that one still |
| 03:21.41 | *** join/#asterisk Assimilate (n=Assimila@66.236.88.194.ptr.us.xo.net) |
| 03:21.44 | nny_1 | ha check this out then |
| 03:22.18 | nny_1 | this one is kind of not done |
| 03:22.18 | nny_1 | http://pastebin.com/m588f2c69 |
| 03:22.20 | jaytee | I've got my * server connected to 2 T-1's. One comes from my Nortel Meridian pbx and the other goes to my telco provider. I have 4 digit dialing between the 2 pbx's by using "tricks" in my asterisk dialplan and in Nortel programming. |
| 03:22.50 | nny_1 | sounds cool |
| 03:23.40 | nny_1 | ok enough geek.. naked woman in the house.. more trash talking later.. |
| 03:23.45 | jaytee | and all my calls route through sipX proxy to Exchange 2007 Unified Messaging for voicemail and for our autoattendant, although the autoattendant is front-ended with an IVR using LumenVox speech recognition for other menu options. |
| 03:24.32 | nny_1 | nice.. i build multiple custom smaller systems.. so I get a little of this or that depending on the needs |
| 03:24.41 | nny_1 | 20-30 phones or so.. next project is 1600 tho |
| 03:24.59 | nny_1 | ok naked woman staring at me.. daslkdsl;a poieerkl;fsdk <CARRIER LOST> |
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| 03:51.24 | UnixDog | thinks that its time to un duct-tape asterisk and see what happens |
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| 04:35.21 | kash | http://factorydirect.ca/catalog/product_spec.php?pcode=MO3200 |
| 04:35.21 | kash | LOL |
| 04:38.58 | coppice | It might be a collectors item when Motorola goes under :-) |
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| 05:18.59 | C4colo | anyone know of an eggdrop script for meetme monitoring? |
| 05:44.26 | jameswf-home | signs you do too mutchnalling with asterisk ... you cant remember if its signaling or sig |
| 05:44.43 | jameswf-home | s/sig/signalling |
| 06:06.03 | rabelais | I have an interesting echo problem that I can't isolate within my network |
| 06:06.52 | rabelais | I have a PSTN line connected from a linksys 3102 (sipura 3000) |
| 06:07.45 | rabelais | in most cases, there is no echo going out through that line from clients that connect |
| 06:08.22 | rabelais | there is only one client (nokia E51) that has an echo problem, but it is only when going out through that route |
| 06:08.57 | rabelais | if the E51's call is routed via vitelity to the PSTN network, there is no echo |
| 06:10.00 | bkw_ | rabelais: which end has the echo |
| 06:10.19 | bkw_ | the end where asterisk is or the far end where the caller is? |
| 06:10.20 | rabelais | bkw_, the E51 hears the echo |
| 06:10.33 | bkw_ | is the E51 connected to asterisk? |
| 06:10.36 | rabelais | yes |
| 06:10.48 | bkw_ | if you hear the echo most likely the problem is on the far side |
| 06:10.55 | bkw_ | does it happen all the time? |
| 06:10.59 | bkw_ | or just sometimes? |
| 06:11.01 | rabelais | only with the E51 |
| 06:11.08 | rabelais | with other clients, there is no echo |
| 06:11.15 | bkw_ | that is an odd one |
| 06:11.23 | bkw_ | does it happen sip to sip via the same asterisk box? |
| 06:11.26 | rabelais | and it only happens if the E51 is going through the PSTN device, otherwise there is no echo |
| 06:11.33 | rabelais | no, I don't believe so |
| 06:11.39 | bkw_ | check it out and see |
| 06:11.46 | bkw_ | could be bad hardware |
| 06:12.08 | bkw_ | ever since RoHS compliance .. things tend to go bad faster |
| 06:12.18 | rabelais | but the craziest thing....it used to be fine |
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| 06:13.28 | Qwell | bkw_: O.o |
| 06:13.39 | rabelais | ok, I just tried a call from E51 -> asterisk -> x-lite client on laptop |
| 06:13.41 | bkw_ | rabelais: you dropped it didn't you |
| 06:13.45 | Qwell | seriously? |
| 06:13.58 | rabelais | call was crystal clear, no echo |
| 06:14.06 | bkw_ | Qwell: how are you doing shortie? |
| 06:14.08 | rabelais | and no, I did not drop it |
| 06:14.12 | Qwell | decent |
| 06:14.20 | bkw_ | rabelais: I was just joshing with you :P |
| 06:14.32 | rabelais | :P |
| 06:14.37 | bkw_ | I have been playing with EC2 all day. |
| 06:14.53 | bkw_ | I have yet to master the whole creating my own AMI from scratch.. I keep missing something critical |
| 06:15.04 | Qwell | AMI? |
| 06:15.10 | bkw_ | Amazon Machine Image |
| 06:15.15 | Qwell | right |
| 06:15.26 | bkw_ | then to top it off I can't sleep |
| 06:15.29 | bkw_ | not that I don't wanna |
| 06:15.29 | Qwell | aren't there tools to convert from a vmware image or something? |
| 06:15.38 | bkw_ | I think so but where is the fun in that? |
| 06:15.38 | Qwell | one would imagine so |
| 06:15.46 | Qwell | make it easy |
| 06:15.57 | bkw_ | I have also created fedor, ubuntu, gentoo and centos OpenVZ templates |
| 06:16.03 | C4colo | AMI in #asterisk means something different |
| 06:16.11 | bkw_ | C4colo: yes I know that :P |
| 06:16.14 | Qwell | C4colo: I'm pretty sure he's aware |
| 06:17.14 | C4colo | yea, I just like pointing out the obvious |
| 06:17.22 | C4colo | multiple times if I think I can get away with it |
| 06:18.09 | bkw_ | Qwell: have you guys tried to live migrate an asterisk Xen instance from machine to machine? |
| 06:18.14 | rabelais | thoughts anyone? I don't even know what to debug...independently, everything is working... |
| 06:18.24 | Qwell | bkw_: never used Xen |
| 06:18.33 | bkw_ | I can tell you it doesn't work |
| 06:18.42 | plik | is there a way I can expand a variable in the dial plan, pipe it thru sed, and set the output as another variable ? |
| 06:18.54 | Qwell | plik: in 1.6 there is func_shell |
| 06:19.02 | plik | and in 1.4? |
| 06:19.14 | Qwell | Set(foo=${SHELL(echo bar)}) |
| 06:19.16 | bkw_ | Qwell: if you have two boxes... and you transfer from one to the other.. as long as the clock is ahead of the current one its fine.. but if you do the reverse of that it hangs forever |
| 06:19.42 | bkw_ | if the machine's ram is low say 256 megs of ram you only get a small blip in the audio stream |
| 06:19.47 | Qwell | that sounds like a Xen issue. I suspect that wouldn't happen by just setting the clock backwards. |
| 06:19.47 | C4colo | whould System() do what you are looking for? |
| 06:19.51 | Qwell | well, maybe. dunno |
| 06:19.54 | bkw_ | Qwell: no its not a xen issue |
| 06:19.55 | Qwell | C4colo: System doesn't return a var |
| 06:19.56 | bkw_ | trust me |
| 06:20.07 | plik | C4colo: no, tried that :( |
| 06:20.18 | Qwell | C4colo: there might be a backport of func_shell... sec |
| 06:20.19 | C4colo | ah |
| 06:20.25 | Qwell | erm, plik |
| 06:20.29 | plik | ta |
| 06:20.50 | Qwell | hmm, nope |
| 06:20.56 | Qwell | shouldn't be too difficult to do though. |
| 06:21.01 | plik | bah... time for an upgrade then |
| 06:21.37 | plik | cheers though |
| 06:21.50 | plik | goes away to think of other possibilities |
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| 07:37.20 | cplx | hi guys.. anyone here pretty full bottle with a CME or callmanager & Asterisk integration? |
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| 09:06.56 | *** join/#asterisk ibnolqaiyem (n=ibnolqai@41.196.251.156) |
| 09:08.35 | ibnolqaiyem | if i want to make asterisk as pbx for my company , and i have many analog phones , what is device i need? |
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| 09:12.02 | C4colo | ibnolqaiyem, there are lots of devices that provide FXS ports |
| 09:12.13 | C4colo | depending on how many you have and how you want to connect them to the pbx |
| 09:12.27 | C4colo | analog cards are a good option |
| 09:12.38 | C4colo | there are also TDMoE and USB channel banks |
| 09:13.16 | C4colo | if you only have a few analog phones (4-6) using ATAs can be a cheap solution |
| 09:13.25 | ibnolqaiyem | C4colo, what you mean about (how you want to connect them to the pbx)? |
| 09:13.34 | C4colo | the cost per FXS port is about half on ATAs than on analog TDM cards |
| 09:13.47 | C4colo | well, do you want to use a PCI device, a USB device or an Ethernet device? |
| 09:14.32 | ibnolqaiyem | if i have many telephone, say 100 phones, what is good choice for me? |
| 09:14.39 | C4colo | buy SIP phones |
| 09:14.57 | C4colo | honestly, that many phones will be hard to integrate into a single pbx |
| 09:15.17 | C4colo | I think sangoma has cards that can be expanded up to 48 ports |
| 09:15.45 | C4colo | so two machines with one set of those cards each would give you 96 phones |
| 09:16.38 | C4colo | you would need to set up a trunk between the two servers though |
| 09:16.52 | C4colo | what country are you in? |
| 09:17.43 | ibnolqaiyem | C4colo, am i needing channel bank? |
| 09:18.01 | JT | pci cards are a shit option imho |
| 09:18.12 | JT | the best is a sip fxs gateway |
| 09:18.25 | JT | then t1 fxs channel bank next best |
| 09:18.41 | C4colo | yea, for that many I would use ethernet channel banks |
| 09:18.58 | C4colo | TDMoE or SIP trunks |
| 09:19.06 | JT | forget TDMoE |
| 09:19.11 | JT | SIP or T1 |
| 09:19.21 | C4colo | or forget TDMoE |
| 09:19.42 | C4colo | as I apparently have no idea what I'm saying |
| 09:19.45 | JT | TDMoE is not widely adopted |
| 09:19.50 | C4colo | I'll let you take this one JT |
| 09:19.51 | JT | so it's a bit limiting |
| 09:20.16 | JT | sorry i didn't mean to be an arse, just my opinion on the situation |
| 09:20.27 | C4colo | my opinion is to buy SIP phones |
| 09:20.36 | JT | well that's mine too |
| 09:20.40 | *** join/#asterisk Vec (n=Vec@78-86-163-97.zone2.bethere.co.uk) |
| 09:20.58 | JT | but sometimes there are niche cases where it's ok to keep analogue, i have no idea if this applies to ibnolqaiyem or not |
| 09:21.36 | C4colo | lets see 4x Rhino 24-port channel banks = 7584 |
| 09:21.38 | C4colo | usd |
| 09:22.17 | JT | sip? |
| 09:22.45 | C4colo | or 96 aastra 51i = 11520 usd |
| 09:23.07 | JT | C4colo: are those sip gateways or T1 channel banks? |
| 09:23.31 | ibnolqaiyem | C4colo, JT , are you mean i need channel bank and SIP or T1? |
| 09:23.56 | C4colo | ah 1t |
| 09:23.57 | C4colo | t1 |
| 09:23.59 | JT | ibnolqaiyem: option is either a SIP to FXS gateway .... OR a T1 channel bank and T1 PCI card |
| 09:24.19 | C4colo | I thought voipsupply had a better selection of channel banks |
| 09:24.25 | C4colo | all I'm coming up with are the rhinos |
| 09:24.32 | JT | the best T1 channel banks can be bought on ebay |
| 09:24.36 | JT | Adtran or Adit |
| 09:24.40 | JT | or CAC |
| 09:26.59 | *** join/#asterisk mandh (n=mandh@82.137.216.38) |
| 09:27.25 | C4colo | here you go, a SIP 32port gateway for $3000 |
| 09:27.26 | mandh | Hello |
| 09:27.38 | JT | what brand? |
| 09:27.41 | C4colo | so 3 of those for 9,000 usd |
| 09:27.43 | C4colo | Patton |
| 09:27.57 | JT | the 24 ports are closer to $1500 iirc |
| 09:27.57 | C4colo | I can't beleive the selection on these sites is so limited |
| 09:28.08 | JT | Audiocodes or Mediatrix |
| 09:28.53 | C4colo | yea, well I would be quoting those prices if I found them |
| 09:29.02 | C4colo | this is what they are giving me... lol |
| 09:29.59 | C4colo | 24-port audiocodes for 1300 |
| 09:30.30 | C4colo | sip |
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| 09:30.50 | C4colo | 4x of those is about $5500 |
| 09:31.08 | JT | http://www.telephonydepot.com/product_p/105-066-124-fxs.htm |
| 09:31.29 | C4colo | yep |
| 09:31.33 | C4colo | exact page I was looking at |
| 09:32.02 | C4colo | was odd to see it pop on top of the other one |
| 09:32.23 | C4colo | like a rift in the space-time continum |
| 09:32.26 | JT | the mediatrix unit isn't showing on telephonydepot |
| 09:33.39 | C4colo | anyway, enough of those would give you the fxs ports you need |
| 09:33.56 | C4colo | it may be cheaper to go that way, however sip phones are not that much more and would provide much more functionality |
| 09:34.52 | C4colo | you probably want to have someone who does this sort of thing come out and take look at the site and provide an expert opinion and an estimate for the upgrades |
| 09:35.12 | C4colo | an installation with 100+ phones is a big task for a first project |
| 09:39.51 | ibnolqaiyem | if i have ip phone , will i need channel bank? |
| 09:40.22 | JT | no |
| 09:40.25 | JT | ethernet switch |
| 09:40.34 | JT | preferably ethernet switch with PoE support |
| 09:40.53 | ibnolqaiyem | ok, which case that i must have t1 channel bank? |
| 09:41.01 | C4colo | zyxel sells 24-port power over ethernet switches |
| 09:41.18 | C4colo | if your service coming from the phone company is T1 then yes |
| 09:41.20 | JT | ibnolqaiyem: if you use analogue phones |
| 09:41.45 | C4colo | SIP phones are ethernet devices, like a computer, they just go on the network and find their wy back to the asterisk server over the network |
| 09:42.04 | C4colo | there is no other devices required between the SIP phones and the asterisk server other than normal ethernet network equipment |
| 09:42.38 | C4colo | if you already have a network built to provide network to the computers, most SIP phones have two ethernet ports and you can use the existing wiring providing network to the computer to also feed the phones |
| 09:42.57 | ibnolqaiyem | what is difference btw sip fxs gateway & t1 channel bank? |
| 09:43.04 | C4colo | this can reduce the requirements for network equipment upgrades |
| 09:43.12 | C4colo | t1 channel bank needs a T1 card in the pbx |
| 09:43.26 | C4colo | sip channel banks use Ethernet and do not need anything but to be plugged into the network |
| 09:43.30 | JT | ibnolqaiyem: sip gateway converts to voip with an ethernet port |
| 09:44.01 | JT | t1 channel bank converts fxs ports to channelised RBS on a T1 port |
| 09:44.33 | C4colo | how do your phone lines come from the telephone company? |
| 09:45.07 | ibnolqaiyem | if i don't need voip , what is good solution for me, sip gateway or t1 channel bank? |
| 09:45.15 | ibnolqaiyem | for pbx only |
| 09:45.44 | JT | ibnolqaiyem: you need to explain the scenario better. sip phones are generally a better choice than analogue phones |
| 09:47.03 | C4colo | how do your phone lines come from the telephone company? |
| 09:47.33 | ibnolqaiyem | C4colo, can you explain more? |
| 09:47.46 | C4colo | your dialtone, how do you get it? |
| 09:47.53 | C4colo | outbound calls |
| 09:47.59 | C4colo | phone lines |
| 09:48.09 | JT | PHONES <---> PBX <---> ??? |
| 09:48.11 | C4colo | how are they provided to you by your telephone company? |
| 09:48.32 | C4colo | do you have a T1, E1, J1, PRI, BRI, Analog lines, SIP trunks? |
| 09:48.45 | ibnolqaiyem | C4colo, analog line |
| 09:48.51 | C4colo | ok, how many of them? |
| 09:48.56 | ibnolqaiyem | one |
| 09:49.01 | ibnolqaiyem | sorry |
| 09:49.01 | C4colo | for 100 people? |
| 09:49.04 | ibnolqaiyem | 2 lines |
| 09:49.11 | JT | for 100 handsets? |
| 09:49.15 | C4colo | 100 people are sharing two lines? |
| 09:50.31 | C4colo | is most of the calling going between extensions on the same network? |
| 09:51.10 | C4colo | another question is, do all 100 phones need their own extension number? |
| 09:51.11 | ibnolqaiyem | C4colo, i am not professional in phone system |
| 09:51.21 | ibnolqaiyem | sorry |
| 09:51.31 | ibnolqaiyem | i am plan to do that |
| 09:51.39 | ibnolqaiyem | thus i ask |
| 09:51.45 | C4colo | you probably have what is commonly called a roll-over group |
| 09:51.58 | C4colo | with many phone lines, but only two phone numbers |
| 09:53.15 | ibnolqaiyem | C4colo, how many phone lines you suggest to server 100 phones? |
| 09:55.01 | C4colo | it depends on the usage |
| 09:55.27 | C4colo | if you are in sales and lots of calls will be going in and out I would do about 20-40 lines |
| 09:55.41 | C4colo | if most people only make a few calls a day 10 - 20 |
| 09:56.06 | C4colo | if you are a call center and everyone will be on the phone at the same time I would suggest 90-100 |
| 09:56.35 | C4colo | but it also depends on if you will be using SIP trunks too |
| 09:56.45 | C4colo | it all depends on your usage |
| 09:56.57 | C4colo | the type of business you are and what you will need |
| 09:57.12 | C4colo | for a hotel you could probably serve 100 rooms with 6-10 lines |
| 09:59.28 | ibnolqaiyem | C4colo, JT thank you very much |
| 09:59.47 | C4colo | sounds like you need to get some more information together and do some more research |
| 09:59.57 | ibnolqaiyem | yes |
| 10:00.00 | C4colo | moving to VoIP is not a quick or easy move |
| 10:00.07 | C4colo | it needs to be thought out and planned well |
| 10:00.28 | ibnolqaiyem | what is resource you suggest for information? |
| 10:00.48 | C4colo | honestly, someone who is an expert locally would be very valuable |
| 10:01.11 | C4colo | or if you cant' find one, someone who would be willing to travel |
| 10:01.52 | C4colo | I say this because someone who does VoIP phone systems for a living will have much experience with the technology and know the best thing for your situation |
| 10:02.30 | C4colo | to learn it well, you need time and start with small projects and work your way up to bigger ones |
| 10:03.56 | C4colo | I need to sleep, I wish you good luck on your project |
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| 10:10.03 | ibnolqaiyem | C4colo, thank you , i hope good dream for you |
| 10:21.07 | gramulhaozin | hey guys |
| 10:21.15 | gramulhaozin | anyone have had trouble with no audio on IVR's |
| 10:21.20 | gramulhaozin | but regular audio on calls |
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| 10:28.17 | JT | ibnolqaiyem: also take a look at: |
| 10:28.21 | JT | ~thebook |
| 10:28.21 | jbot | 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://tfot.leifmadsen.com |
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| 10:58.07 | Linker3000L | GXP2000 question for someone.... |
| 10:58.55 | Linker3000L | If I program a speed dial button, it always uses Account 1 even if I set the option to 'account n'. Can you prefix the number with something that forces a specific account (1-4) to be used? Thanks |
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| 11:08.01 | tzafrir_laptop | gramulhaozin, no audio from pressing digits? |
| 11:08.06 | tzafrir_laptop | You use SIP? |
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| 15:26.37 | kimc | trying to debug and AGI with core set debug 50 |
| 15:27.05 | kimc | this generates lots of output not directly related to the AGI |
| 15:27.44 | kimc | wonder how I can do this better with less output.. |
| 15:28.19 | kimc | like without the ouput related to the SIP processing assoctiated with the test call |
| 15:28.59 | styelz | set verbose instead of debug ? |
| 15:29.11 | kimc | good lets try that |
| 15:29.16 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 15:29.54 | outtolunc | ut oh, you woke him up <G> |
| 15:30.08 | styelz | runs away |
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| 15:32.49 | styelz | plugs in |
| 15:42.16 | *** join/#asterisk raz (n=y@unaffiliated/raz) |
| 15:42.52 | raz | i have set up asterisk with sip, now how can people call me directly? my-extension@myip? :) |
| 15:44.39 | [TK]D-Fender | raz: "allowguest=yes" and "context=somewhere" under [general] in sip.conf |
| 15:44.55 | raz | ok thx :) |
| 15:45.44 | eXistenZ | hey [TK]D-Fender |
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| 15:56.49 | angryuser | [TK]D-Fender i have 3 account from same provider to call out, they are limited toone communication at the same time, i have used a scripts with group() check and gotoif to manage automaticly with what peer to call out, is there any other way of doing that ? |
| 15:58.50 | angryuser | lets say if i group=1 inside sip.conf for that 3 peers, can i call out with that group and lets * to manage ? like (ZAP/g) groups ? |
| 15:59.13 | [TK]D-Fender | angryuser: No. "group" is only for zap. |
| 15:59.33 | [TK]D-Fender | angryuser: If they only allow 1 call each, just dial them back to back. Each that refuses will just fall to the next |
| 16:00.24 | angryuser | well dynamic i am setting a group beore call, and veriying a number in the group; works fine |
| 16:00.58 | angryuser | and faster imo |
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| 16:05.48 | [TK]D-Fender | angMy way takes 3 lines, no variables, no race conditions or need to care. |
| 16:06.53 | angryuser | yep, but each call takes more time |
| 16:07.09 | *** join/#asterisk h0 (n=fakhir@unaffiliated/fakhir) |
| 16:07.30 | [TK]D-Fender | angryuser: As answers go, "No" comes pretty quick. |
| 16:08.25 | angryuser | localy, even faster |
| 16:12.45 | *** join/#asterisk rbd (n=rbd@adsl-074-229-183-112.sip.rmo.bellsouth.net) |
| 16:13.22 | tsikis | hello there i have setup my asterisk it sends and receives calls using the gw0 but only sends calls using the sip provider(proxy01.sipphone.com) any help why it aint receiving?) |
| 16:13.56 | rbd | hi, I'd like to have an entry in my extensions that matches any ext between 10000 and 99999999 .... for 10000 to 99999 I can do something like _ZXXXX ...can I do _ZX{4,7} ? |
| 16:14.07 | ManxPower | tsikis: we have no idea what you are talking about. Asterisk has no interface called "gw0" |
| 16:14.37 | ManxPower | rbd: you can't do that. |
| 16:14.43 | tsikis | sorry i have a spa3000 |
| 16:14.57 | rbd | ManxPower: I thought so (didn't see {} in the docs) ... then what would you advise? |
| 16:14.58 | ManxPower | these are not regexs |
| 16:15.20 | rbd | ok |
| 16:15.53 | ManxPower | rbd: seems simplier to just to a generic match, if the dialplan does not match anything else it will match the generic _ZXXX.,1,Whatever |
| 16:16.33 | tsikis | so this is of no importance the problem is why i cannot receive sip calls while i can send through asterisk after registering with sipphone but not receive |
| 16:16.37 | ManxPower | I do something similar to catch dialed extensions that are not valid |
| 16:16.38 | rbd | hmm, the dialplan is somewhat complex, so I'd rather try to explicitly match everywhere to avoid bug chasing wherever possible... would I have to split it out then? like _ZXXXX, _ZXXXXX, _ZXXXXXX? |
| 16:16.51 | ManxPower | rbd: ALL dialplans are complex. |
| 16:17.15 | rbd | more complex than most, then |
| 16:18.13 | rbd | yeah, actually, I think _ZXXXX. will work...as all other extns in the dialplan are 4 digits |
| 16:18.15 | rbd | thanks |
| 16:24.10 | ManxPower | rbd: IIRC, the order is "most specific match in current context, then search included contexts in the oder specified in extensions.conf. Other than that order withing a context means nothing. |
| 16:25.34 | *** join/#asterisk mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net) |
| 16:25.34 | *** mode/#asterisk [+o mog] by ChanServ |
| 16:26.01 | tsikis | hello there again i will rephrase the question i can send voip calls by registering with the sip provider but i cannot receive? any ideas that would be helpfull? |
| 16:26.14 | *** part/#asterisk UnixDog (n=UnixDog@181.128.204.68.cfl.res.rr.com) |
| 16:27.38 | angryuser | twisted sendis voip calls and registering is not the same thing |
| 16:28.10 | angryuser | twisted 'sip show registry' under cli and you will see the status of you registration |
| 16:29.08 | tsikis | sip show registry does not show anything but sip show peers shows the sip registered |
| 16:34.35 | angryuser | tsikis sip show peers show peers registered inside YOUR network |
| 16:35.21 | tsikis | ok so sip show registry doesnt show anything |
| 16:35.37 | angryuser | tsikis so you are not registered, and no trying to |
| 16:35.44 | angryuser | not* |
| 16:36.14 | angryuser | tsikis place registry string under [general] in sip.conf |
| 16:38.58 | tsikis | ok now i got a line with unregistered (isnt asterisk trying to register 2 times?) |
| 16:41.21 | angryuser | tsikis have no idea |
| 16:41.28 | tsikis | dude you rock it worked i had it in a global made by me not in general |
| 16:41.29 | tsikis | thanks |
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| 17:11.10 | tsikis | ok me again i want to send the receiving call to a voip number i tried changine my extention to exten => s,1,Dial(SIP/sipphonenumber:password@proxy01.sipphone.com, 20) but it aint working any ideas? |
| 17:14.56 | [TK]D-Fender | tsikis: go setup a proper peer entry for them in sip.conf and stop dialing direct like that |
| 17:15.21 | tsikis | ok |
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| 17:26.33 | baliktad | I'm a little confused about setting up an asterisk box on my lan to provide PBX-like services to VoIP phones |
| 17:26.49 | baliktad | Can/does Asterisk function as a SIP router? |
| 17:27.55 | rabelais | I'm not sure what you mean by "sip router" |
| 17:28.15 | rabelais | but if you mean something to handle your sip clients and connect them to sip services, then yes |
| 17:28.42 | WilliamK | and/or tdm based services |
| 17:29.32 | baliktad | Currently I use a dd-wrt router, it includes a service called Milkfish (formerly SIPath) that functions as a SIP router, meaning it functions as my outbound proxy and also opens up ports for the RTP streams when necessary |
| 17:29.58 | *** join/#asterisk thansen|laptop (n=thansen@19.243.sfcn.org) |
| 17:30.09 | rabelais | asterisk is a bit more featureful than that |
| 17:30.16 | baliktad | I understand that |
| 17:30.31 | baliktad | I want to know if it replaces the Milkfish functionality |
| 17:30.35 | rabelais | that's more like a sip proxy + nat tweaker |
| 17:30.41 | baliktad | that's exactly what it is |
| 17:30.57 | baliktad | but it's perfect when you just have a handful of VoIP phones inside a lan |
| 17:30.58 | rabelais | asterisk does not touch routing tables |
| 17:31.23 | rabelais | asterisk should not be run as root, so it does not have access to the routing tables |
| 17:31.34 | *** join/#asterisk uTx (n=unix@modemcable074.229-82-70.mc.videotron.ca) |
| 17:31.35 | baliktad | well I wouldn't be able to run asterisk on the router anyway |
| 17:31.45 | Pagautas | hi there, when we're supposed to see asterisk-1.6 release?? |
| 17:32.07 | rabelais | but it's somewhat safer to let asterisk be the sip gateway, handling and routing calls as a middleman |
| 17:32.18 | baliktad | so how exactly do the RTP streams work if I were to set up an asterisk box on the LAN |
| 17:32.39 | rabelais | net call -> asterisk -> lan client |
| 17:32.46 | rabelais | asterisk stays in the media path |
| 17:32.48 | baliktad | Ahh |
| 17:32.57 | baliktad | I have to manually forward all the ports to asterisk |
| 17:33.05 | Qwell | ~sipnat |
| 17:33.06 | jbot | [~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 |
| 17:33.31 | baliktad | thanks Qwell, I will read those resources, it sounds like what I want to do |
| 17:33.38 | *** join/#asterisk Katty (n=angela@adsl-75-12-158-16.dsl.stlsmo.sbcglobal.net) |
| 17:33.52 | Katty | herro |
| 17:33.59 | Qwell | ohai |
| 17:34.08 | rabelais | that would be the easiest way, unless you want to setup a stateful firewall that will dynamically tweak ports and whatnot, similar to milkfish |
| 17:34.46 | rabelais | unless you have an obvious objection to letting asterisk stay in the media path, it's the easiest way to go |
| 17:35.28 | baliktad | It's a small enough setup that I should be fine keeping asterisk in the middle |
| 17:36.18 | rabelais | mind you, you'll have to change your clients to register to the asterisk server now, instead of registering to whatever server/service they were registering to before |
| 17:36.35 | rabelais | and now asterisk will register on the clients' behalf |
| 17:38.14 | baliktad | hmm |
| 17:38.47 | rabelais | otherwise nothing really changes... |
| 17:38.58 | rabelais | asterisk can't just hijack SIP traffic |
| 17:39.03 | *** join/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 17:39.07 | rabelais | it has to be explicitly placed in the middle |
| 17:39.15 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 17:39.34 | grandpapadot | Hi all. What's the best MoH player for 1.4? mpg123 or madplay? And does the 1.4 internal MoH player resume where it left off or just start the track over for every instance? |
| 17:40.54 | *** join/#asterisk hi365_m (n=hi365@213.151.61.251) |
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| 17:43.50 | *** mode/#asterisk [+o russellb] by ChanServ |
| 17:44.16 | grandpapadot | ~centos52bug |
| 17:44.16 | jbot | [~centos52bug] There is a bug compiling Zaptel up to 1.4.11/1.2.26 on CentOS/RHEL 5.2: error in xpp/xdefs.h. Workaround: uncheck "xpp" (Astribank) driver in menuselect. Fixed in SVN. Patches: http://bugs.digium.com/12889 |
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| 17:47.39 | [TK]D-Fender | baliktad: * is NOT a proxy, it is a B2BUA. You can have * reg to your providers, and your internal phones register to it. * sits in the middle and accepts calls from either and processes them how you want |
| 17:48.14 | russellb | i'd rather make asterisk a face to face UA instead of back to back. |
| 17:48.27 | russellb | goes to write the patch |
| 17:48.50 | baliktad | I'm starting to get that [TK]D-Fender |
| 17:49.41 | [TK]D-Fender | baliktad: What are you using as internal endpoints? |
| 17:49.45 | Qwell | jbot: logs |
| 17:49.46 | jbot | it has been said that logs is http://s12/ |
| 17:49.57 | baliktad | Linksys SPA-921's |
| 17:50.29 | [TK]D-Fender | baliktad: In your current scenario your upstream provider does everything. Controls what you dial, and your phones become quite "dumb". |
| 17:51.24 | [TK]D-Fender | baliktad: With * you can do all sorts of things all by yourself and use your provider merely as a termination point if you like. Also let you have * pick the best of several choices, provide voicemail and a million other things |
| 17:51.39 | baliktad | The phones are actually quite configurable, you can use custom dial plans and point at your own VM service, etc. |
| 17:52.02 | jaytee | except make coffee but I'm working on an channel module for that :-) |
| 17:53.26 | baliktad | but I am looking for some more flexibility for routing unanswered calls (time-based forwarding) which is why I am considering asterisk |
| 17:55.15 | [TK]D-Fender | baliktad: An issue like that is the phone has to be responsive to do that forwarding for you. |
| 18:04.21 | baliktad | Default rtp.conf uses 10000 open ports in my firewall, is there a better recommendation for very low usage scenarios? |
| 18:04.42 | ManxPower | baliktad: Each call happening around the same time needs two ports |
| 18:05.00 | ManxPower | so if you never have more than 10 calls, then allocate 20 ports. |
| 18:05.25 | baliktad | that I can handle :) |
| 18:05.26 | ManxPower | Personally I'd allocate 4x the number of ports as the number of calls you expect to have, just to be safe and handle spikes. |
| 18:05.37 | ManxPower | 10 calls, 40 ports |
| 18:05.56 | *** join/#asterisk StooJ_ (n=stooj@stooj.plus.com) |
| 18:07.11 | drmessano | ~RTParanoia |
| 18:07.12 | jbot | well, rtparanoia is [RTParanoia]: 10000 ports, 1 port, who cares.. it's the same daemon listening! |
| 18:07.48 | baliktad | I care that my firewall has to account for all the other services on my lan :) |
| 18:08.30 | drmessano | It's the same daemon |
| 18:08.45 | baliktad | I understand |
| 18:08.53 | drmessano | Yep |
| 18:23.29 | angryuser | i want to set a value in astdb 'Set(DB(internet/connected)=0 or 1 how can i do it from external? asrerisk -rx ? |
| 18:25.19 | icel | can anyone tell me how i would edit my zapata.conf and zaptel.conf to add another pri span to my digium card? |
| 18:25.23 | icel | http://pastebin.ca/1083636 |
| 18:27.51 | *** join/#asterisk tobias (n=tobias@user-0ce2hvs.cable.mindspring.com) |
| 18:27.55 | [TK]D-Fender | angryuser: "help database' |
| 18:29.46 | [TK]D-Fender | icel: http://pastebin.ca/1083640 |
| 18:30.13 | icel | D-Fender: thx |
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| 18:32.41 | icel | [TK]D-Fender: so I need pri_net for the 2nd PRI, I can just specify that under the group line? |
| 18:32.44 | jordanb | Is there a way to make asterisk never time the dial tone out, no matter how long the phone is off-hook? |
| 18:33.05 | [TK]D-Fender | icel: Yup |
| 18:33.12 | icel | thx |
| 18:33.25 | [TK]D-Fender | jordanb: What EXACTLY is providing the tone? |
| 18:33.40 | jordanb | [TK]D-Fender; A TDM400P card. |
| 18:34.31 | [TK]D-Fender | jordanb: I don't believe it is configurable on those.... |
| 18:34.42 | jordanb | :< |
| 18:35.33 | tzafrir_laptop | icel, you can regenerate working ones with zapconf / genzsaptelconf |
| 18:35.51 | icel | k |
| 18:35.58 | icel | tzafrif_laptop: thx |
| 18:36.04 | tzafrir_laptop | genzaptelconf, actually |
| 18:36.21 | angryuser | thnk fender |
| 18:36.29 | tzafrir_laptop | jordanb, on FXS? |
| 18:37.48 | jordanb | tzafrir_laptop; Yeah, the phone is plugged into a FXS port. |
| 18:38.25 | jordanb | The card also has one FXO module on it. |
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| 20:51.23 | Katty | hmm |
| 20:57.04 | *** join/#asterisk UnixDog (n=UnixDog@181.128.204.68.cfl.res.rr.com) |
| 20:57.07 | UnixDog | hey all |
| 21:02.54 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 21:03.03 | UnixDog | configure: error: *** termcap support not found |
| 21:03.08 | UnixDog | .6 I get this error compiling asterisk-1.6 I get this error |
| 21:03.18 | UnixDog | thatwas out of order |
| 21:04.02 | UnixDog | but termcap is installed |
| 21:04.15 | [TK]D-Fender | UnixDog: termcap-devel , libtermcap |
| 21:04.25 | *** join/#asterisk G3R4 (n=geraldor@189.71.33.39) |
| 21:06.58 | UnixDog | libtermcab-devel |
| 21:07.02 | UnixDog | installing now |
| 21:08.30 | UnixDog | got it it configured |
| 21:10.59 | UnixDog | ************************************************** |
| 21:10.59 | UnixDog | *** Install ncurses to use the menu interface! *** |
| 21:10.59 | UnixDog | ************************************************** |
| 21:10.59 | UnixDog | menuselect changes NOT saved! |
| 21:11.13 | UnixDog | and ncurese and ncurses-devel is installed |
| 21:12.46 | UnixDog | any ideas |
| 21:12.56 | UnixDog | this is 1.6-beta9 |
| 21:17.09 | [TK]D-Fender | UnixDog: trash & reextract |
| 21:18.20 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 21:18.40 | UnixDog | the asterisk src |
| 21:18.46 | UnixDog | ok |
| 21:22.20 | UnixDog | this is on a vps |
| 21:24.50 | UnixDog | ok that worked. |
| 21:25.10 | UnixDog | clear |
| 21:25.11 | UnixDog | ls |
| 21:25.33 | L|NUX | hey can some one tell me why this happens that one sip account will register on soft phone but not registering on asterisk ? |
| 21:26.27 | [TK]D-Fender | L|NUX: because you set it up wrong. |
| 21:26.38 | carrar | How can that be!!! |
| 21:26.48 | L|NUX | [TK]D-Fender: well no thats not the case |
| 21:27.01 | carrar | obviously |
| 21:27.09 | [TK]D-Fender | L|NUX: if it was right, it'd have worked. |
| 21:27.09 | L|NUX | provider gives user informaiton like htis |
| 21:27.10 | L|NUX | this |
| 21:27.44 | L|NUX | [TK]D-Fender: user: 021Xxxxxx, auth user: 021xxxxxx@domain.com, pass: xxxxxx, proxy: xxx.xxx.xxx.xxx port: 9060 |
| 21:27.53 | L|NUX | i used this register string |
| 21:28.02 | [TK]D-Fender | L|NUX: 9060? |
| 21:28.11 | L|NUX | register => user:pass@proxy:port/ext |
| 21:28.11 | [TK]D-Fender | L|NUX: and that isn't a register string |
| 21:28.12 | L|NUX | yes |
| 21:28.16 | L|NUX | yes i know |
| 21:28.31 | L|NUX | now i do not know how to send auth user with domain |
| 21:28.39 | L|NUX | i tried i have read voip-info page |
| 21:29.06 | [TK]D-Fender | L|NUX: "domain=asdgjhdg" |
| 21:29.30 | L|NUX | [TK]D-Fender: can you be with me ? |
| 21:29.56 | [TK]D-Fender | L|NUX: I don't do IRC relationships. |
| 21:30.02 | [TK]D-Fender | </sarcasm> |
| 21:30.05 | L|NUX | haha |
| 21:30.15 | L|NUX | what about IM :P |
| 21:30.22 | L|NUX | should i msg over IM :D |
| 21:30.24 | L|NUX | brb |
| 21:30.25 | [TK]D-Fender | L|NUX: transitive |
| 21:30.39 | *** join/#asterisk eXistenZ (n=existenz@unaffiliated/existenz) |
| 21:30.39 | L|NUX | not sending you msg over IM |
| 21:31.18 | rbd | is there an asterisk dialplan variable (or some way to get) the IP address on the interface for which the call came in on? |
| 21:32.12 | [TK]D-Fender | rbd: Go take a CLOSE look at the list of CHANNELS from CLI |
| 21:32.47 | rbd | ah yeah. I was thinking the SIP headers (it's coming in on a sip channel) but that's more direct |
| 21:33.21 | [TK]D-Fender | rbd: And you could try taking a close look at *'s functions. |
| 21:39.28 | rbd | [TK]D-Fender: I see SIPCHANINFO(recvip) but the description says "The source IP address of the peer." (not sure how that differs from SIPCHANINFO(peerip))... |
| 21:40.31 | rbd | I'm probably missing something obvious here, but I want the IP addr of the interface that took in the call on the same asterisk box, not of the peer |
| 21:41.31 | [TK]D-Fender | rbd: All calls are to *, not another device. * is a B2BUA after all. You might want to look at what your channels are BRIDGED to |
| 21:44.37 | L|NUX | [TK]D-Fender: i got this error Got SIP response 400 "Invalid From" back from 58.27.240.22 |
| 21:45.04 | L|NUX | register => 021800XXXX@wateen.net:PWDXXXXXXX@58.27.240.22:9060 |
| 21:45.47 | [TK]D-Fender | L|NUX: And you're doing such a glorious job showing us your broken register attempt too.... |
| 21:46.13 | L|NUX | [TK]D-Fender: umm |
| 21:46.16 | L|NUX | but its not working man |
| 21:46.20 | L|NUX | :( |
| 21:47.20 | L|NUX | any idea |
| 21:47.29 | [TK]D-Fender | L|NUX: Yeah, show us something useful. |
| 21:47.49 | L|NUX | [TK]D-Fender: like what you want me to show, should i pb sip debug ? |
| 21:48.32 | [TK]D-Fender | L|NUX: ummm DUH <------ |
| 21:48.58 | [TK]D-Fender | L|NUX: like a medical examiner after hours of deliberations saying "So you think I should actually look at the patient?" |
| 21:49.13 | L|NUX | [TK]D-Fender: hehe |
| 21:49.24 | *** join/#asterisk flewid (n=flewid@www.flewid.ca) |
| 21:49.24 | L|NUX | hold |
| 21:50.34 | L|NUX | [TK]D-Fender : http://rafb.net/p/FYGOFB79.html |
| 21:50.57 | L|NUX | but same password is working fine on x-lite, x-pro and others |
| 21:51.32 | [TK]D-Fender | L|NUX: Warning: 399 CS Domain not allowed |
| 21:51.42 | *** join/#asterisk korihor (n=korihor@190.39.163.45) |
| 21:52.08 | [TK]D-Fender | L|NUX: go comment out EVERYTHING overly specific from sip.conf |
| 21:52.32 | L|NUX | should i make new sip.conf ? |
| 21:52.46 | L|NUX | commenting whole sip.conf take too much time :P |
| 21:53.52 | [TK]D-Fender | ... |
| 21:54.22 | [TK]D-Fender | register => 021800XXXX:PWDXXXXXXX@58.27.240.22:9060 |
| 21:55.18 | L|NUX | ok |
| 21:55.21 | L|NUX | doing that |
| 21:58.42 | L|NUX | again same error |
| 21:58.52 | L|NUX | i am using register => 021800XXXX:PWDXXXXXXX@58.27.240.22:9060 now |
| 21:58.53 | [TK]D-Fender | PB |
| 21:59.00 | L|NUX | hold |
| 22:00.29 | L|NUX | http://rafb.net/p/MVBieL45.html |
| 22:02.59 | L|NUX | any thoughts |
| 22:03.55 | [TK]D-Fender | L|NUX: yes. Go look at your soft-phone registering and see the difference |
| 22:04.15 | L|NUX | should i paste debug of softphone ? |
| 22:05.41 | *** join/#asterisk d-k-t (n=dt@vpn21.witopia.net) |
| 22:06.00 | L|NUX | http://rafb.net/p/Gh6izo92.html |
| 22:08.06 | [TK]D-Fender | L|NUX: clue : realm="wateen.net" |
| 22:08.22 | L|NUX | i added that |
| 22:08.27 | L|NUX | in sip.conf |
| 22:11.27 | L|NUX | any thing else brother :) |
| 22:17.43 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:18.12 | L|NUX | if you notice To: <sip:wateen.net> is empty in asterisk |
| 22:18.13 | *** join/#asterisk trelane (i=trelane@lan.trelane.net) |
| 22:18.26 | *** join/#asterisk ctaloi (n=ctaloi@pool-72-90-82-34.syrcny.fios.verizon.net) |
| 22:18.27 | L|NUX | however its not empty in softphone |
| 22:21.54 | L|NUX | [TK]D-Fender: any idea |
| 22:22.17 | [TK]D-Fender | L|NUX: just go try stuff and get googleing |
| 22:22.29 | L|NUX | OKAY |
| 22:30.45 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 22:36.36 | *** join/#asterisk slima (i=slima@unaffiliated/slima) |
| 22:41.07 | *** join/#asterisk doolph (n=doolph@201.218.241.66) |
| 22:41.08 | doolph | hi |
| 22:41.14 | doolph | anyone can help me with call transfers |
| 22:43.46 | [TK]D-Fender | doolph: maybe if you actually showed us what you were doing and whats going wrong... |
| 22:44.15 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 22:44.15 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 22:48.50 | *** join/#asterisk d3wayne (n=dwayne@76.29.245.9) |
| 22:48.50 | *** mode/#asterisk [+o d3wayne] by ChanServ |
| 22:54.02 | L|NUX | [TK]D-Fender: can you look into this problem for me i can give you access to machine :$ |
| 22:54.14 | L|NUX | still trying but not working |
| 22:54.24 | [TK]D-Fender | L|NUX: No, you're just going to have to look harder |
| 22:54.33 | L|NUX | i did brother |
| 22:55.21 | [TK]D-Fender | L|NUX: MORE. |
| 22:55.27 | L|NUX | :( |
| 22:55.29 | L|NUX | i did brother |
| 22:55.37 | [TK]D-Fender | L|NUX: More means GO NOW. |
| 22:55.44 | [TK]D-Fender | L|NUX: More means not stopping |
| 22:55.47 | L|NUX | yes |
| 22:55.48 | L|NUX | i know |
| 22:55.56 | [TK]D-Fender | L|NUX: "did" is past tense. |
| 22:56.10 | L|NUX | yes |
| 22:56.10 | [TK]D-Fender | L|NUX: What does your provider say to use? |
| 22:56.28 | L|NUX | well basically they says we do not support asterisk |
| 22:56.29 | C4colo | My DIDs are present tense |
| 22:56.46 | L|NUX | C4colo: okay thanks for letting us know about your DIDs :) |
| 22:56.53 | L|NUX | C4colo: do not take me wrong |
| 22:57.07 | C4colo | I always do |
| 22:57.19 | C4colo | wait, what are we talking about again? |
| 22:57.26 | [TK]D-Fender | L|NUX: Go try seting your user agent to look like X-Pro |
| 22:57.37 | L|NUX | [TK]D-Fender : did that :) |
| 22:57.51 | L|NUX | CSeq: 102 OPTIONS |
| 22:57.51 | L|NUX | User-Agent: X-PRO build 1101 |
| 22:57.51 | L|NUX | Max-Forwards: 70 |
| 22:58.39 | L|NUX | they are using Motorola-IMS/3.1 |
| 23:06.57 | *** join/#asterisk DaPrivateer (i=Privatee@crimson.66fruit.com) |
| 23:09.43 | *** join/#asterisk DarkRift (n=dark@bas10-montreal02-1177584191.dsl.bell.ca) |
| 23:24.37 | baliktad | I am configuring my first asterisk server, trying to follow along TFOT book |
| 23:25.10 | baliktad | having some trouble getting hooked up to my voip provider |
| 23:25.37 | baliktad | here's what I've done so far: http://pastebin.ca/TI2ceSBP |
| 23:26.32 | *** join/#asterisk ManxPower (n=manxpowe@178.sub-70-220-100.myvzw.com) |
| 23:27.04 | baliktad | can anyone suggest what's wrong or how I can find out myself? |
| 23:28.09 | ManxPower | The answers you seek are within, grasshopper. |
| 23:29.41 | baliktad | I'm not even really sure what I can do to see if I'm successfully registering with my provider |
| 23:31.16 | ManxPower | "sip show registry" |
| 23:32.13 | baliktad | sip.flowroute.com:5060 07705313 105 Registered |
| 23:32.30 | [TK]D-Fender | Wow, a passworded pastebin, how.... practical |
| 23:32.49 | ManxPower | you have successfully registered with flowroute |
| 23:32.59 | baliktad | ...so the problem is somewhere on my asterisk box |
| 23:33.07 | ManxPower | [TK]D-Fender: You thought you had to be bright to do VoIP? |
| 23:33.47 | [TK]D-Fender | ManxPower: "For everyone else..... there's Vonage" |
| 23:35.18 | baliktad | I may not know a lot about asterisk yet but I am willing to learn. I also spent the last hour poring over TFOT to try and figure it out myself |
| 23:35.36 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
| 23:36.58 | [TK]D-Fender | baliktad: You're not describing to problem, and worse not showing it to use |
| 23:38.22 | baliktad | The problem is I am not able to dial out (busy signal immediately after number) or in (silence) |
| 23:38.43 | baliktad | I'm willing to provide any information that's helpful |
| 23:38.52 | ManxPower | baliktad: put a copy of the CLI output for a failed call on pastebin. I assumed you knew the drill. |
| 23:40.07 | *** join/#asterisk stencil (n=stencil@206-248-156-88.dsl.teksavvy.com) |
| 23:40.34 | baliktad | the CLI doesn't show anything right now when I try and place a call |
| 23:41.56 | [TK]D-Fender | baliktad: Call in? Call out? Go do "sip debug at * CLI and look for traffice |
| 23:42.11 | [TK]D-Fender | baliktad: And as you' mentioned SIP & NAT before, go read the guide : |
| 23:42.13 | [TK]D-Fender | ~sipnat |
| 23:42.14 | jbot | [~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 |
| 23:42.15 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^ |
| 23:42.56 | baliktad | ahh there's some SIP traffic |
| 23:45.07 | *** join/#asterisk ][ologramMan (n=giuseppe@host35-206-dynamic.2-87-r.retail.telecomitalia.it) |
| 23:45.16 | ][ologramMan | Hi all |
| 23:45.55 | ][ologramMan | I am trying to register a N78 to asterisknow but it fails, any idea? |
| 23:46.06 | baliktad | http://pastebin.ca/1083875 |
| 23:46.30 | baliktad | my voip phone is 192.168.0.200, asterisk on 192.168.0.200 |
| 23:46.41 | ManxPower | ][ologramMan: This is NOT 2nd level AsteriskNOW support. |
| 23:47.06 | ManxPower | baliktad: you can't have both be on the sam eIP. |
| 23:47.07 | baliktad | err, asterisk on 192.168.0.116 |
| 23:47.08 | ][ologramMan | Hi ManxPower, I am also using SJPhone at it works first try, I am really puzzled |
| 23:47.09 | baliktad | sorry |
| 23:47.51 | ManxPower | baliktad: A single mistake like that could open up your asterisk system to anyone in the world could make calls thru your server and have you pay for it. You should be more careful. |
| 23:48.18 | baliktad | a mistake like what? |
| 23:48.28 | ManxPower | wrong IP, a typo. |
| 23:49.16 | baliktad | understandably, I take a lot more care when I make configuration changes to a live system than an ad-hoc support chat |
| 23:51.24 | wasabi | i find it amazing that people would give a damn about making calls through somebodys system, by hacking your asterisk box, today... when they could just talk to the same people online |
| 23:52.04 | ][ologramMan | anyone else is happy with nokia ip phones and asterisk? I don't know where to search for help on this since the laptop connects without problems |
| 23:52.12 | [TK]D-Fender | baliktad: Looking for 14258828080 in phones (domain 192.168.0.116) SIP/2.0 404 Not Found |
| 23:52.33 | [TK]D-Fender | baliktad: Its telling you what context in your dialplan its trying to match that number to. It doesn't match. go look at why |
| 23:52.45 | *** join/#asterisk angryuser (n=sldf@88.140.123.21) |
| 23:55.05 | baliktad | my config is http://pastebin.ca/1083882 from what I understand, that number would match lines 16 and 17 and attempt to dial using the context (?) described on line 46 |
| 23:57.01 | [TK]D-Fender | baliktad: Look close at the CONTEXT it is look ing at. |
| 23:57.09 | [TK]D-Fender | baliktad: You are quite mistaken. |
| 23:58.37 | baliktad | I appreciate your help [TK]D-Fender, I am still learning how all the sections of sip.conf work |
| 23:59.40 | [TK]D-Fender | baliktad: Quick version. call comes in, tries to match a peer, Whatever section (if any) accept the call thats the one whose context (as specified) is used to match the number dialed against the dialplan. |
| 23:59.42 | baliktad | I suspect it has something to do with the context described in lines 46-56 (perhaps line 51 context=incoming_calls ?) |