| 00:18.13 | *** join/#asterisk ferdna (~yup@cpe-67-10-220-35.elp.res.rr.com) |
| 00:26.44 | *** join/#asterisk UnixDev (~UnixDev@unaffiliated/unixdev) |
| 00:26.59 | UnixDev | is there somewhere online that one can download the PSAP database for 911 routing? |
| 00:43.03 | *** join/#asterisk mjordan (~mjordan@user-69-1-6-49.knology.net) |
| 00:43.33 | *** join/#asterisk ferdna (~yup@cpe-67-10-220-35.elp.res.rr.com) |
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| 01:06.28 | *** join/#asterisk james_zhu (~Administr@183.16.211.132) |
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| 01:27.06 | *** join/#asterisk sequencer (~something@196.218.255.29) |
| 01:27.12 | sequencer | hi all :) |
| 01:27.56 | sequencer | i have installed Asterisk NOW with GUI, but am having a trouble with the SIP trunk i guess, it shows up as 105 and i cant make calls, any suggestions ? |
| 01:31.08 | *** join/#asterisk mhaddog (~mhaddog@z65-50-116-231.ips.direcpath.com) |
| 01:45.46 | p3nguin | sequencer: Reinstall it without a GUI and we'll help you here. |
| 01:53.34 | sequencer | p3nguin thank you , but using a GUI is much more easier in an office environment :s |
| 01:53.51 | p3nguin | Then expect no support in this channel. |
| 01:55.44 | p3nguin | I'm not saying don't use a GUI. I'm just saying this isn't the place for help with it. |
| 01:55.58 | james_zhu | show the CLI info? |
| 01:56.05 | p3nguin | huh? |
| 01:56.20 | sequencer | yes i can show any info in the CLI |
| 01:56.35 | james_zhu | based on the CLI info, we can help you |
| 01:56.49 | sequencer | i got couple of issues actually.. and would really appreciate your help guys |
| 01:56.56 | sequencer | this is the first issue i have: |
| 01:57.16 | james_zhu | solve the problem one by one? |
| 01:57.19 | sequencer | Received SIP subscribe for peer without mailbox: |
| 01:57.21 | sequencer | yes :) |
| 01:57.34 | p3nguin | Give it a mailbox. |
| 01:57.40 | james_zhu | have you set mailbox in sip.conf |
| 01:57.53 | sequencer | let me see.. |
| 01:58.00 | p3nguin | He's using a GUI; they don't use sip.conf for that. |
| 01:58.08 | p3nguin | Hence our not supporting GUIs here. |
| 01:58.35 | james_zhu | I see, should be with sip ext |
| 01:58.46 | james_zhu | check the sip exten |
| 01:58.49 | p3nguin | If you're using FreePBX, go ask in the #freepbx channel. |
| 01:59.18 | sequencer | am using the basic gui from diguim |
| 01:59.37 | james_zhu | that one is based on freepbx |
| 01:59.39 | sequencer | ive been using freepbx for almost 6 months and am having alot of issues with it |
| 01:59.45 | p3nguin | Asterisk GUI? I don't know if the asterisk-gui channel ever has anyone in it. |
| 01:59.58 | sequencer | it hass.. me and another 2-3 dudes :s |
| 02:00.31 | sequencer | i have already setup the voice mail settings |
| 02:00.48 | sequencer | but i dont see the voice mail folders in /var/spool/asterisk/voicemail |
| 02:00.59 | sequencer | anyways.. |
| 02:01.10 | sequencer | this is not that important.. i have a greate problem :s |
| 02:02.05 | sequencer | 10.250.0.5:5060 N 445433765 105 Registered Fri, 19 Aug 2011 14:57:07 |
| 02:02.17 | sequencer | this is my sip show registry |
| 02:02.23 | p3nguin | Okay, so you registered to an ITSP? |
| 02:02.42 | sequencer | ITSP ? |
| 02:02.46 | p3nguin | ~itsp |
| 02:02.46 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 02:02.49 | sequencer | yes |
| 02:02.55 | sequencer | using SIP technology |
| 02:03.13 | p3nguin | Okay, registering to the ITSP is good. Now they can send you phone calls. |
| 02:03.22 | sequencer | the only problem is |
| 02:03.31 | sequencer | i am unable to send or recieve calls |
| 02:03.44 | sequencer | there seems to be a problem with the SIP trunk |
| 02:03.46 | p3nguin | Did you create your peer entry? |
| 02:03.56 | p3nguin | There's no such thing as a SIP trunk. |
| 02:03.56 | sequencer | yes, using the GUI |
| 02:04.07 | p3nguin | And we don't support GUIs here. |
| 02:04.16 | sequencer | i understand that |
| 02:04.46 | p3nguin | So create your peer, then create any necessary extensions for calls from the ITSP. |
| 02:05.21 | sequencer | i believe it would be ok if i added a peer in sip.conf ? |
| 02:05.37 | p3nguin | You should have a peer which matches calls from your ITSP. |
| 02:05.51 | p3nguin | That allows you to send those particular calls into a specific context. |
| 02:05.58 | sequencer | right |
| 02:09.23 | p3nguin | So... |
| 02:09.29 | p3nguin | The problem is what? |
| 02:09.41 | p3nguin | Did you check to see if calls reach your system? |
| 02:10.20 | p3nguin | If you configured the peer and you registered, and if you configured the ITSP's side to route calls to your account, calls should be there. |
| 02:10.27 | sequencer | the problem is, i cant make or recieve calls |
| 02:10.36 | p3nguin | If you don't know if you configured a peer, "sip show peers" will show you. |
| 02:10.36 | sequencer | the ITSP is already configured an in place |
| 02:11.12 | p3nguin | Just because you can't receive calls does not mean calls are not making it to you. You could be lacking the appropriate extension for the calls. |
| 02:11.19 | sequencer | trunk_1/7736275793 10.250.0.5 5060 Unmonitored |
| 02:11.50 | p3nguin | Run "sip set debug on" and then call your phone number which is meant to arrive on Asterisk. |
| 02:12.08 | p3nguin | I hope you have your mobile phone handy. |
| 02:12.20 | sequencer | when i try to make an outbound call i get a message saying "if you would like to make a call please hangup and try to make your call again " |
| 02:12.33 | sequencer | and on CLI i see its trying to make a call to a fail-over trunk |
| 02:12.52 | p3nguin | I'm not sure what that means yet. |
| 02:12.53 | sequencer | well.. p3nguin there is something i didnt mention :"> |
| 02:13.23 | sequencer | i have already 40 sip phones ( extensions ) registered to the asterisk |
| 02:13.37 | sequencer | its almost impossible to track with sip set debug on |
| 02:14.06 | sequencer | ill try to capture the output |
| 02:14.24 | p3nguin | Okay, just so we're clear, you have 40 phones, which are not extensions. |
| 02:14.34 | p3nguin | Phones are just phones. |
| 02:14.48 | p3nguin | Extensions are something else. |
| 02:15.01 | p3nguin | You can debug based on IP or peer name. |
| 02:15.10 | p3nguin | sip set debug peer trunk_1 |
| 02:15.33 | sequencer | didnt know this one! |
| 02:15.48 | p3nguin | sip set debug <tab> |
| 02:15.54 | p3nguin | That will show you possible ways to complete the command. |
| 02:16.12 | p3nguin | All the CLI commands are that way. |
| 02:16.25 | p3nguin | si<tab> gives you sip, for example. |
| 02:17.14 | sequencer | great |
| 02:17.19 | sequencer | now we're getting somewhere |
| 02:20.05 | sequencer | i got the log |
| 02:20.38 | sequencer | http://pastebin.com/Y9sprKMu |
| 02:21.14 | sequencer | call starts at line 100 |
| 02:23.52 | sequencer | after* |
| 02:25.44 | p3nguin | *shrug* |
| 02:25.50 | sequencer | ? |
| 02:25.58 | p3nguin | This is one of the reasons we don't do GUIs here. |
| 02:26.12 | sequencer | am sorry ? |
| 02:26.20 | p3nguin | There is no "trunk failover" bullshit in plain asterisk. At least not until you create one. |
| 02:26.34 | p3nguin | And if you created one, I could support it. |
| 02:26.38 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 02:26.51 | p3nguin | And all the macro stuff... what's with that? |
| 02:27.06 | p3nguin | I want to help you. I really do. |
| 02:27.15 | sequencer | and i really appreciate it |
| 02:27.25 | sequencer | i know its not simple at all :s |
| 02:27.47 | sequencer | i dont have a failover trunk anyways |
| 02:28.05 | sequencer | it just fails on my main and tries to look for a different one. |
| 02:28.11 | sequencer | then it hangs up :s |
| 02:28.21 | p3nguin | Here's how it's done in asterisk: create your peer to match calls from a particular host, send those calls into a context, create extensions in that context to do something useful with the calls. |
| 02:28.39 | sequencer | i understand that, in plain asterisk |
| 02:29.26 | p3nguin | There is no failover anything until someone creates it. And quite frankly, I don't want to look through a bunch of canned configuration to figure out why it's doing what it's doing. |
| 02:29.35 | sequencer | but it would be a greater hassle once i need to use enterprise functions, such as time conditions, call conferencing, paging, intercom, call transferforward |
| 02:29.50 | sequencer | you are correct |
| 02:29.55 | sequencer | absoloutly right |
| 02:30.09 | p3nguin | Why would it be a hassle? The rest of us here are capable of doing all that stuff without a GUI. |
| 02:30.20 | sequencer | am not a Guru |
| 02:30.27 | sequencer | at least in asterisk |
| 02:30.32 | p3nguin | Most of the people coming here aren't either. |
| 02:30.39 | p3nguin | They still do it. |
| 02:31.02 | sequencer | i think it might be a codec issue.. let me check.. |
| 02:31.31 | p3nguin | Let's ignore for a second the fact that you're using a GUI. Drop down to the basic level of things that we've already covered. |
| 02:31.52 | p3nguin | If your peer for the ITSP doesn't work, check it, fix it, then make a call with it. |
| 02:32.04 | sequencer | thats what am trying to do |
| 02:32.13 | p3nguin | If calls are failing over to another peer, fix the broken one. |
| 02:32.34 | sequencer | i know that the ITSP works, because it was working couple of hours ago before i transferred to a different version of asterisk |
| 02:32.36 | p3nguin | Pretend that it's configured in sip.conf. |
| 02:32.39 | sequencer | i used to use freepbx |
| 02:33.31 | sequencer | hangup cause code 16 |
| 02:34.17 | sequencer | normal clearing :s |
| 02:35.35 | p3nguin | No clue. |
| 02:35.51 | p3nguin | If it were mine, I'd change my peer settings until it worked. |
| 02:36.28 | sequencer | exactly what am trying to do |
| 02:36.43 | sequencer | even though i dont know where the settings are |
| 02:36.53 | sequencer | so am playing around with the settings i can see |
| 02:38.07 | p3nguin | Can you pastebin your peer definition? |
| 02:38.30 | sequencer | this is the problem , i dont seem to find it |
| 02:39.17 | p3nguin | In asterisk, sip.conf is where they belong. |
| 02:39.24 | p3nguin | That's all I can tell you. |
| 02:39.32 | sequencer | i already dug into there |
| 02:39.35 | sequencer | nothing.. |
| 02:39.45 | p3nguin | Then they are in an included file, probably. |
| 02:39.56 | sequencer | when i change the codecs i started getting cngestion |
| 02:40.02 | sequencer | so i am getting somewhere |
| 02:40.36 | p3nguin | I think the pastebin I saw said the peer only supports ulaw. |
| 02:41.24 | sequencer | and it should also support g729 |
| 02:41.48 | sequencer | but maybe i need to install the licenses for that |
| 02:42.02 | p3nguin | Yep. |
| 02:42.32 | p3nguin | But what I saw said ulaw only. |
| 02:42.46 | p3nguin | The paste is expired, so I can't look again. |
| 02:42.48 | sequencer | i set it to ulaw |
| 02:43.04 | p3nguin | I thought it said you were trying to use all codecs but the peer only supported ulaw. |
| 02:43.09 | sequencer | let me try to llok fr the peer settings |
| 02:43.13 | sequencer | i did |
| 02:43.23 | sequencer | it declined all the other codecs with congestion |
| 02:43.53 | p3nguin | disallow=all, then only allow the codec you want it to use. |
| 02:44.05 | sequencer | right |
| 02:44.07 | p3nguin | If what I think I saw is right, allow=ulaw only. |
| 02:46.06 | sequencer | i did that |
| 02:46.17 | sequencer | now am still getting the same old problem.. |
| 02:46.20 | sequencer | its fine.. |
| 02:46.23 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 02:46.36 | sequencer | ill try to find the peer settings |
| 02:46.53 | p3nguin | Good luck with your GUI! |
| 02:47.07 | p3nguin | When you can't make it work, ditch and and we'll help you when problems arise. |
| 02:47.37 | p3nguin | s/and/it/ |
| 02:48.12 | sequencer | i appreciate it guys :) |
| 02:48.33 | sequencer | i got somewhere now |
| 02:49.34 | sequencer | i suppose that all the config files willbe in one folder, not sccattered around, right ? |
| 02:49.47 | p3nguin | They might be in one directory. |
| 02:49.54 | p3nguin | But folders are for Windows. |
| 02:50.14 | sequencer | yeah |
| 02:51.36 | sequencer | am copying all the config files to my pc |
| 02:51.43 | sequencer | and ill search all for the peer name |
| 02:51.54 | p3nguin | Why would you do that? |
| 02:51.58 | p3nguin | grep -r works just fine. |
| 02:52.28 | sequencer | so i would keep a backup locally |
| 02:55.10 | sequencer | bingo |
| 02:55.14 | p3nguin | Maybe I misunderstood what you meant. |
| 02:55.14 | sequencer | not found anywhere :s |
| 02:55.39 | sequencer | lets try the grep -r :s |
| 02:55.54 | p3nguin | You know how to use it? |
| 02:56.04 | sequencer | got it |
| 02:56.07 | sequencer | users.conf |
| 02:56.13 | p3nguin | ~users.conf |
| 02:56.13 | infobot | [~users.conf] users.conf is a flaming pile of sh1t that takes the fine control of several perfectly usable asterisk config files and reduces them to the lowest common denominator and makes your system behave like a "toaster grade" PBX system. |
| 02:56.27 | sequencer | lol |
| 02:56.46 | sequencer | cant agree more |
| 02:56.56 | WIMPy | Didn't know that one |
| 02:57.07 | sequencer | why in hell.. |
| 02:57.13 | sequencer | insecure=port,invite |
| 02:57.23 | sequencer | while i put it as very :s |
| 02:57.23 | p3nguin | That's typical. |
| 02:57.33 | p3nguin | very isn't valid anymore. |
| 02:57.46 | sequencer | but i did set it that way :s |
| 02:58.00 | p3nguin | But port,invite means the same thing that very meant before. |
| 02:59.27 | sequencer | is there a way to reload the file after i modified it ? |
| 02:59.39 | sequencer | i did dialplan reload |
| 02:59.39 | p3nguin | core restart when convenient |
| 02:59.47 | sequencer | alight |
| 02:59.55 | p3nguin | dialplan reload only reloads the dialplan, which is extensions.conf and any included files. |
| 03:00.15 | sequencer | ok |
| 03:02.59 | sequencer | how can i register a sip ? |
| 03:03.08 | p3nguin | What does that mean to you? |
| 03:03.13 | sequencer | umm.. |
| 03:03.20 | sequencer | a voice trunk |
| 03:03.27 | sequencer | register -> |
| 03:03.39 | p3nguin | sip register statements go in sip.conf. |
| 03:07.11 | sequencer | http://pastebin.com/WGrLwC9W |
| 03:07.16 | sequencer | would this work anyway ? |
| 03:08.11 | p3nguin | It might work. 'very' won't be good for the insecure parameter, though. |
| 03:08.25 | sequencer | any other issues you see ? |
| 03:08.51 | p3nguin | I wouldn't have used all those unnecessary parameters if I were configuring it. |
| 03:09.18 | p3nguin | The main concern of mine is with the host address. |
| 03:09.38 | p3nguin | That's a private address, but I have to assume that you aren't local to your ITSP. |
| 03:09.51 | sequencer | yes i am |
| 03:09.56 | sequencer | am using a private port |
| 03:09.59 | p3nguin | You have a hosted PBX? |
| 03:10.23 | sequencer | no, but my provider aupplies the SIP over a private network |
| 03:10.29 | p3nguin | Alright. If you're okay with that address, then I guess I am, too. |
| 03:10.36 | p3nguin | It's not common to see that here. |
| 03:10.37 | sequencer | yeah, its right |
| 03:10.42 | sequencer | i know that |
| 03:10.56 | sequencer | it took me a whole day to figure out how to connect to that one |
| 03:10.59 | sequencer | very uncommmon |
| 03:11.21 | sequencer | setting up the correct gateway for eath ethernet card on the pbx |
| 03:11.24 | p3nguin | Nothing stands out to me other than the insecure=very being wrong. |
| 03:11.33 | sequencer | ill change it |
| 03:13.52 | sequencer | am looking at DID_trunk_1 |
| 03:14.03 | sequencer | that one is for incoming routes :s |
| 03:14.19 | p3nguin | peers work in both directions. |
| 03:14.47 | sequencer | therefore if i cant call out i wont get calls |
| 03:15.03 | p3nguin | That's not necessarily true. |
| 03:15.20 | sequencer | viola ! |
| 03:15.28 | sequencer | i recieved a call :S |
| 03:15.30 | p3nguin | Like a violin? |
| 03:15.58 | sequencer | like a voila ? |
| 03:16.29 | sequencer | ok so i recieved a call |
| 03:16.32 | sequencer | wheew |
| 03:16.33 | p3nguin | http://www.baroquecello.com/images/viola.jpg |
| 03:16.46 | sequencer | lol |
| 03:16.53 | sequencer | i know what a violin/viola is |
| 03:17.01 | p3nguin | oh |
| 03:17.30 | sequencer | http://wiki.answers.com/Q/What_does_voila_mean |
| 03:17.31 | sequencer | ;) |
| 03:17.39 | p3nguin | I'm always amused when people do something wonderful and shout out about this violin-like instrument. |
| 03:17.49 | sequencer | lol |
| 03:18.01 | sequencer | its not an instrument |
| 03:18.06 | sequencer | check the link ;) |
| 03:18.12 | sequencer | vwah-lah |
| 03:18.14 | sequencer | :p |
| 03:18.21 | p3nguin | A viola isn't an instrument? I'm pretty sure it is. It's also a plant. |
| 03:18.28 | sequencer | viola is |
| 03:18.32 | p3nguin | (2215.20) <sequencer> viola ! |
| 03:18.33 | sequencer | voila isnt ;) |
| 03:18.40 | sequencer | typo! |
| 03:18.58 | sequencer | i know a girl named viola |
| 03:18.59 | p3nguin | Anyway... |
| 03:19.01 | p3nguin | You fixed the peer. |
| 03:19.02 | sequencer | yeah.. |
| 03:19.04 | sequencer | nope |
| 03:19.06 | sequencer | i didnt |
| 03:19.08 | p3nguin | no?! |
| 03:19.10 | sequencer | am recieving calls |
| 03:19.14 | sequencer | but not making any |
| 03:19.21 | p3nguin | That's good. Now fix your dialplan. |
| 03:19.25 | sequencer | so it has to be with the dialplan |
| 03:19.26 | sequencer | right |
| 03:19.29 | p3nguin | Or add a peer for the phone. |
| 03:19.59 | p3nguin | If you have a peer for the phone, check its context. |
| 03:20.57 | sequencer | its all in that users |
| 03:21.12 | p3nguin | I guess that's what asterisk-gui does. |
| 03:22.43 | sequencer | oh wow |
| 03:22.50 | sequencer | you wont believe this |
| 03:23.00 | p3nguin | Try me. |
| 03:23.03 | sequencer | [CallingRule_outs] |
| 03:23.03 | sequencer | exten = _.,1,Macro(trunkdial-failover-0.3,${trunk_1}/${,${EXTEN:0})},,trunk_1,) |
| 03:23.16 | sequencer | this is my dialplan |
| 03:23.20 | sequencer | how pathetic ? |
| 03:23.42 | p3nguin | I'm sure someone spent a lot of time thinking how to make that work, too. |
| 03:24.26 | sequencer | i think this should be;.. |
| 03:24.47 | sequencer | Dial(trunk_1/${EXTEN:0}); |
| 03:24.51 | sequencer | ? |
| 03:25.05 | sequencer | if am not mistaken |
| 03:25.11 | p3nguin | Dial(SIP/trunk_1/${EXTEN}); |
| 03:25.16 | sequencer | oh right |
| 03:26.16 | sequencer | not making it :s |
| 03:27.13 | sequencer | <PROTECTED> |
| 03:27.32 | p3nguin | I guess you know which context your phone has assigned. |
| 03:27.41 | sequencer | oh my bad |
| 03:28.51 | sequencer | lets dance |
| 03:29.16 | sequencer | :D |
| 03:29.46 | sequencer | some minor things to do later on today |
| 03:29.50 | sequencer | but this is perfect |
| 03:29.54 | sequencer | thanks man! |
| 03:30.25 | p3nguin | I suppose you'll figure out how the GUI manages that stuff later. |
| 03:31.13 | sequencer | i dont think so |
| 03:31.20 | sequencer | ill have to go by the config files |
| 03:31.36 | sequencer | i cant use the gui now or it will overwrite all the changes |
| 03:31.51 | sequencer | want me to try you with something ? ;) |
| 03:32.01 | p3nguin | I know FreePBX overwrites all the changes you make manually the first time you press that big apply button, but I don't know what Asterisk GUI does. |
| 03:32.38 | sequencer | the same thing.. |
| 03:32.46 | p3nguin | YOu tested it? |
| 03:32.49 | sequencer | they all write new config files then reloads |
| 03:33.06 | sequencer | am sorry ? |
| 03:33.17 | p3nguin | I don't think Asterisk GUI is quite so destructive. |
| 03:36.15 | sequencer | after what ive seen ? |
| 03:36.21 | sequencer | its meant to be |
| 03:36.23 | sequencer | i dunno why |
| 03:37.23 | sequencer | ive been trying to fix this since 12 AM |
| 03:37.34 | p3nguin | What time is it now? |
| 03:37.38 | sequencer | 6:40 AM |
| 03:38.04 | sequencer | not to mention the configuration itself took me 9 hours yesterday |
| 03:38.56 | sequencer | thanks buddy! |
| 03:39.51 | sequencer | i gotta run before i sleep in the office |
| 03:39.52 | sequencer | lol |
| 03:40.10 | p3nguin | :) |
| 03:40.15 | sequencer | :) |
| 03:40.16 | sequencer | cya |
| 03:40.16 | p3nguin | nighty night! |
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| 06:34.00 | prgmrchris | polycom support site down for anyone else? |
| 06:34.11 | prgmrchris | http://support.polycom.com/PolycomService/home/index.htm |
| 06:38.46 | ChannelZ | is dead |
| 06:38.48 | mtbf | Yup. |
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| 10:07.57 | salz212 | Hello All |
| 10:13.13 | salz212 | Can any one provide me HOW TO OpenSer+Asterisk Integration? |
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| 10:49.17 | KNERD | what are the "torcher menus"? |
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| 14:42.47 | jstapleton | anybody else have google voice trunk problems starting circa yesterday? outbound calls just ring forever. inbound is spotty as well? |
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| 15:02.24 | pabelanger | jstapleton: google broke something, I see an issue reported in JIRA |
| 15:05.02 | jstapleton | pabelanger: i figured as much. ;-) do you have a link to the JIRA issue? thanks again! |
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| 15:19.56 | seraphie | pabelanger: pm please |
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| 16:39.51 | ruben23 | hi guys, any suggestions on this log error on my asterisk server..? -----------------> http://i54.tinypic.com/2rhxlqh.jpg |
| 16:39.55 | ChannelZ | https://issues.asterisk.org/jira/browse/ASTERISK-18301 |
| 16:40.16 | ChannelZ | ruben23: bogus wav files? |
| 16:41.20 | ruben23 | <PROTECTED> |
| 16:41.53 | ChannelZ | Make good wav files? |
| 16:42.13 | ChannelZ | Is this custom sounds, or music on hold, or what |
| 16:44.36 | pabelanger | jstapleton: https://issues.asterisk.org/jira/browse/ASTERISK-18301 |
| 16:49.43 | stallione | Is it possible to customize the SIP client string that Asterisk uses to register with a SIP provider? (for example, instead of showing up as 'Asterisk PBX' can I change it to 'My Home SoftPhone' or something like that ) ? |
| 16:53.18 | Gugge | Yes |
| 16:53.24 | Gugge | in sip.conf |
| 16:54.16 | stallione | Gugge: Thx. checking it out now! |
| 16:56.26 | *** join/#asterisk sequencer (~something@196.218.255.29) |
| 16:56.31 | sequencer | Hi all |
| 16:56.35 | sequencer | p3nguin :) |
| 16:56.59 | stallione | asterisk.adsi is where I found it ? would that be correct? |
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| 17:13.26 | *** join/#asterisk dubey (~sandeep@14.97.201.111) |
| 17:13.32 | dubey | hello |
| 17:15.43 | sequencer | how can i disable notices in CLI ? |
| 17:16.42 | dubey | I have five office in different states, every month i pay thousands of doller for communication via telephone. someone has suggested me asterisk. |
| 17:17.42 | dubey | Is it possible that i can setup asterisk on EC2 instance and communicate with my remote offices in real time including video/chat etc. ? |
| 17:48.51 | p3nguin | sequencer: I suppose you could adjust it in logger.conf. |
| 17:52.31 | p3nguin | stallione: No, sip.conf is where you adjust sip settings. |
| 17:53.10 | p3nguin | stallione: Look at the useragent parameter. |
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| 18:09.19 | p3nguin | Let me guess... |
| 18:09.26 | p3nguin | another a2billing question. |
| 18:09.31 | x1user | :) |
| 18:09.49 | p3nguin | Maybe one of these days you'll figure out that no one here uses a2billing. |
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| 18:28.43 | sequencer | i dont use it |
| 18:28.47 | sequencer | p3nguin |
| 18:28.54 | stallione | Thanks P3nguin. Would it be possible to completely fool the SIP provider and make my Asterisk behave like a Grandstream Device? (are there any other parameters that need to be sent?) |
| 18:29.01 | p3nguin | sequencer: You don't use what? |
| 18:29.09 | sequencer | a2billing ;) |
| 18:29.28 | sequencer | anyways, after i came in today, i found that all of my lines are down |
| 18:29.44 | sequencer | my sip isnt registering at all :s |
| 18:29.58 | sequencer | sip show registry shows 0 registrations |
| 18:30.56 | sequencer | no ! not by peer :s |
| 18:31.47 | stallione | :) |
| 18:31.57 | p3nguin | stallione: I think it's just the useragent. To use asterisk with magicjack, only a special useragent string has to be used, so I would think that's enough for the grandstream as well. |
| 18:32.19 | stallione | excellent! Thanks. |
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| 18:33.35 | penguin | <PROTECTED> |
| 18:33.44 | p3nguin | You can set it in the peer definition used for the peer requiring you to use grandstream products. |
| 18:34.47 | prgmrchris | p3nguin: what happens when you connect it to a pbx and suddenly start doing 20,000 minutes/month do they shut you off |
| 18:35.04 | prgmrchris | id imagine their definition of unlimited isnt really unlimited |
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| 18:35.23 | p3nguin | I think unlimited means like 3500 minutes per month. |
| 18:35.27 | prgmrchris | ah |
| 18:36.07 | prgmrchris | i hate how everybody uses the word unlimited now and doesnt mean unlimited |
| 18:36.15 | p3nguin | yeah |
| 18:36.17 | prgmrchris | seems like every greedy company is getting away with that marketing |
| 18:36.25 | p3nguin | They should say limited if that's what they mean. |
| 18:36.33 | prgmrchris | verizon has unlimited 5gb data |
| 18:36.40 | prgmrchris | yay |
| 18:36.41 | penguin | lol |
| 18:38.15 | prgmrchris | I've had "unlimited" gprs, edge, 3g, and never really seen any kind of actual cap. Sure they were slower but I didn't have to worry about getting surprised on my next bill. If my edge from 5+ years ago could 3gb/day and 90gb a month how is 4G at 5gb an improvement of the service? |
| 18:38.57 | p3nguin | It's newer and it is a higher number, so it's better! |
| 18:39.01 | prgmrchris | yea |
| 18:39.28 | p3nguin | That's the common mentality in the technology world. |
| 18:39.35 | prgmrchris | i also like it at the datacenter level too "ZOMG 100mbps 29.99/mo" then you find out its 100gb/month |
| 18:39.48 | prgmrchris | or they shape you down once you start doing a certain amount |
| 18:39.59 | prgmrchris | lawlz |
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| 18:44.46 | sequencer | p3nguin |
| 18:44.51 | sequencer | i have a question here |
| 18:45.08 | sequencer | if it fails to register, shouldn't i get a retransmit ? |
| 18:45.27 | sequencer | i dont see my peer trying to connect even with my sip set debug peer .. |
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| 19:15.39 | voiper | Hi, I am trying to send a call from asterisk 1.6.2.20 to a voipswitch server using SIP. I would like to send the username and password as part of the SIP request. I have tried different methods with no luck like Dial(SIP/username:password@host/${exten} with no luck. If anyone can help me getting the right sytax, that will be great |
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| 19:38.15 | [sr] | i'm tired |
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| 20:47.55 | linuxgecko | I'm working with the google-voice setup on the wiki, and suddenly, i can't make calls. looking into the server, i see that my sip connection is registered, and it works fine. 'jabber show connections' shows I'm connected, but 'gtalk show channels' shows no active channels. what can make this not work, with no config changes? |
| 20:48.55 | linuxgecko | I'm using 1.8.4.2 |
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| 20:58.41 | ChannelZ | https://issues.asterisk.org/jira/browse/ASTERISK-18301 |
| 21:14.40 | ChannelZ | dang I am getting a shit-ton of these "Got presense packet from xxx@gmail.com, someone not in our roster!!!!!" messages from Jabber |
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| 22:56.39 | pdtpatrick | Question - has anyone used SIP Trunking for their home box .. and are there any they would refer that is reasonable in terms of pricing? |
| 22:58.28 | carrar | ~itsp |
| 22:58.28 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
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| 23:10.19 | pdtpatrick | ~itsplist-us |
| 23:10.20 | infobot | Here are some popular ITSPs (USA) starting with the more respected ones: http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com , http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net , http://voip.ms and http://flowroute.com |
| 23:12.39 | pdtpatrick | carrar, thanks :) |
| 23:15.26 | p3nguin | But SIP doesn't do trunking; there's only SIP. |
| 23:15.52 | p3nguin | IAX2 does trunking. |
| 23:17.31 | pdtpatrick | well i'm still on the learning curve -- Thanks for that heads up. Now i know that :) |
| 23:17.47 | p3nguin | It's good that you're told now, then. |
| 23:17.50 | p3nguin | Some people NEVER get it. |
| 23:18.27 | pdtpatrick | yeah i'm never opposed to learning -- that's why i ask or just stay in the room to listen |
| 23:21.00 | p3nguin | I hope you've already learned the difference between extensions and phones, as well. |
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