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| 00:05.56 | *** mode/#asterisk [+o file] by ChanServ |
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| 00:23.26 | thazza | hey all. I have a problem with asterisk 1.6.1.12 and realtime with IAX setup.. My iax requires the requirecalltoken=no option. however I can't work out how to add to the mysql database |
| 00:29.17 | Katty | :> |
| 00:29.21 | Katty | i have chili |
| 00:30.04 | Katty | homemade chili at that |
| 00:30.52 | ChannelZ | and it burns burns burns |
| 00:30.53 | thazza | No takers on RealTime setup with IAX? |
| 00:30.56 | ChannelZ | the ring of fire |
| 00:31.22 | Katty | ChannelZ: the chili is HOT |
| 00:32.00 | Katty | i put 4 oz ogreen chilis into it |
| 00:32.16 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 00:33.01 | jaytee | did you use cumin? chipotle pepper? |
| 00:33.27 | Katty | cumin, chili powder, garlic salt, onion powder, seasoned salt, black pepper |
| 00:33.57 | Katty | ground turkey, 2 cans of chili ready tomatos, and a can of tomato sauce |
| 00:34.03 | Katty | and chopped onion |
| 00:34.32 | Katty | oh, and minced garlic....and cheese on the top at the end |
| 00:35.10 | jaytee | garlic salt? how bourgeois |
| 00:35.24 | Katty | bourgeois does not parse. |
| 00:35.49 | jaytee | ah, the minced garlic on top is good but I'd have substituted the garlic salt with the minced IN the chili |
| 00:35.59 | Katty | nono |
| 00:36.06 | Katty | you saute the meat with garlic and onion |
| 00:36.11 | jaytee | ok |
| 00:36.23 | Katty | then add the cans of tomatos and tomato sauce |
| 00:36.28 | Katty | add in seasonings. |
| 00:36.33 | Katty | simmer for 30 minutes |
| 00:36.36 | jaytee | sounds like quicky mix |
| 00:36.58 | Katty | it doesn't take any time at all to throw together |
| 00:37.23 | Katty | 1 C of chili is 145 calories |
| 00:38.15 | Katty | the original recipe calls for kidney or chili beans, but ryan doesn't like the beans...it also calls for 1/2 C brown sugar, but ryan doesn't like his chili sweet |
| 00:38.41 | jaytee | mine takes a bit longer. I buy the ground beef ground coarse for chili and lightly brown it then add it to the chili stock. the stock is chili powder, tomato paste, crushed tomatoes, pinto beans, cumin, dash of cayenne and a tsp of ground chipotle pepper. |
| 00:39.53 | Katty | i don't think i've ever seen chipotle pepper anywhere |
| 00:40.14 | jaytee | McCormick spices sells it |
| 00:40.32 | jaytee | and ground Ancho too |
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| 00:41.06 | jaytee | right now I've got a chicken marinading in a mango/chipotle marinade |
| 00:41.45 | jaytee | it's a liquid marinade but there's a company that makes a mango/chipotle spice rub mix that's pretty good too |
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| 00:46.52 | twanny796 | is there a web based software that manages asterisk installed on another computer? |
| 00:47.10 | p3nguin_ | ssh |
| 00:47.47 | twanny796 | p3nguin_: :) |
| 00:47.55 | p3nguin_ | That's how normal people do it. |
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| 00:57.00 | BadHorsie | What's the meaning of "FMPR" and "FMGL" on a follow me? |
| 00:57.11 | hipitihop | I sometimes get "pbx.c:4390 __ast_pbx_run: Timeout, but no rule 't' in context 'phones' .. can someone point me at what this means |
| 00:59.42 | Chainsaw | hipitihop: A timeout occured, and your dial plan has no extension for it to jump to. It can't handle the error as a result. |
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| 01:07.11 | hipitihop | Chainsaw, sorry what timed out ? |
| 01:07.42 | Chainsaw | hipitihop: Likely a call in the phones context. Hard to say really, you're not giving me much to work with. |
| 01:08.30 | p3nguin_ | Here's what I put it a lot of contexts: |
| 01:08.32 | p3nguin_ | exten => t,1,Playback(vm-goodbye) |
| 01:08.33 | p3nguin_ | exten => t,n,Hangup() |
| 01:08.58 | p3nguin_ | Now if there is a timeout, the system says goodbye and hangs up. |
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| 01:09.27 | hipitihop | p3nguin_, is that aimed at me ? |
| 01:09.42 | p3nguin_ | Pretty much. |
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| 01:10.08 | hipitihop | ok thanks, will check my phones context add |
| 01:10.10 | p3nguin_ | It wasn't a solution to your problem, since you haven't given any helpful details, but it is related to your timeout not having anything to do. |
| 01:16.22 | hipitihop | Chainsaw, p3nguin_sorry about the sparse info... I'm ver new to * so I'm not sure what is relavant .. just wokring my way through the online book. |
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| 01:26.43 | p3nguin_ | hipitihop: Typically, verbose output leading up to the questionable lines would be useful to see why such lines have been produced. For SIP calling, sip debug output it usually requested. |
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| 01:31.06 | hipitihop | p3nguin_, understand... actually watching the verbose output actually tells me now that is is actually related to me trying to make an outgoing call |
| 01:31.28 | capitan | hello everyone! |
| 01:31.31 | p3nguin_ | Nevertheless, a timeout occurred on that call. |
| 01:31.39 | capitan | so... simple question... the asterisk documentation on voip-info.org says macros can only have the s extension... |
| 01:31.48 | p3nguin_ | are you sure? |
| 01:32.08 | capitan | is this still true? or do newer asterisk versions support other extensions on macros? |
| 01:32.27 | p3nguin_ | I think I've seen some macros with other extensions. |
| 01:33.11 | capitan | thanks p3nguin_ |
| 01:33.19 | capitan | maybe i can explain my problem... |
| 01:33.26 | capitan | i'm trying to use freepbx with my voip phone |
| 01:33.50 | capitan | but the macro-user-callerid executes at h,1 as though it' |
| 01:33.57 | capitan | s,22 |
| 01:34.13 | capitan | and hangs up |
| 01:34.25 | capitan | i'm curious whether i'm using a freepbx version that's too new for my asterisk version |
| 01:34.55 | p3nguin_ | No clue. We don't do FreePBX here. |
| 01:35.00 | p3nguin_ | ~freepbx |
| 01:35.01 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 01:35.14 | capitan | right :S sorry... |
| 01:35.30 | capitan | i figured i would ask generic asterisk questions here to help me with my (clearly freepbx) issue... |
| 01:35.31 | p3nguin_ | As far as Asterisk is concerned, I think the macro is usually s, but can be something else if you explicity choose to use something else. |
| 01:36.19 | p3nguin_ | It's a FreePBX issue, because Asterisk doesn't have the [macro-user-callerid] context until FreePBX adds it. |
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| 01:37.05 | p3nguin_ | I believe I have exactly one macro context in my extensions.conf, and it does use the s extension. |
| 01:37.23 | capitan | hehe |
| 01:38.19 | capitan | right... my question is too generic of a dialplan question to ask there, and too specific of a freepbx question to ask here :( |
| 01:38.51 | capitan | i guess i could write my own macro outside of freepbx that exhibits this behavior and turn it into an asterisk specific question :P |
| 01:39.34 | p3nguin_ | Macros are being deprecated in favor of Goto, if I understood the conversations correctly. |
| 01:40.08 | capitan | really? |
| 01:40.37 | capitan | but then you have to manage your own "call stack" so to speak, no? |
| 01:41.25 | p3nguin_ | I'm not sure what you mean. |
| 01:41.50 | capitan | i mean... how do you get back to where you were? |
| 01:42.10 | p3nguin_ | Oh, maybe it was a GoSub and Return that it was being deprecated in favor of. |
| 01:42.16 | capitan | ah |
| 01:42.31 | p3nguin_ | I believe I misspoke, and I'm glad you asked what you did to make me realize it. |
| 01:42.33 | capitan | makes sense... never understood the respective pros/cons of the two |
| 01:42.56 | p3nguin_ | <@leifmadsen> GoSub() is preferred over Macro() |
| 01:43.14 | capitan | that name looks very familiar :P |
| 01:43.33 | p3nguin_ | <bmoraca_work> Orbixx, in 1.4, you're looking for Macros...in 1.6, check out "gosub" |
| 01:43.33 | capitan | i'll take that as an authoritative answer ;) |
| 01:43.44 | p3nguin_ | <ManxPower-work> dlynes_laptop: use a gosub instead of a macro if using 1.6 |
| 01:43.55 | capitan | hehe... p3nguin_... what are you an archive? :P |
| 01:44.09 | p3nguin_ | Yep. |
| 01:44.59 | capitan | hmmmm... still all quiet in #freepbx :( |
| 01:45.04 | p3nguin_ | I use 1.4, and I still use GoSub()/Return(). |
| 01:45.14 | capitan | is there a #freepbx on other irc nets? |
| 01:45.21 | p3nguin_ | I doubt it. |
| 01:45.33 | capitan | ya looks like everyone's on freenode these days |
| 01:45.40 | capitan | it's the "in" place to be ;) |
| 01:46.08 | p3nguin_ | It's certainly _a_ place to be... don't know much more than that. |
| 01:50.52 | *** join/#asterisk fakhir_ (~fakhir@unaffiliated/fakhir) |
| 01:53.49 | capitan | darn... close... but not close enough :( https://issues.asterisk.org/view.php?id=14122 |
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| 02:19.38 | capitan | p3nguin_, thanks for all your help... but do you know what happens if you define a macro twice? |
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| 03:08.14 | hipitihop | in my sip.conf is it possible to have one entry for my sip provider account for both inbound and outbound calls ? |
| 03:08.35 | p3nguin_ | Yes, and that's ordinarily how you would configure it. |
| 03:08.50 | p3nguin_ | Rarely do you need to break it apart into in and out. |
| 03:09.30 | hipitihop | p3nguin_, so if it works for incoming and it is type=friend then it should work for outgoing |
| 03:09.59 | p3nguin_ | Probably, but change it to type=peer. |
| 03:10.30 | hipitihop | doesn't that mean I can only use it for incoming ? |
| 03:10.34 | p3nguin_ | nope |
| 03:10.49 | p3nguin_ | Only type=user is restricted to one-way calls. |
| 03:14.33 | hipitihop | ok, and so what is the typical outgoing entry in extension.conf |
| 03:15.32 | p3nguin_ | exten => NXXNXXXXXX,1,Dial(SIP/${EXTEN}@youritsppeer) |
| 03:15.46 | p3nguin_ | That's obviously for a 10-digit outgoing number. |
| 03:15.58 | p3nguin_ | NANP |
| 03:17.17 | hipitihop | so no credntials are specified appart from the provider domain |
| 03:18.24 | p3nguin_ | That depends. |
| 03:19.23 | p3nguin_ | If you are registering dynamically, you'll also be providing a username and secret in your peer definition. |
| 03:19.42 | p3nguin_ | If you are statically registered, you might not be sending the credentials. |
| 03:20.08 | p3nguin_ | It also depends on if their system requires authentication on each call. |
| 03:20.11 | hipitihop | I register via a register entry in my [general] section |
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| 03:21.04 | hipitihop | does that mean I don't dynamically register ? |
| 03:21.14 | p3nguin_ | You'll probably want to use a username and secret in your peer definition unless someone has specifically told you not to. |
| 03:21.30 | p3nguin_ | Using a register statement is for dynamic registration. |
| 03:22.07 | p3nguin_ | That is how you tell your ITSP where you are and where to send calls when it receives a call on your DID. |
| 03:22.08 | hipitihop | ok.. I guess that makes sense and means my system regiters to take inbound calls. |
| 03:22.49 | p3nguin_ | Without the register statement, they would not know where you are and how to get a call to you. |
| 03:22.55 | hipitihop | so that also means, I probably need credentials specified in the outgoing context |
| 03:23.14 | p3nguin_ | No, you put the credentials in the sip peer definition. |
| 03:23.27 | p3nguin_ | What ITSP are you using? |
| 03:23.39 | hipitihop | gotalk in Australia |
| 03:26.46 | p3nguin_ | I'm not seeing Asterisk configuration samples on their site, so we'll have to do it a different way. |
| 03:27.17 | hipitihop | this is what I get using your extension sample http://pastebin.ca/1804374 |
| 03:27.57 | p3nguin_ | Put it in pastebin.com if you would like for me to view it. |
| 03:28.09 | p3nguin_ | I still can't figure out the fascination with .ca |
| 03:28.59 | hipitihop | sorry, only used it because someone here asked me to another time so was just assuming it was this channels preference :-) |
| 03:30.04 | hipitihop | ok, here http://pastebin.com/m1b4ea87 |
| 03:30.14 | p3nguin_ | I can't figure out why my browsers won't like it unless I use its IP address. It's not a DNS issue, because I can dig or host lookup the IP address... but that's just a bother to do all the time. |
| 03:31.08 | p3nguin_ | It looks like gotalk didn't know how to call 101. |
| 03:31.16 | p3nguin_ | I hope you understand the problem with that. |
| 03:31.55 | p3nguin_ | Let's try it a little different way. One moment while I write up some rules. |
| 03:31.56 | hipitihop | I get the same regardless of the number I try and call |
| 03:33.10 | p3nguin_ | Give me an example of a phone number that you dial to reach someone when calling them on POTS/PSTN. |
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| 03:34.46 | p3nguin_ | 03 7010 5678 ? |
| 03:36.07 | hipitihop | yes, although don't know if that is a valid number... also mobile calls might start with 0402xxxxxx |
| 03:37.11 | hipitihop | would the dialplan string from my ATA which had no problems making callings prior to asterisk help ? |
| 03:38.37 | p3nguin_ | http://pastebin.com/d37a08915 |
| 03:39.31 | p3nguin_ | This assumes your sip peer is listed as [gotalk] |
| 03:40.02 | p3nguin_ | [gotalk] will contain type=peer along with your username and secret. |
| 03:40.14 | hipitihop | aah, maybe that's where I am going wrong... my peer in the sip conf is [mysipuserid] |
| 03:40.46 | p3nguin_ | It can be an arbitrary string, but it should make sense to you or anyone who you want to understand it. |
| 03:41.46 | hipitihop | but your point is that whatever string is in the exten => section .... @blah must be [blah] in the sip peer definition |
| 03:42.22 | p3nguin_ | If you want the Dial() command to use that peer and the credentials configured within it, yes. |
| 03:42.35 | hipitihop | seems to obvious ;-) |
| 03:46.49 | p3nguin_ | http://pastebin.com/d278bf445 |
| 03:48.36 | p3nguin_ | That incoming line sends calls to a SIP device with the name of 100. It's just an example, since your phone probably isn't 100. |
| 03:49.59 | hipitihop | yes I think I follow that... my exsisting one is an [internal] context and it dails my ATA defined as friend in sip.conf |
| 03:50.27 | hipitihop | exten => s,1,Dial(SIP/LinksysPAP) |
| 03:50.59 | p3nguin_ | The ATA should also be type=peer. |
| 03:51.46 | hipitihop | even though the handset off it will initiate most calls ? |
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| 03:51.58 | p3nguin_ | I don't see how that is relevant. |
| 03:52.00 | hipitihop | but as you said, peer does allow both directions |
| 03:53.45 | p3nguin_ | The difference from a call standpoint is how the phone is authenticated. user/friend is by username, peer is by IP address and port. |
| 03:54.21 | p3nguin_ | The different in a programming standpoint is that friend creates two items in memory. |
| 03:57.35 | p3nguin_ | And you won't want to use the 's' extension because a phone number will be called. The s extension is for calls without an explicit number being called. |
| 03:58.09 | p3nguin_ | exten => 100,1,Dial(SIP/LinksysPAP) |
| 03:58.31 | p3nguin_ | Now your extension is 100. Your device is still LinksysPAP. |
| 03:59.46 | p3nguin_ | This is a perfect example of how an "extension" is not a phone, like so many people try to say. |
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| 04:05.44 | hipitihop | p3nguin_, seems like we are getting close, no errors but I can't hear anything. |
| 04:06.28 | p3nguin_ | Is your system behind NAT? |
| 04:06.59 | hipitihop | sure but had no problem earlier when incoming was working so I haven't changed anything |
| 04:07.08 | p3nguin_ | Oh. |
| 04:07.22 | p3nguin_ | So incoming calls worked earlier? |
| 04:07.36 | hipitihop | and router is setup to forward anything on 5060 back to my * box |
| 04:07.43 | p3nguin_ | They had two-way audio, also? |
| 04:08.00 | hipitihop | yes incoming worked and two way audio was perfect. |
| 04:08.16 | p3nguin_ | You also have to forward the RTP ports. But if you had audio, I guess you did that already. |
| 04:08.42 | p3nguin_ | What's the status now? Incoming calls don't work? |
| 04:08.48 | hipitihop | I just took a call from my partner on my mobile and she said she can't call in .. so I have broken it :-) |
| 04:09.05 | hipitihop | no I had done nothing previously with the rtp ports. |
| 04:09.13 | p3nguin_ | hmm |
| 04:09.30 | hipitihop | so I'm going to have to go back to basics and review the recent changes |
| 04:09.44 | p3nguin_ | How could audio work if the RTP ports aren't forwarded? |
| 04:10.01 | hipitihop | I'm just going to flick the ATA back to talking direct with my provider |
| 04:11.27 | hipitihop | p3nguin_, agree with you, if forwarding the rtp ports, I assume you are talking about that 20000 range, then I have not done so and I swear the incoming test did work fine |
| 04:12.03 | p3nguin_ | 10000-20000 should be default. You can reduce it down if you aren't going to have 5000 simultaneous calls. |
| 04:12.46 | hipitihop | are thes udp tcp or both ? |
| 04:12.51 | p3nguin_ | all UDP |
| 04:13.05 | p3nguin_ | Just forward the same ports as what you have configured in rtp.conf. |
| 04:14.20 | p3nguin_ | Since you are behind NAT, you should have also configured nat= localnet= and externip= (or externhost=) in sip.conf. |
| 04:16.40 | hipitihop | hmm ok, as I said, all broken at the moment I'll have to backtrack and then apply what you have tought me... I really do appreciate your time |
| 04:17.15 | hipitihop | ..sorry but have to pop out for 30 min or so |
| 04:17.45 | p3nguin_ | No problem. Hopefully we'll be able to get it all worked out pretty soon. |
| 04:18.09 | hipitihop | what timezone are you in p3nguin_ |
| 04:18.17 | p3nguin_ | CST |
| 04:18.33 | p3nguin_ | (USA) |
| 04:18.50 | p3nguin_ | GMT -6 |
| 04:19.09 | hipitihop | so getting late for you |
| 04:19.18 | p3nguin_ | 22:19 |
| 04:19.53 | hipitihop | ok back in 30 |
| 04:20.20 | *** join/#asterisk skrite (~skrite@69.55.26.25) |
| 04:20.53 | skrite | hey all |
| 04:21.28 | *** join/#asterisk coppice (~chatzilla@94.201.17.210.dyn.pacific.net.hk) |
| 04:24.14 | ChannelZ | ok I am simultaneously fascinated and frustrated by watching curling |
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| 04:24.55 | kc8pxy | ChannelZ: frustrated? |
| 04:26.07 | ChannelZ | I have no idea what is going on |
| 04:26.47 | kc8pxy | ChannelZ: iirc it's something between bowling and darts. |
| 04:27.25 | p3nguin_ | I was thinking like bowling and shuffleboard, but I can kinda see darts. |
| 04:27.26 | ChannelZ | Like a GB team just went and pretty much landed a rock right in the center.. but as they were doing it, one of the guys from the OTHER team (CAN) was brushing the ice right as it was hitting the center |
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| 04:28.01 | p3nguin_ | He was probably trying to polish the ice so it would keep going all the way through the circle. |
| 04:29.00 | ChannelZ | so the other team is allowed to interfere? |
| 04:29.12 | p3nguin_ | To an extent, I guess. |
| 04:30.58 | ChannelZ | I see now |
| 04:31.03 | ChannelZ | is reading a bit on wikipedia |
| 04:31.21 | ChannelZ | The other team can only jump in once the stone crosses the line through the middle |
| 04:32.07 | ChannelZ | I also thought the sweeping was to rough up the ice and slow the thing down but apparently it's the opposite |
| 04:32.11 | p3nguin_ | And the sending team cannot cross that line, right? |
| 04:32.34 | ChannelZ | Well it says once it crosses the line only one person can brush, not both |
| 04:35.37 | p3nguin_ | Both guys brushed that one the whole way down. |
| 04:38.54 | p3nguin_ | It's pretty neat how they make it curl and change its path. |
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| 04:43.03 | Katty | wanders in |
| 04:44.14 | jaytee | hi |
| 04:44.37 | Katty | hi jaytee |
| 04:45.30 | eppigy | HELLO |
| 04:45.32 | eppigy | I AM DAVE |
| 04:45.40 | jaytee | yes you are! |
| 04:45.48 | jaytee | :-) |
| 04:45.49 | Katty | are you sure |
| 04:45.56 | Katty | are you sure you're not... Bob |
| 04:45.59 | Katty | or Mary |
| 04:48.22 | p3nguin_ | or a name less common, like Waldo? |
| 04:48.40 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-rhrfdrafohqnlaug) |
| 04:49.54 | p3nguin_ | hipitihop: Time's up! If you aren't here, you're late. |
| 04:50.04 | hipitihop | :-) just arrived |
| 04:50.20 | p3nguin_ | Damn, you said 30 minutes and you meant it. |
| 04:51.58 | hipitihop | ok testing inbound I get this trace NOTICE[2373]: chan_sip.c:19546 handle_request_invite: Call from '09xxxxx' to extension 's' rejected because extension not found. |
| 04:52.15 | p3nguin_ | Okay, your ITSP doesn't know to send your phone number to you. |
| 04:52.30 | p3nguin_ | You can probably remedy that by adding it to the register statement. |
| 04:52.56 | p3nguin_ | register => username:secret@sip.gotalk.com/yourphonenumber |
| 04:53.09 | hipitihop | trying... |
| 04:53.43 | p3nguin_ | That's how I solved it when sipgate used to send no phone number to me. |
| 04:53.45 | hipitihop | although not sure to include area codes and or country prefix |
| 04:54.05 | p3nguin_ | You don't know your DID number? |
| 04:54.31 | hipitihop | not sure what DID is .. I know my userid and I know my phone number. |
| 04:54.58 | p3nguin_ | Your DID is the phone number that someone would dial from the PSTN to reach you. |
| 04:55.24 | p3nguin_ | It typically includes the area code, but not the country code. |
| 04:55.41 | hipitihop | ok, however should I include country prefix also e..fg Aus = 61 |
| 04:55.53 | hipitihop | sorry, ok |
| 04:56.24 | p3nguin_ | It annoys me when ITSPs don't know what phone number they should be sending. |
| 04:58.12 | hipitihop | ok that has changed somehting.. now I get NOTICE[2373]: chan_sip.c:17269 handle_response_invite: Failed to authenticate on INVITE to '"04xxxx" |
| 04:58.12 | *** join/#asterisk GameGamer43|Mac (~GameGamer@CPE-65-27-76-78.new.res.rr.com) |
| 04:58.42 | p3nguin_ | Okay, uncomment that line in your peer that says insecure= |
| 04:58.47 | *** join/#asterisk moy (~moy@CPE001cdfec4cee-CM00080dab8485.cpe.net.cable.rogers.com) |
| 04:59.04 | hipitihop | already uncommented, has incesure=invite |
| 04:59.13 | Katty | holy friggin snickderdoodles batman, it's 11 |
| 04:59.46 | p3nguin_ | It is already insecure=invite and still produced that message? |
| 05:00.38 | hipitihop | p3nguin_, yes if we are talking about sip.conf and my gotalk pper section |
| 05:02.58 | hipitihop | p3nguin_, although I have not followed your pastebin to the letter, still tweaking with my original so I hope that is causing confusion |
| 05:03.08 | hipitihop | ^ not |
| 05:05.51 | p3nguin_ | While I did provide the paste as a guideline, it should work if you used it copy/paste into your conf. |
| 05:09.11 | *** join/#asterisk GameGamer43|Mac (~GameGamer@CPE-65-27-76-78.new.res.rr.com) |
| 05:12.07 | hipitihop | p3nguin_, my current one http://pastebin.com/m2fd79cae |
| 05:13.15 | p3nguin_ | How do you plan to call "LinksysPAP" from a phone? |
| 05:13.24 | p3nguin_ | I mean, how do you intend to dial it? |
| 05:14.30 | hipitihop | not sure what you mean... I want in general, any call coming in from gotalk to be routed to the ATA so it rings and we pickup |
| 05:14.46 | p3nguin_ | http://pastebin.com/d5c46dfe8 |
| 05:15.10 | hipitihop | sorry if I'm being a complete newbie, I'm sure the penny will drop soon |
| 05:16.10 | p3nguin_ | You cannot pick up your phone and dial "LinksysPAP" on it. |
| 05:16.19 | p3nguin_ | So don't make an extension called "LinksysPAP". |
| 05:16.58 | p3nguin_ | Dial(SIP/LinksysPAP) makes the call go to the ATA. |
| 05:16.59 | hipitihop | is that what the internal thing does ? |
| 05:17.17 | p3nguin_ | See my post. I corrected yours. |
| 05:18.25 | p3nguin_ | At this point, the internal context is irrelevant. But I want you to understand that no one can pick up their phone and dial L i n k s y s P A P on it. |
| 05:19.30 | p3nguin_ | They could press those keys, but it would come out as 5465797727. |
| 05:19.34 | hipitihop | you mean from a logicl pov ? iow, if I was using a softphone, could I not treat the ATA with a ahndset on it as a normal phone/extension ? |
| 05:20.09 | p3nguin_ | So it would need to be exten => 5465797727,1,Dial(SIP/LinksysPAP) if that's what your intention was. |
| 05:20.32 | p3nguin_ | phones are not extensions. |
| 05:21.00 | p3nguin_ | Extensions are RULES on what to do when Asterisk receives the phone numbers or letters that are being sent for a call. |
| 05:21.57 | p3nguin_ | exten => 5465797727,1,Dial(SIP/LinksysPAP) <-- This says if you dial 5465797727 on a phone, it will place a call to the SIP device by the name of LinksysPAP. |
| 05:22.07 | p3nguin_ | The extension is 5465797727. |
| 05:22.14 | p3nguin_ | The device is LinksysPAP. |
| 05:22.16 | hipitihop | indeed... I just because I have these things in my config does not mean it is an intention :-) these configs are a combination of samples, parts from THE book etc .. my only intention at the moment is to be able to take and make calls from my ATA via my sip account |
| 05:23.00 | p3nguin_ | If you don't understand this basic facts, you won't be making and receiving calls. |
| 05:23.08 | p3nguin_ | s/this/these/ |
| 05:23.37 | p3nguin_ | I have given you the dialplan, but for some reason you choose to ignore it. |
| 05:24.26 | jaytee | kinda reminds me of jmcdowell |
| 05:24.45 | p3nguin_ | Heh, yeah. Did you know he put tk on ignore? |
| 05:25.02 | hipitihop | p3nguin_, I'm not ignoring it and I don't intend to offend either.. I'm jsut trying to understand as I go instead of blindly pasting |
| 05:25.04 | jaytee | hahaha, nothing like cutting off your nose to spite your face |
| 05:28.28 | p3nguin_ | hipitihop: Maybe I can go over it one more time to help you understand it. LinksysPAP is your device's name, as configured in sip.conf (at least that's what you told me). That is not your extension, it is your device. Extensions are not devices. exten => 5465797727,1,Dial(SIP/LinksysPAP) creates extension 5465797727 and makes it dial your device if 5465797727 is called. |
| 05:29.22 | p3nguin_ | I doubt that you really want someone to have to press in 5465797727 from another phone on your system, though, so I choose to create extension 100 to dial your device instead. |
| 05:29.31 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 05:29.49 | p3nguin_ | http://pastebin.com/d5c46dfe8 |
| 05:30.05 | jaytee | hipitihop, make sure you carefully and fully read Chapter 5 in the book if you want to understand your dialplan, contexts, etc. |
| 05:30.48 | jaytee | it will help you avoid errors moving forward when you want to do more advanced call processing |
| 05:31.10 | p3nguin_ | Extensions are just rules to say what to do when numbers (or letters) are received. |
| 05:32.32 | p3nguin_ | Asterisk receives a call to 2123, it knows to call my phone because of this rule: exten => 2123,1,Dial(SIP/mydevice,48) |
| 05:32.45 | p3nguin_ | My extension is 2123, my device is "mydevice". |
| 05:33.17 | p3nguin_ | And it is using chan_sip to reach it, as designated by the "SIP/" part of the Dial() command. |
| 05:33.35 | hipitihop | yup that makes sense now |
| 05:34.06 | *** join/#asterisk graphix (gfx@38.111.17.107) |
| 05:34.42 | p3nguin_ | exten => _X.,1,Dial() <-- this extension is a pattern match, matching ANY two or more numbers. |
| 05:34.48 | hipitihop | ok dial plan 100% as per yours... still getting invite error though |
| 05:34.59 | p3nguin_ | Of course, that is solved in sip.conf. |
| 05:35.17 | p3nguin_ | You still have insecure=port,invite ? |
| 05:35.44 | hipitihop | no have insecure=invite |
| 05:35.54 | p3nguin_ | Try port,invite and see what happens. |
| 05:36.08 | p3nguin_ | I also included that in my paste earlier. |
| 05:37.34 | hipitihop | ahh my bad totally missed the sip.conf part of that post , standby, matching |
| 05:45.42 | hipitihop | p3nguin_, ok some progress |
| 05:46.23 | hipitihop | p3nguin_, I can call out and I get audio, at least the remote end..getting to aother number's voice mail. |
| 05:46.58 | p3nguin_ | Your outgoing call works and has two-way audio? |
| 05:47.21 | hipitihop | p3nguin_, I'll paste the results of the incoming which seemed to now answer but my ATA did not ring and I couldn't hear anything |
| 05:48.13 | p3nguin_ | I'm going to need to see your sip.conf, your extensions.conf, and the debug output if you want me ot solve the problem. |
| 05:50.36 | hipitihop | p3nguin_, hold on just trying to get to an outside number wher eI can confirm I have both way.. so far onyl getting mailboxes so can confirm I can hear that end |
| 05:53.14 | hipitihop | p3nguin_, confirmed, two way at least to a mobile/cell works and good clear audio both ways. |
| 05:54.04 | p3nguin_ | Okay, that's good. So you're using exten => _X.,1,Dial(SIP/${EXTEN}@gotalk) for outgoing? |
| 05:54.28 | hipitihop | yes |
| 05:54.56 | p3nguin_ | Depending on the different ways you can dial and depending on what gotalk will accept, you WILL have to refine that later. |
| 05:55.09 | hipitihop | sure |
| 05:55.35 | hipitihop | now, incoming exten => _X.,1,Dial(SIP/LinksysPAP) |
| 05:55.49 | p3nguin_ | I have at least three different outgoing extens for 7, 10, and 11-digit dialing for outgoing calls. |
| 05:56.48 | *** join/#asterisk GameGamer43|Mac (~GameGamer@CPE-65-27-76-78.new.res.rr.com) |
| 05:56.56 | p3nguin_ | such as exten => NXXXXXX,1,Dial(SIP/1312${EXTEN}@voipms) |
| 05:56.59 | hipitihop | now tried incoming again and same, I get a trace in the cli with ringing and answered ... but as mentioned it doesn't ring the ATA so stanby and prep another pastebin |
| 05:57.22 | p3nguin_ | and exten => NXXNXXXXXX,1,Dial(SIP/1${EXTEN}@voipms) |
| 05:57.43 | *** part/#asterisk GameGamer43|Mac (~GameGamer@CPE-65-27-76-78.new.res.rr.com) |
| 05:57.43 | p3nguin_ | err, forgot my underscores in there. |
| 05:58.27 | p3nguin_ | I didn't forget them in the pastes I made for you, though. |
| 05:59.55 | *** join/#asterisk nickaugust (~anonymous@216-160-175-100.hlrn.qwest.net) |
| 06:00.23 | hipitihop | sorry I'm so slow.. working with my remote * box so only have nano over ssh at the moment so copy pasting is a bit of a multi-part thing |
| 06:01.31 | p3nguin_ | You could always use pastebinit. |
| 06:02.02 | p3nguin_ | It's tricky when you want to hide your passwords, though. You have to make sure you hide them before pasting. |
| 06:02.55 | hipitihop | thanks I'll make a note to look at that.. obviously I don't want to just paste real sip.conf becauxse all secrets are there |
| 06:03.19 | p3nguin_ | That's a pastebin tool for the command line, if you didn't know. |
| 06:07.11 | hipitihop | p3nguin_, http://pastebin.com/m18847c6b |
| 06:07.25 | ChannelZ | we need an asterisk-aware pastebin that will automatically greek passwords and IPs |
| 06:07.49 | p3nguin_ | Not IPs, that makes troubleshooting a network-related issue difficult. |
| 06:08.07 | ChannelZ | yeah but sometimes people are pussies about it and want to change them |
| 06:08.09 | hipitihop | so from what I understand, I suspect my problem is related to the context defined in LinksysPAP, the context is "phones" however the internal is currently commented out in phones. |
| 06:08.43 | p3nguin_ | hipitihop: You aren't making any call from the device to other internal devices, so it is irrelevant. |
| 06:09.43 | p3nguin_ | Where's your gotalk peer definition? |
| 06:09.59 | *** join/#asterisk sysreq (~sysreq@unaffiliated/sysreq) |
| 06:10.00 | hipitihop | left it out as it is as per your spec |
| 06:10.20 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 06:10.23 | p3nguin_ | And if you're natted like you told me, where's your nat, localnet, and externip or externhost? |
| 06:10.55 | hipitihop | <shrug> I need that in [general] in sip.conf ? |
| 06:11.04 | *** join/#asterisk dkirker-openmobl (~dkirker@openmobl/ceo/dkirker) |
| 06:11.14 | p3nguin_ | Every time you paste me incomplete configs, it's wasting more time that we could be using to get this fixed. |
| 06:11.43 | p3nguin_ | It's after 00:00 now, so someone else will have to take over soon if you don't hurry up. |
| 06:11.54 | hipitihop | p3nguin_, sorry, tried to save time as the peer laso had many secrets, standby. |
| 06:12.35 | p3nguin_ | This is seriously a 10 minute job. We've been discussing it for hours so far. |
| 06:13.43 | hipitihop | http://pastebin.com/m4df4436f |
| 06:14.05 | p3nguin_ | context=incoming-calls |
| 06:14.08 | p3nguin_ | [incoming_calls] |
| 06:14.16 | p3nguin_ | These do not coincide. |
| 06:14.33 | p3nguin_ | They are supposed to. |
| 06:15.02 | hipitihop | eek, indeed, typo |
| 06:15.03 | p3nguin_ | And you still don't have any nat info. |
| 06:15.24 | p3nguin_ | If you are natted, you HAVE TO HAVE the nat stuff configured. |
| 06:15.27 | p3nguin_ | ~sipnat |
| 06:15.27 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 06:16.03 | p3nguin_ | nat= localnet= externip= (or externhost= ) |
| 06:16.30 | p3nguin_ | They go in the general section. |
| 06:21.09 | p3nguin_ | Actually, there are a few other settings I would configure, as well. |
| 06:23.20 | hipitihop | p3nguin_, http://pastebin.com/m58d5a972 |
| 06:25.25 | p3nguin_ | That outgoing call there at the bottom. It was successful? |
| 06:25.28 | *** join/#asterisk CrashHD (~CrashHD@c-76-114-27-96.hsd1.ca.comcast.net) |
| 06:25.48 | hipitihop | that's an incoming attempt from my mobile to my sip number |
| 06:26.16 | hipitihop | same number as we appended to the registration |
| 06:26.17 | p3nguin_ | You still haven't corrected the context. |
| 06:26.22 | p3nguin_ | context=incoming-calls |
| 06:26.27 | p3nguin_ | IT is wrong. |
| 06:26.40 | hipitihop | sorry, didn't repaste the new one.. yes corrected to use _ |
| 06:27.31 | p3nguin_ | "sip reload" and "dialplan reload" and then show me the entire failed incoming call. |
| 06:29.03 | p3nguin_ | So far, that piece of a call was not an incoming call. |
| 06:30.18 | hipitihop | same as before. |
| 06:30.32 | hipitihop | are you saying this looks like an outgoing call ? |
| 06:30.37 | p3nguin_ | I don't know what that means. Show me something useful. |
| 06:30.49 | p3nguin_ | <PROTECTED> |
| 06:30.52 | p3nguin_ | <PROTECTED> |
| 06:30.56 | p3nguin_ | That's an outgoing call. |
| 06:31.05 | idespinner | hipitihop, i looked at the pastebin , looks like an outgoing call to me aswell |
| 06:31.08 | p3nguin_ | Show me something useful for a failed incmoing call. |
| 06:32.15 | *** join/#asterisk albasheers (~basheer@89.148.43.51) |
| 06:32.22 | *** part/#asterisk albasheers (~basheer@89.148.43.51) |
| 06:32.56 | p3nguin_ | If what you put in that pastebin contained your entire [general] section of sip.conf, I would be surprised if anything works. |
| 06:33.00 | hipitihop | http://pastebin.com/m14a1446 |
| 06:33.25 | p3nguin_ | That's still an incoming call. |
| 06:33.42 | p3nguin_ | Maybe you don't know how to enable sip debuggin? |
| 06:33.54 | p3nguin_ | Err, still not an incoming call. |
| 06:34.20 | hipitihop | p3nguin_, are we talking cross purposed ? ... I am trying to call my asterisk from a mobile phone, so I would expect it to come up as an incoming call |
| 06:34.22 | p3nguin_ | sip set debug or sip set debug on (depending on asterisk version). |
| 06:34.35 | hipitihop | I'm on 1.6 |
| 06:34.49 | p3nguin_ | That's what I'm talking about. Incoming... from the PSTN to your DID number. |
| 06:35.02 | p3nguin_ | What you showed me is an outgoing call. |
| 06:35.38 | hipitihop | p3nguin_, it is what comes up in teh cli when I attempt to call in from my mobile.... standby enabling debug |
| 06:35.42 | idespinner | hipitihop, could you do a 'sip show peer gotalk' ? |
| 06:36.36 | p3nguin_ | I'm still wondering if sip reload wasn't issued after making changes. |
| 06:37.38 | idespinner | p3nguin_, thats what i'm thinking aswell, a sip show peer might unconver this... |
| 06:37.42 | p3nguin_ | yep |
| 06:37.43 | hipitihop | idespinner, http://pastebin.com/m2e78a1e5 |
| 06:37.57 | p3nguin_ | <PROTECTED> |
| 06:38.07 | hipitihop | p3nguin_, I reloaded sip and dialplan |
| 06:38.09 | p3nguin_ | So that little tidbit wasn't an incoming call. |
| 06:39.33 | idespinner | hipitihop, looks good there, could you also do a 'sip show peer LinksysPAP' ? |
| 06:39.45 | hipitihop | is feeling very guilty, after wasting p3nguin's time and now has two people engaged on an trivial excercise |
| 06:40.09 | p3nguin_ | I'm still waiting on a sip debug from a failed incoming call. |
| 06:41.15 | hipitihop | idespinner, http://pastebin.com/m6142c13d |
| 06:41.21 | hipitihop | p3nguin_, getting to that now |
| 06:43.38 | idespinner | hipitihop, yea, that second sip show peer looks good aswell. definatley need a full sip debug of you calling into your gotalk number with your cellphone |
| 06:44.08 | hipitihop | yup got the debug, pages of it, sanitising now |
| 06:48.20 | hipitihop | http://pastebin.com/m53291615 and I hope it's clean enough |
| 06:49.14 | p3nguin_ | Looking for 07xxxxxxxx in phones (domain 203.xxx.xxx.xxx) |
| 06:49.21 | p3nguin_ | I don't get it. |
| 06:49.47 | idespinner | yea... |
| 06:49.54 | p3nguin_ | I want you to show me your ENTIRE sip.conf. |
| 06:50.06 | p3nguin_ | Stop leaving pieces out. The whole thing is needed. |
| 06:50.23 | idespinner | other context's could be conflicting |
| 06:50.31 | p3nguin_ | By the way, this is the last time I will be asking for it. |
| 06:50.43 | p3nguin_ | I think a dozen times is plenty. |
| 06:51.50 | hipitihop | I thought I had done that in previous pastebins but standby... I understand it's late and I'm frustrating you. |
| 06:52.22 | p3nguin_ | I have five ITSPs configured on my system, and I don't have this same issue with any of them. |
| 06:55.57 | hipitihop | http://pastebin.com/m485d1aa4 |
| 06:56.10 | *** join/#asterisk benngard (~benngard@90-230-92-67-no148.tbcn.telia.com) |
| 06:58.11 | idespinner | the only thing I can think of is that the IP address of gotalk is being registered as 2 seperate peers, and one of them has the context of [phones]... |
| 06:58.38 | idespinner | you're not registered to gotalk twice are you? |
| 06:59.00 | p3nguin_ | sip show registry? |
| 06:59.18 | idespinner | maybe... thats a good thought... |
| 06:59.41 | hipitihop | sip.gotalk.com:5060 N myuserid 105 Registered Sun, 21 Feb 2010 16:57:53 |
| 07:00.01 | idespinner | sip show peers? |
| 07:00.11 | hipitihop | just checking my laptop to make sure nothing is logged in either |
| 07:02.06 | hipitihop | p3nguin_, as I mentioned quite some time back, I had no problem with incoming, the ATA would ring and I had audie for both ends... I started today trying to solve outgoing. now we seem to have reversed the situation |
| 07:04.07 | hipitihop | but I'm no where near comfortable enough with all this to know which changes have had what effect... certainly appending my did to the registration stopped the need for 's' and only other main things is we switched a couple of things from friends to peers |
| 07:04.42 | idespinner | hipitihop, basicaly the asterisk cli logs you post for incoming is not what I would expect as the incoming call is hitting the [phones] or [go-talk] context instead of the [incoming-calls] context.... |
| 07:05.39 | idespinner | so from what you posted here: http://pastebin.com/m58d5a972 it looks like SIP/myuserid-dc0f6198 is dialing SIP/07xxxxxxxx@gotalk |
| 07:05.47 | p3nguin_ | yep |
| 07:05.49 | p3nguin_ | exactly |
| 07:06.06 | idespinner | is SIP/myuserid-dc0f6198 your PAP2 or is it gotalk or is it anything you can identify? |
| 07:06.32 | idespinner | im assuming you scrubbed 'myuserid' in there |
| 07:06.44 | hipitihop | yes, that is my real gotalk userid |
| 07:07.13 | hipitihop | so wherever you see myuserid it is a replacment for the real gotalk one |
| 07:07.49 | idespinner | ok, so for whatever reason, incoming calls from gotalk are hitting you outgoing context(phones or gotalk), despite that your context set in sip.conf for gotalk is set to incoming-calls |
| 07:08.11 | hipitihop | if you look on my scrubbed sip.conf the registration ... has the same number but I called it 'userid' in the registration |
| 07:09.57 | *** join/#asterisk benngard (~benngard@90-230-92-67-no148.tbcn.telia.com) |
| 07:10.35 | hipitihop | although note the context has been fixed in http://pastebin.com/m485d1aa4 |
| 07:11.20 | idespinner | hipitihop, if we removed the denis context completley and did a sip reload... |
| 07:11.30 | idespinner | im wondering if that resolves the issue... |
| 07:11.39 | *** join/#asterisk albasheers (~basheer@89.148.43.51) |
| 07:11.56 | idespinner | its a long shot, but i dont have to much else to build on |
| 07:13.46 | hipitihop | nope |
| 07:14.33 | idespinner | is the asterisk cli different? |
| 07:15.17 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 07:15.46 | hipitihop | trying to find original debug to compare standby |
| 07:15.48 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 07:16.18 | *** part/#asterisk albasheers (~basheer@89.148.43.51) |
| 07:17.53 | p3nguin_ | Here's a working sip.conf with one ITSP and one phone, behind NAT: http://pastebin.com/d53d2f344 |
| 07:18.25 | *** join/#asterisk Sargun (~Sargun@atarack/Staff/Sargun) |
| 07:22.36 | *** join/#asterisk ehsjoar (~ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 07:22.56 | hipitihop | p3nguin_, do you want to match that as best possisble ? or do you want me to take it as is and substitute my ip's, userid etc. |
| 07:27.25 | hipitihop | to my untrained eye the debug and end result certainly the same |
| 07:35.23 | hipitihop | ok p3nguin_and idespinner, sorry to take up so much of your time.. can't thank you enough at your attempts to help |
| 07:36.04 | hipitihop | I'll switch my ATA back to direct sip provider for now so I can have normal calls again |
| 07:39.25 | hipitihop | is there a general asterisk cli command to stop all sip related activity so it's not trying to register or receive calls at the same time as my ATA |
| 07:40.43 | p3nguin_ | module unload chan_sip |
| 07:44.24 | hipitihop | ok, thanks again for help, have to run out to pickup from airport ... sorry to have been so much of a pita |
| 07:45.02 | p3nguin_ | Here's a working sip.conf with two ITSPs and one phone, behind NAT: http://pastebin.com/d4bcd86c1 |
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| 07:48.01 | hipitihop | yes I looked at that p3nguin thanks.... when I return I will try and apply as much from that as possisble obviously with my own ip's and userids etc |
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| 09:41.23 | FSB_1 | How can I make asterisk play a dial tone? |
| 09:41.48 | FSB_1 | I am trying to make it so that if I press a double 0 I will be able to make outbound calls. |
| 09:42.04 | ChannelZ | core show application PlayTones |
| 09:42.14 | ChannelZ | oh |
| 09:42.18 | ChannelZ | well it doesn't really work like that |
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| 09:43.02 | ChannelZ | I suppose you could set extension 00 to do PlayTones to give you a phony dialtone, and then WaitExten for the number |
| 09:43.45 | FSB_1 | Am I thinking right if I use extension 00 to goto another context which is allowed to make outbound calls? |
| 09:43.48 | ChannelZ | but ideally you'd want to stop the dialtone after someone presses the first digit |
| 09:44.16 | ChannelZ | Well why do you want to make people do this 00 business anyway? |
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| 09:45.33 | FSB_1 | It's a common thing to do in sweden. |
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| 09:48.55 | ChannelZ | are these sip phones or analog phones? |
| 09:49.54 | FSB_1 | Right now I'm just playing around, but it's a common thing to have the telephone switches configured like this. |
| 09:50.45 | ChannelZ | we have sort of a similar thing in the US for business PBXs, to dial 9 for an outside line |
| 09:51.12 | ChannelZ | on SIP phones though you can usually program the phone's dialplan to do this.. so that when you pick up and dial 9, the phone it's self makes a new dialtone |
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| 09:59.14 | *** join/#asterisk vk4akp (~Ken@c114-77-251-186.ipswc3.qld.optusnet.com.au) |
| 09:59.23 | vk4akp | Hi guys. |
| 09:59.31 | vk4akp | Anyone around? |
| 10:00.03 | vk4akp | I have a really different / strange question realating to Asterisk. |
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| 10:13.00 | vk4akp | All these names and no one here? |
| 10:13.46 | coppice | you might find asking a question is more productive |
| 10:16.18 | tzafrir | FSB_1, this is a rather simple thing to do: basically a context with WaitExten |
| 10:16.33 | tzafrir | Technically it is a sort of an IVR with Asterisk |
| 10:16.56 | tzafrir | (same implementation: basic dialplan) |
| 10:17.06 | vk4akp | Ah |
| 10:17.08 | vk4akp | People. |
| 10:17.10 | vk4akp | :) |
| 10:17.22 | tzafrir | infobot, tell vk4akp about ask |
| 10:17.30 | tzafrir | (bots as well) |
| 10:18.07 | vk4akp | OK. We are trying to get a Asterisk MeetMe conference running on an install that is on a virtual server provided by a hosting company. |
| 10:18.32 | vk4akp | The issue is Ztdummy. |
| 10:18.41 | vk4akp | Is there a way to do the timing? |
| 10:18.48 | tzafrir | Have you tried using DAHDI instead? Specifically: latest DAHDI? |
| 10:18.54 | vk4akp | Because I'm not sure if the normal kernal way will work on a virtual server. |
| 10:19.07 | tzafrir | (Enable core timing) |
| 10:19.15 | vk4akp | I haven't tried anything yet. Looking for advice on the best way to proceede. |
| 10:19.27 | tzafrir | What version of Asterisk is it? |
| 10:19.35 | vk4akp | OK. So I will looki into dahdi. |
| 10:19.41 | vk4akp | Hang ten I'll look for the version. |
| 10:20.33 | vk4akp | 1.2.27 |
| 10:22.09 | vk4akp | LOL. INfobot says you are all here against your will. LOL |
| 10:22.16 | vk4akp | Someone should proof read their bot. hahahah. |
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| 10:34.38 | vk4akp | OK. So more info needed. |
| 10:34.51 | vk4akp | will Dahdi run on the 1.2.27 release? |
| 10:35.19 | vk4akp | Is ther ea link or direction that explains where to get and how to install Dahdi dummy? (Name?) |
| 10:37.50 | vk4akp | Humm. Looks like it might be called Dahdi_dummy ??? |
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| 10:44.47 | vk4akp | Hey guys any answer? |
| 10:45.05 | vk4akp | Need to get some info so I can request to move forward with this. |
| 10:46.33 | tzafrir | vk4akp, no. That version only works with Zaptel. But why do you use it? |
| 10:46.51 | vk4akp | I don't. |
| 10:46.53 | vk4akp | I run 1.4 |
| 10:47.02 | vk4akp | But this is another box on the other side of the world. |
| 10:47.16 | vk4akp | It has an old install of Gentoo and Asterisk on it. |
| 10:47.21 | vk4akp | Not my box basically. |
| 10:48.13 | vk4akp | But I have to add a conference on it for DonkeyBallz |
| 10:48.13 | vk4akp | So the answer is that I have to request to update to Asterisk 1.4.22 or higher yes? |
| 10:49.20 | tzafrir | vk4akp, if you want dahdi |
| 10:49.28 | vk4akp | OK |
| 10:49.36 | tzafrir | but before you do that: do you have a problem with building kernel modules? |
| 10:50.02 | vk4akp | And do you know for sure that Dahdi_Dummy in this (enable core timing) mode will work on a virtually hosted box? |
| 10:50.24 | vk4akp | Humm. Good question. This is my first time dealing with a virtually hosted box. |
| 10:50.59 | tzafrir | vk4akp, do you currently have dahdi running? |
| 10:51.09 | vk4akp | no |
| 10:51.21 | vk4akp | it's 1.2.27 remember. |
| 10:51.36 | tzafrir | you can try building (but not installing) dahdi-linux and dahdi-tools |
| 10:52.09 | vk4akp | what will that do? |
| 10:52.49 | vk4akp | What if I emerge them? |
| 10:52.56 | tzafrir | If you enable core timing in include/dahdi/dahdi_config.h , you'll only need to insmod dahdi.ko itself . Otherwise, you'll also need to insmod dahdi_dummy |
| 10:53.20 | vk4akp | OK Thats good info thanks. |
| 10:53.24 | tzafrir | a very basic test that it works, even without using (or building) dahdi-tools: |
| 10:53.25 | vk4akp | Let me doco that b4 I move on. |
| 10:54.23 | tzafrir | time head -c8000 /dev/dahdi/pseudo >/dev/null |
| 10:54.37 | vk4akp | Is Dahdi 2.2.02 new enough? |
| 10:54.38 | tzafrir | This should give you no errors, and should take ~1 second |
| 10:55.04 | tzafrir | if it gives an error or takes a time that is much different: something is wrong with the timing |
| 10:55.20 | tzafrir | I would prefer 2.2.1, |
| 10:55.32 | tzafrir | any special reason for not using it? |
| 10:56.00 | vk4akp | OK. I can wget and hand compile a newer one but 2.2.0.2 is available now in portage. If that is likely to work it's very quick for me to install it. |
| 10:56.12 | tzafrir | (note that what I suggest to you is to download the source tarball, build it, but *not* install anything from it) |
| 10:56.38 | vk4akp | OK. 2.2.1 it is then. Umm. LInk for SVN or somethign please? |
| 10:57.15 | tzafrir | http://downloads.asterisk.org/pub/telephony/dahdi-linux/ |
| 10:57.18 | tzafrir | or svn: |
| 10:57.45 | tzafrir | svn co http://svn.asterisk.org/svn/dahdi/linux/branches/2.2/ |
| 10:57.51 | vk4akp | tnx |
| 11:10.35 | vk4akp | <PROTECTED> |
| 11:10.43 | vk4akp | Is this th eline I uncomment before doing a make? |
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| 11:18.50 | tzafrir | vk4akp, yes, just uncomment that |
| 11:19.10 | tzafrir | (I hope to eventually make that enabled by default and do away with dahdi dummy) |
| 11:19.23 | vk4akp | OK |
| 11:19.29 | vk4akp | Yep done that and a make. |
| 11:19.49 | vk4akp | NOw I am a bit queezy about doing a insmod on a virtually hosted box. This won't kill anythign will it? |
| 11:22.41 | vk4akp | insmod: error inserting 'dahdi.ko': -1 Invalid module format |
| 11:25.41 | vk4akp | So do I need to do a make install now instead and then insmod dahdi_dummy instead? |
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| 11:42.09 | tzafrir | vk4akp, what is the error you actually see in the kernel logs? |
| 11:42.38 | vk4akp | OMG. I wouldn't even know where to look for that. |
| 11:42.50 | tzafrir | dmesg | tail |
| 11:42.59 | vk4akp | can't I just insmod dahdi_dummy ? |
| 11:43.00 | tzafrir | or the logs in /var/log |
| 11:43.01 | vk4akp | OK Looking. |
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| 11:43.18 | tzafrir | no. If you insmod, you have to resolve dependencies yourself |
| 11:44.24 | vk4akp | dahdi: disagrees about version of symbol struct_module |
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| 12:01.17 | vk4akp | Ah. Damn I know. The Kernel source revision doesn't match teh installed revision. :( |
| 12:01.19 | vk4akp | Grrr. |
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| 12:08.00 | vk4akp | The kernel version is 2.6.16.29.xs3.1.0.289.2650 |
| 12:08.15 | vk4akp | And I think that's a virtual kernel provided by the hosting company. |
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| 12:16.39 | tzafrir | vk4akp, hmm... so you can't run your own kernel? |
| 12:17.00 | vk4akp | MM. That is somethign I will have to look into and ask. |
| 12:17.22 | vk4akp | Becasue of the time difference the people who pay for the box are asleep right now. |
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| 12:17.35 | vk4akp | But I have given them all the information and what part we are up to. |
| 12:17.52 | vk4akp | I googled that kernel revision number and find it is a vmware kernel. |
| 12:17.59 | vk4akp | So the plot thickens. :) |
| 12:18.09 | vk4akp | It will be interesting to see if ther eis a work around for this. |
| 12:18.23 | vk4akp | I guess it is of reasonable iportance to Asterisk as well. |
| 12:18.37 | vk4akp | As it could mean that virtually hosted box's can't run a aconference. :( |
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| 12:23.19 | thazza | Hey All. |
| 12:36.41 | Amorsen | vk4akp: You're unlikely to get anywhere with 1.2.27 + vmware + meetme |
| 12:37.09 | vk4akp | Is there any way to create a timing source for meetme conference with out having access to the kernel source? |
| 12:37.26 | vk4akp | I'm fairly sure they will let me update teh asterisk to 1.4 |
| 12:37.54 | vk4akp | But I think the kernel source could be an issue. |
| 12:38.05 | Amorsen | Meetme is dependent on dahdi_dummy, not just for timing |
| 12:38.44 | vk4akp | OK so the stumbling block for now is the VM kernel + source etc. |
| 12:39.40 | Amorsen | Well I'll be impressed if you get good enough timing from a vmware guest that it will work decently, but I suppose it isn't completely impossible |
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| 12:40.36 | Amorsen | You could try a test with 1.6.2 and app_conference, that would remove the need for dahdi |
| 12:41.11 | tzafrir | vk4akp, rather: either use app_conference (an external app) or upgrade to 1.6.2 and use app_confbridge |
| 12:41.20 | Amorsen | Sorry, app_confbridge |
| 12:41.40 | vk4akp | OK. Interesting. |
| 12:41.46 | vk4akp | I will put that to them also. TNX> |
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| 12:44.14 | vk4akp | OK. Thanks for your help all tonight. |
| 12:44.21 | vk4akp | I think this ends my options for this evening. |
| 12:44.29 | vk4akp | All the Best! |
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| 13:13.31 | tuxx- | o-hai. |
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| 13:27.39 | Kobaz | haha |
| 13:27.40 | Kobaz | Katty: |
| 13:27.51 | Kobaz | Katty: chatroulette got a story on slashdot |
| 13:28.06 | Kobaz | Katty: via the new york times no less |
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| 15:42.27 | ProperPHB | ~savemoney |
| 15:42.28 | infobot | <Gremlin> It's okay, but I was hoping to save some money with a soldering iron and a bunch of sound cards. |
| 15:42.51 | ProperPHB | I'm proud of my money saving prowess. ;) |
| 15:43.18 | ProperPHB | Sorry I haven't been here in a while. Our debt collection call center got burned down by some arsons. |
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| 15:52.47 | ProperPHB | By the way, I'm Gremlin. |
| 15:53.10 | ProperPHB | I would be here as Gremlin, but the post it note on my monitor with my passwords was lost in the fire. |
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| 15:54.51 | Pan3D | sorry about the fire, I need to make marshmallows |
| 15:55.25 | tzafrir_laptop | ProperPHB, arson? Are you sure it's not your soldering iron? |
| 15:57.16 | ProperPHB | Well, we're working out of my boss's apartment now doing boiler room cold calling. |
| 15:57.34 | tzafrir_laptop | ProperPHB, anyway, can't you ask the sysops to mail you a new password or something? |
| 15:57.46 | ProperPHB | The SIP trunk is doing not so well on residential DSL. :) |
| 15:57.58 | ProperPHB | Perhaps. |
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| 16:00.09 | nightrid3r | ProperPHB: /nickserv help setpass |
| 16:02.30 | nightrid3r | hmmm can't find out how to mail key |
| 16:04.49 | voipmonk | he said sysops |
| 16:05.13 | voipmonk | your age is showing |
| 16:05.15 | voipmonk | :) |
| 16:06.22 | Kobaz | heh |
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| 16:07.17 | jaytee | morning leif |
| 16:07.24 | leifmadsen | morning |
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| 16:25.27 | iq | Hi |
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| 17:03.33 | p3nguin_ | fsb_1: You want DISA for the double 0 and an getting dialtone. (in case no one told you.) |
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| 17:07.40 | *** join/#asterisk bn-7bc (bjarne@2001:16d8:ee6c:0:219:e3ff:fe3b:d34a) |
| 17:10.53 | *** part/#asterisk bn-7bc (bjarne@2001:16d8:ee6c:0:219:e3ff:fe3b:d34a) |
| 17:11.57 | *** join/#asterisk bn-7bc (bjarne@2001:16d8:ee6c:0:219:e3ff:fe3b:d34a) |
| 17:12.25 | *** join/#asterisk d-k-t-3 (~D@112.202.232.46) |
| 17:15.46 | *** join/#asterisk GGD (~GGD@ip72-196-241-104.dc.dc.cox.net) |
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| 17:16.47 | *** join/#asterisk bn-7bc (bjarne@mac.lan.noare-1.holmedal.net) |
| 17:18.15 | ariel_ | hello everyone |
| 17:20.49 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 17:24.51 | *** join/#asterisk smooth_penguin (~smoove@59.95.23.181) |
| 17:24.53 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 17:27.53 | FSB_1 | p3nguin_: Thanks, |
| 17:28.13 | FSB_1 | p3nguin_: But when trying out DISA I get a busytone as soon as I try to dial anything. |
| 17:28.50 | FSB_1 | Do I need WaitExten()? |
| 17:29.07 | tzafrir_laptop | FSB_1, you need WaitExten . You don't really need DISA |
| 17:29.43 | p3nguin_ | fsb_1: Optionally, use pattern matching for your outgoing context, where the extension is something like this: exten => _00XXXXXXXX,1,Dial(SIP/${EXTEN:2}@itsppeer) |
| 17:29.49 | tzafrir_laptop | Just send the user over to a different contenxt where WaitExten is used |
| 17:30.17 | p3nguin_ | You really shouldn't be trying to construct some illogical dialplan mechanism with WaitExten(). |
| 17:31.11 | Kobaz | heh |
| 17:31.35 | p3nguin_ | You don't really need DISA if you use the pattern matching method, but DISA will provide that second "false" dialtone. |
| 17:33.52 | Chainsaw | On the subject of tones, is there no way to set the cadence for the ringing tone? |
| 17:34.09 | Chainsaw | (My FXS gateway does it correctly, Asterisk always seems to produce an american single ring) |
| 17:34.34 | p3nguin_ | indications.conf |
| 17:34.44 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 17:35.16 | carrar | You need to conform to the American ring tone |
| 17:35.24 | carrar | otherwise we will invade your country |
| 17:35.43 | p3nguin_ | We might do it anyway. |
| 17:35.50 | Chainsaw | p3nguin_: That has ringcadence for the physical bell (callee), not the confirmation tone (caller). |
| 17:37.04 | p3nguin_ | ring won't set what you want to set? |
| 17:37.28 | p3nguin_ | for example: ring = 413+438/400,0/200,413+438/400,0/2000 |
| 17:37.45 | Chainsaw | p3nguin_: No, the physical ring is correct but the confirmation tone (*please* note the difference here, it is quite important) is continuous, american-style. |
| 17:38.11 | p3nguin_ | ringcadence is the physical bell, ring is the RINGING SOUND. |
| 17:38.24 | p3nguin_ | Ringing() |
| 17:38.28 | p3nguin_ | PlayTones(ring) |
| 17:38.35 | FSB_1 | Ahhh |
| 17:38.36 | FSB_1 | Fuck it |
| 17:38.45 | FSB_1 | I'll just put my self in a special context. |
| 17:38.57 | carrar | You are special |
| 17:39.07 | carrar | [shortbus] |
| 17:39.18 | FSB_1 | waits for the trolling to commence. |
| 17:40.00 | p3nguin_ | I almost spit tea on the monitor over "special context" being [shortbus]. |
| 17:41.09 | p3nguin_ | chainsaw: Which locale of ringer sound do you want? (I don't know where you are) |
| 17:41.11 | carrar | What kind of tea! |
| 17:41.16 | Chainsaw | p3nguin_: uk |
| 17:41.36 | Chainsaw | p3nguin_: Which is a double short ring. |
| 17:41.47 | Chainsaw | p3nguin_: Not a continuous ring. |
| 17:41.48 | p3nguin_ | Plain ole Lipton black tea (probably with orange pekoe). |
| 17:41.49 | *** join/#asterisk smooth_penguin (~smoove@59.95.58.54) |
| 17:42.48 | p3nguin_ | chainsaw: Did you set country=uk in the general section of indications.conf? |
| 17:42.54 | Chainsaw | p3nguin_: Yes, I did. |
| 17:43.29 | Amorsen | If you're using dahdi, you also need to set it somewhere in its config |
| 17:43.40 | Amorsen | IIRC |
| 17:43.41 | Chainsaw | Amorsen: This is through SIP, but thanks. |
| 17:43.43 | p3nguin_ | ring = 400+450/400,0/200,400+450/400,0/2000 does it for me. |
| 17:44.06 | Chainsaw | p3nguin_: Ah, I think I know what happened here. |
| 17:44.20 | Chainsaw | p3nguin_: I think I have an old indications.conf on a newer Asterisk. (Because I don't see that ring setting anywhere) |
| 17:44.26 | p3nguin_ | oh |
| 17:44.33 | Chainsaw | p3nguin_: That would do it, wouldn't it. Sorry about that. |
| 17:44.48 | *** join/#asterisk rocksfrow (~kyle@pool-71-179-183-143.bltmmd.fios.verizon.net) |
| 17:45.12 | p3nguin_ | I bet it confused people calling your system and getting a US ring tone. |
| 17:46.37 | rocksfrow | does anybody use the rhino failover cards? |
| 17:46.49 | rocksfrow | or anything comparable, |
| 17:47.01 | rocksfrow | or maybe the fonebridges? |
| 17:49.18 | *** join/#asterisk Rajmohan (~raj@122.165.25.171) |
| 17:51.03 | Kobaz | rocksfrow: i use WTI a/b switches |
| 17:51.36 | rocksfrow | ??redundancy |
| 17:51.39 | rocksfrow | notta? lol |
| 17:51.41 | rocksfrow | hrm.. |
| 17:51.46 | p3nguin_ | What is a "notta?" |
| 17:51.57 | rocksfrow | ??redundancy didn't return anything |
| 17:51.57 | rocksfrow | lol |
| 17:52.01 | rocksfrow | notta |
| 17:52.07 | p3nguin_ | I don't know what a notta is. |
| 17:52.09 | rocksfrow | not a thing |
| 17:52.15 | p3nguin_ | hmm |
| 17:52.21 | Kobaz | http://www.urbandictionary.com/define.php?term=notta |
| 17:52.36 | rocksfrow | hey there you go, look at definition 1 |
| 17:52.37 | rocksfrow | ll |
| 17:52.43 | rocksfrow | "not a thing" |
| 17:52.58 | Katty | hi |
| 17:53.02 | rocksfrow | anyway... |
| 17:53.07 | p3nguin_ | Maybe someone got confused with the Spanish word "nada," meaning nothing. |
| 17:53.28 | Katty | bmoraca_work: ping |
| 17:53.34 | rocksfrow | p3nguin_, it is what it is |
| 17:53.57 | Kobaz | http://www.bicomsystems.com/products/C/P/797/255_2797/ who makes that server they use |
| 17:54.18 | Kobaz | looks like NEC is stamped on the bottom right |
| 17:54.24 | smooth_penguin | hi Katty |
| 17:54.29 | Katty | hi smooth_penguin |
| 17:54.47 | p3nguin_ | It does say NEC. |
| 17:55.06 | smooth_penguin | well NEC just assembles |
| 17:55.10 | smooth_penguin | and resells |
| 17:55.15 | rocksfrow | Kobaz, do you use the single card? |
| 17:55.18 | p3nguin_ | http://www.nec.com/global/prod/express/ |
| 17:55.26 | Kobaz | http://www.necam.com/Servers/FT/ |
| 17:55.39 | Kobaz | looks cool |
| 17:55.57 | rocksfrow | Kobaz, http://www.wti.com/AFS-Series/AFS-RJ45-Channel-Card.html ? |
| 17:55.57 | Kobaz | rocksfrow: two servers, one t1 card in each server |
| 17:56.23 | Kobaz | rocksfrow: that's not for a pc |
| 17:56.31 | Kobaz | rocksfrow: that's for their blade-style a/b switches |
| 17:56.39 | rocksfrow | oh so you have the entire encolusure? |
| 17:56.56 | rocksfrow | yeah.. |
| 17:57.03 | Kobaz | hmm |
| 17:57.13 | Kobaz | they make a three a/b that's 1u |
| 17:57.16 | Kobaz | where did it go |
| 17:57.27 | rocksfrow | ah.. |
| 17:57.31 | rocksfrow | yeah that'd be sweet |
| 17:57.37 | rocksfrow | discontinued? |
| 17:57.44 | p3nguin_ | WTF... Banquet TV dinners are barely even a snack, now. How lame. |
| 17:58.01 | Kobaz | hopefully not, it's an amazing product |
| 17:58.04 | Kobaz | telnet interface and everything |
| 17:58.05 | Nugget | telnet is eeeeeeevil! |
| 17:58.06 | Katty | recommends hormeal completes |
| 17:58.13 | Kobaz | auto switchover via voltage feed |
| 17:58.37 | Kobaz | http://www.wti.com/AB-Data-Switches/PLS-345-Physical-Layer-Switch.html |
| 17:58.42 | Kobaz | maybe they stopped making it? |
| 17:58.59 | rocksfrow | kobaz, have you ever checked out the rhinos? |
| 17:59.06 | Kobaz | rocksfrow: i've looked |
| 17:59.12 | Kobaz | i've had such bad experiences with rhino |
| 17:59.17 | Kobaz | that i haven't considered buying one yet |
| 17:59.19 | rocksfrow | oh really? |
| 17:59.26 | Kobaz | they make really shitty t1 cards |
| 17:59.30 | rocksfrow | hrm..i'm also considering the fonebridge2 |
| 17:59.34 | Kobaz | bad firmware |
| 17:59.47 | rocksfrow | what t1 cards do you use? |
| 17:59.49 | Kobaz | spent 5 months trying to get a dual span working |
| 17:59.51 | rocksfrow | i have dig |
| 17:59.53 | Kobaz | sangoma |
| 18:00.36 | Kobaz | after i switched to sangoma cards, about 2 months later i get an email saying they found the problem and it was bad dsp code... "here try this firmware" |
| 18:00.41 | Kobaz | no thanks... sangoma works out of the box |
| 18:01.01 | *** join/#asterisk mog (~mog@c-71-228-185-24.hsd1.al.comcast.net) |
| 18:01.01 | *** mode/#asterisk [+o mog] by ChanServ |
| 18:01.14 | Kobaz | they were good about buying back all my extra rhino hardware though |
| 18:01.28 | Kobaz | had about $15k in rhino boards |
| 18:01.42 | rocksfrow | damn |
| 18:02.04 | Kobaz | they were working fine for a while... and then i bought a new card for a project... ran out of dual spans |
| 18:02.26 | Kobaz | i've had other problems with them too... bad dtmf recognitions, dialtone problems with analog lines, pickup problems with analog lines |
| 18:02.48 | rocksfrow | now this is with their t1 cards right, not the actual failover unit |
| 18:02.57 | Kobaz | and then since it took that long to debug the problem with the dual span, it's like... wtf is gonna happen if a different card has problems |
| 18:03.06 | rocksfrow | right |
| 18:03.08 | Kobaz | yeah, i haven't used the failover unit |
| 18:03.12 | Kobaz | it probably works fine |
| 18:03.30 | Kobaz | so far i've had zero problems with sangoma cards...they are amazing |
| 18:03.35 | rocksfrow | so how does the failover unit interface |
| 18:03.39 | Kobaz | i thought i had a problem with one, but it turned out to be asterisk |
| 18:03.45 | rocksfrow | it looks like the rhino is dependent on the main server |
| 18:03.49 | Kobaz | it's just a simple relay |
| 18:03.59 | Kobaz | it takes power from the host computer |
| 18:04.06 | rocksfrow | oh only power |
| 18:04.06 | rocksfrow | ? |
| 18:04.11 | Kobaz | but if it loses power, it switches to the failover |
| 18:04.24 | rocksfrow | okay that makes sense |
| 18:04.40 | rocksfrow | the rhino also has a 2pin attachment to auto-restart the master |
| 18:04.43 | rocksfrow | on failure..i like that |
| 18:04.49 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 18:06.04 | Kobaz | the rhino board is also much cheaper than the wti switch... which is $700 |
| 18:06.46 | rocksfrow | yyyyyyyyyyeah. |
| 18:06.51 | rocksfrow | damn key stick |
| 18:09.23 | Katty | grumps |
| 18:11.43 | rocksfrow | i'm interested to hear if anybody has success using the fonebridge2 |
| 18:14.25 | *** join/#asterisk correcaminos (~laguilar@201.201.46.106) |
| 18:17.13 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 18:18.30 | Fulg0re | is it possible to do faxes over asterisk/voip too? |
| 18:18.35 | p3nguin_ | yes |
| 18:18.50 | Fulg0re | is it hard to setup? |
| 18:19.01 | eppigy | TRANAJO |
| 18:19.03 | p3nguin_ | It is not recommended to do fax over voice over IP, but it does work most of the time. |
| 18:19.31 | p3nguin_ | It is no more difficult to set up than any other configuration. |
| 18:20.00 | Fulg0re | it is better just to get a analog line? |
| 18:20.11 | p3nguin_ | For reliablity purposes, yes. |
| 18:20.34 | p3nguin_ | If you don't care about that and just want to experiment, go ahead and set up faxing on Asterisk. |
| 18:20.44 | p3nguin_ | Make sure you use ulaw for your codec. |
| 18:21.08 | Fulg0re | i'd have to go to my local telco to get the analog line right? can't cut them out? |
| 18:21.19 | p3nguin_ | or alaw if you aren't in North America. |
| 18:26.59 | *** join/#asterisk benngard (~benngard@90-230-92-67-no148.tbcn.telia.com) |
| 18:27.41 | *** join/#asterisk dobry (~d@95.111.7.95) |
| 18:33.28 | Fulg0re | if i forward a voip number internationally....would there be international charges? |
| 18:35.00 | p3nguin_ | I guess that depends on exactly what you mean. |
| 18:39.40 | Fulg0re | basically its so someone living in a different country can have a local north american number |
| 18:39.53 | Fulg0re | but he wants it on his cell phone on not tied down to a internet line |
| 18:40.01 | p3nguin_ | Yes, but you did not define "forward." |
| 18:40.27 | Fulg0re | oh, well im not sure the best way to do that so i said "forward" :) |
| 18:41.29 | *** join/#asterisk ahall (~dentist@shell.ev6.net) |
| 18:41.43 | p3nguin_ | If you have a US DID on your Asterisk system in the US, and calls to his DID Dial() to his cell phone number in another country, then you will pay for international calling termination rates. |
| 18:42.57 | *** join/#asterisk iq (~iq@unaffiliated/iq) |
| 18:43.02 | p3nguin_ | The only free calling would be by SIP URI. |
| 18:44.24 | Fulg0re | sip uri? |
| 18:45.18 | Fulg0re | and would those call to his did to his cell phone in his country be over voip or regular ptsn lines? |
| 18:45.19 | *** join/#asterisk evil_gordita (~evilgordi@ip70-188-50-186.rn.hr.cox.net) |
| 18:45.33 | p3nguin_ | Depends on how you configure things. |
| 18:46.07 | Fulg0re | what would be a good way to configure it to avoid long distance charges? |
| 18:46.35 | p3nguin_ | SIP URI dialing is like Dial(SIP/1234@your.domain.com) where the call is only SIP and goes directly to a host by the name of your.domain.com. |
| 18:46.58 | Fulg0re | ohhhh |
| 18:48.35 | p3nguin_ | You could configure a VoIP gateway at his house... it would register to the ITSP in the US with his US DID (or to your Asterisk system), and calls to his device would route out his PSTN connection. |
| 18:48.35 | Fulg0re | but that couldnt go to a cell phone? |
| 18:49.00 | p3nguin_ | That configuration would be like if he picks up his home phone and dials his cell phone number. |
| 18:49.46 | Fulg0re | but if he did that and called his local cell would it be a international call anyway? |
| 18:50.09 | p3nguin_ | Is his house in the same location as his cell company? |
| 18:50.28 | Fulg0re | yes |
| 18:50.37 | p3nguin_ | Then it would be a local call. |
| 18:50.42 | Fulg0re | he currently has something like that... |
| 18:51.18 | Fulg0re | he's got a vonage box with a us number...but if he calls his local cell number which is in india, using the vonage box its a international call |
| 18:51.22 | p3nguin_ | Call to the US DID comes to the device. The device calls from his house phone line to his cell phone. |
| 18:51.44 | Fulg0re | ooooh, so the box would have to have another connection to his local telco? |
| 18:51.53 | p3nguin_ | he doesn't have POTS/PSTN phone service? |
| 18:52.20 | Fulg0re | yes he has that too...but they are different lines |
| 18:52.27 | p3nguin_ | The device would use his local telco to call his cell phone. |
| 18:52.55 | p3nguin_ | The same as if he picked up his phone and dialed his cell phone. |
| 18:52.58 | Fulg0re | the vonage box goes over his internet connection, and the pots line goes to his telco, but those 2 are not connected |
| 18:53.13 | p3nguin_ | Forget the Vonage ATA. |
| 18:53.24 | Fulg0re | k |
| 18:53.35 | p3nguin_ | Let's use a Linksys SPA-3102 as an example. |
| 18:54.12 | Fulg0re | k, bought up the product page |
| 18:54.14 | p3nguin_ | Connect the 3102 between his telephone handset and the wall jack. Also connect it to the internet. |
| 18:55.03 | p3nguin_ | Configure the 3102 accordingly (more on that later). Now if he picks up his phone and dials a number locally, he uses his local telco to connect the call. |
| 18:55.33 | p3nguin_ | If someone calls his home phone number, he can also answer the phone when it rings in. |
| 18:56.05 | p3nguin_ | With me so far? |
| 18:56.30 | Fulg0re | yep |
| 18:56.50 | p3nguin_ | Now send a call to his device over the internet. |
| 18:57.01 | p3nguin_ | The 3102 takes the VoIP call. |
| 18:57.43 | p3nguin_ | The 3102 can be configured to ring the handset, as well as dial out to another phone number. |
| 18:58.36 | Fulg0re | and the other phone number can be his local cell |
| 18:58.43 | p3nguin_ | exactly |
| 18:59.12 | *** join/#asterisk ChrisWi (~admin@mx2.wwserver.net) |
| 18:59.17 | p3nguin_ | So it would be the same as if he picked up his home phone handset and dialed his cell number via his local telco. |
| 18:59.40 | Fulg0re | so he would need internet connection, this spa3102, and local phone line for it to all work |
| 18:59.49 | p3nguin_ | yes |
| 19:00.15 | p3nguin_ | He already has two of the three. |
| 19:01.00 | *** join/#asterisk Tim_Toady (~moi@77.49.236.7.dsl.dyn.forthnet.gr) |
| 19:01.28 | Fulg0re | just talked to him, he doesnt have a home phone line |
| 19:01.47 | p3nguin_ | Then you're going to end up paying international termination rates. |
| 19:01.50 | Fulg0re | let me see if he can get that set up though |
| 19:02.06 | Fulg0re | are international termination rates high? |
| 19:02.29 | p3nguin_ | Depends. What are the first 5 digits of his cell phone number? |
| 19:02.51 | p3nguin_ | If I wanted to pick up my phone in the US and call him... the first 5 digits. |
| 19:02.52 | Fulg0re | 01 91 98 |
| 19:02.57 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 19:03.04 | Fulg0re | than it would be 011 91 98 |
| 19:03.10 | p3nguin_ | One moment while I look it up. |
| 19:04.43 | p3nguin_ | India Mobile, 9198 prefix, $0.03030000 USD per minute |
| 19:04.57 | p3nguin_ | That's what it costs me using VoIP.ms services. |
| 19:05.06 | p3nguin_ | 3 cents per minute |
| 19:05.18 | *** join/#asterisk Kamel (klo_028@c-76-123-106-217.hsd1.fl.comcast.net) |
| 19:05.22 | Fulg0re | that doesnt sound bad |
| 19:05.31 | Fulg0re | i think |
| 19:05.47 | p3nguin_ | You can call for a lot of minutes before you justify a local phone line at his house. |
| 19:05.51 | Fulg0re | so it would be the u.s. did cost, plus 3 cents a minute on top of that |
| 19:05.56 | Kamel | i'm looking for a good american SIP provider, primary concern is price. any suggestions? (sorry, i've looked for an extended period of time and am coming up with nothing) |
| 19:06.04 | Kamel | unlimited preferably, only one line required |
| 19:06.47 | p3nguin_ | fulg0re: Correct. You can get an unlimited DID (actually limited to around 3000 minutes) for $6.95 per month. |
| 19:07.21 | p3nguin_ | kamel: Check Flowroute, VoIP.ms, and CallCentric... in that order. |
| 19:07.26 | Kamel | i find it difficult to spend $20/month when i can use skype for $60/yr, using a SIP proxy for skype is very unappealing to me :( |
| 19:07.41 | Kamel | p3nguin_: thanks a ton for your suggestions, means a lot to me |
| 19:07.44 | p3nguin_ | I haven't spend $20 on my phone services in the past 5 months. |
| 19:07.57 | p3nguin_ | together. |
| 19:09.02 | Fulg0re | is there some other configuration required to get the international termination and setup to get the call routed there? |
| 19:09.04 | p3nguin_ | kamel: Do you need a phone number in a specific area code? |
| 19:09.43 | p3nguin_ | fulg0re: Nope. I just have to make sure international calling is enabled in my control panel, otherwise I got a circuit busy error. |
| 19:10.54 | Kamel | p3nguin_: well, i'd prefer to get one in my local area, but it's generally not difficult to come by. if necessary, i could always use a google number to route my calls to have a local area code |
| 19:11.31 | p3nguin_ | kamel: You can get free incoming calls (a free DID) with three different companies that I know of, but you don't get to choose the area code. |
| 19:12.12 | Kamel | p3nguin_: does it still charge per minute? |
| 19:12.31 | p3nguin_ | kamel: IPkall will give you a phone number in WA. IPcomms will give you one in RI. And sipgate does have a small selection, but you aren't likely to find one local to you. |
| 19:12.45 | p3nguin_ | kamel: Those are free DIDs, with free inbound minutes. |
| 19:13.00 | p3nguin_ | kamel: You can't make phone calls outbound with those free services. |
| 19:13.32 | p3nguin_ | kamel: For outbound calling, you need to get termination services from either flowroute, voip.ms, or callcentric (other companies cost more). |
| 19:14.23 | p3nguin_ | kamel: Flowroute charges $0.0098 per minute for outbound calls to US numbers. VoIP.ms changes $0.0105 per minute for calls to US numbers. |
| 19:14.27 | *** join/#asterisk nickaugust (~anonymous@216-160-175-100.hlrn.qwest.net) |
| 19:14.33 | p3nguin_ | charges |
| 19:14.57 | p3nguin_ | CallCentric has an unlimited outbound calling package for around $20 per month. |
| 19:16.08 | p3nguin_ | kamel: Do you primarily take incoming calls or make outgoing calls? |
| 19:17.05 | Kamel | honestly a mix between the 2, but just slightly more outgoing i'd say |
| 19:17.26 | Kamel | i don't use a lot though |
| 19:17.26 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 19:18.13 | raden_work | Kamel, you can use skypw with asterisk if you need cheap :) |
| 19:18.36 | Kamel | i think what i'm going to do is get a local landline with no features at all i think can be had for $13/monthish, then use voip for long distance |
| 19:18.45 | p3nguin_ | I would probably sign up for a free DID phone number with IPcomms, then sign up with Flowroute for outgoing calling. With this configuration, you'll spend under 1 cent per minute for outgoing calls and incoming is free. |
| 19:19.07 | raden_work | Kamel, why not just get a unlimited inbound line for $6 a month VOIP ? |
| 19:19.10 | Kamel | raden_work: i was under the belief that you had to use a skype proxy which required to have a computer running the full skype client for it right? |
| 19:19.32 | raden_work | Kamel, you using asterisk ? |
| 19:19.37 | ChannelZ | no you can use chan_skype - $60 |
| 19:19.51 | raden_work | ChannelZ, thats $60 per channel correct ? |
| 19:19.58 | raden_work | ChannelZ, 1 time fee ? |
| 19:20.03 | ChannelZ | yes |
| 19:20.15 | Kamel | raden_work: not yet, in the design phase right now, currently i only a skype ATA usb box and i'm hating it |
| 19:20.16 | raden_work | ChannelZ, have you been using it ? |
| 19:20.23 | ChannelZ | A little yes |
| 19:20.31 | raden_work | how it been working ? |
| 19:20.39 | raden_work | Kamel, what are your needs ? |
| 19:20.57 | Kamel | raden_work: not well if i'm honest, i hate the skype client for a large amount of reasons |
| 19:21.04 | ChannelZ | Works fine though I don't use Skype Out |
| 19:21.36 | raden_work | ChannelZ, Skype out works great :) |
| 19:21.59 | raden_work | Kamel, asterisk has a module for $60 per channel so you no longer need to use skype client |
| 19:22.11 | p3nguin_ | kamel: In the configuration that I presented, you'll be able to make 1300 minutes of outbound calls for the same price as the phone line you're wanting to buy. |
| 19:22.35 | Kamel | raden_work: i'm currently just looking for a good home setup, but once i get everything set up am considering using it for some business needs, but overall very light even with the business requirements. faxing is important, and i'd love to be able to use 56k dialup as a backup internet service in the event that my ISP is down, but that doesn't really change my phone plans |
| 19:22.50 | *** join/#asterisk KingDavidNYC (~Chris1232@pool-96-224-162-219.nycmny.east.verizon.net) |
| 19:22.52 | Kamel | just that i will probably end up with atleast a most basic incoming POTS phone line |
| 19:22.56 | *** join/#asterisk TimeRider (~steve@78.32.26.1) |
| 19:23.28 | raden_work | Kamel, # of lines , budget ? how many min a month |
| 19:23.57 | raden_work | and you should follow p3nguin_ 's suggestions |
| 19:24.05 | p3nguin_ | Suit yourself. I'm trying to give you the best option for the least price. If you make more than 1300 minutes of outbound calls, then your $13 phone line would be justified. |
| 19:24.06 | raden_work | and a 56k for backup ? whats the point ? |
| 19:24.12 | raden_work | find a local wifi provider cheaper |
| 19:24.44 | Kamel | 2 lines, preferably less than $6/month but probably an optimistic goal (i have no specific limitation on spending, just prefer not spend more), and i would say the minutes even on the busiest month will barely hit 1000 (inc+out combined) |
| 19:24.50 | raden_work | DSL or Cable as main | WIFI for failover asterisk SIP with p3guins suggestions or chanskype if you need cheap |
| 19:25.25 | raden_work | Kamel, I have 14 channels / Lines $22 a month and 4 channels are unlimited inbound |
| 19:25.32 | Kamel | p3nguin_: not arguing, just stating i'm going to probably be getting an additional basic POTS line, and only use it for faxing and dialin |
| 19:25.49 | KingDavidNYC | Hi everybody, has anybody programmed in trixbox?, is it true that I can just write my code on the extensions_override.conf file and trixbox will use my code instead of the one on extensions.con? |
| 19:25.55 | raden_work | Kamel, use email to fax service $5 a month |
| 19:26.16 | raden_work | KingDavidNYC, #trixbox |
| 19:26.32 | KingDavidNYC | raden_work: good idea |
| 19:27.04 | raden_work | Kamel, what you are wanting todo is not difficult |
| 19:29.43 | Kamel | raden_work: yea, i'm aware, just looking for the best/cheapest option, did not know about the skype client for asterisk |
| 19:29.59 | raden_work | its a module not a client |
| 19:30.11 | raden_work | and its $60 per channel |
| 19:30.54 | Kamel | one time fee or annually or? |
| 19:31.22 | p3nguin_ | For $60, I can have a year of regular non-skype calls. |
| 19:33.00 | raden_work | p3nguin_, I was just looking at that me to there rates are actually high compared to my wholesale rates |
| 19:33.06 | ChannelZ | it's a one-time $60 to buy a license for the channel driver, then monthly through Skype at whatever rates |
| 19:33.14 | Kamel | i see, that sounds like what i'd want... i am ok with using skype, i just don't like their pc client so if i could get away from that it sounds great |
| 19:33.16 | raden_work | Kamel, one time buy you still need to pay skype out feee |
| 19:33.27 | Kamel | ChannelZ: ah, i see, that's not bad |
| 19:33.32 | raden_work | Kamel, use callcentric |
| 19:33.38 | raden_work | get the office unlimited plan |
| 19:33.51 | raden_work | $9 a month 3 inbound channels unlimited incoming |
| 19:34.06 | raden_work | or vitelity |
| 19:34.27 | *** join/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
| 19:34.38 | raden_work | Kamel just do pay per call !!! no month fees |
| 19:36.09 | Kamel | lol thanks for your help guys, honestly i'm pretty new to all of this so sorry if i haven't been making much sense to you, i'm still learning |
| 19:36.45 | drmessano | Skype for Asterisk has been decent.. I havent paid for any of their PSTN services, just client <> client |
| 19:37.07 | KingDavidNYC | that #trixbox room is so empty, looks like a funeral |
| 19:37.31 | drmessano | KingDavidNYC, we've been noticing a nasty smell coming from over there for 2 years |
| 19:38.10 | KingDavidNYC | lol, I wish I would keep away from it(trixbox) myself |
| 19:38.25 | drmessano | So do it |
| 19:38.50 | Kamel | i currently use their unlimited in/out and am satisfied with it, i just hate the skype client and my current ata is extremely bad (to make a call, i must dial ##0011__________* where the #'s in the middle is the areacode+number i'd like to call and it's horribly annoying |
| 19:38.51 | KingDavidNYC | cant, the customer has a trixbox box |
| 19:39.12 | Kamel | also breaks many features on the physical phone, like calling a missed call back from caller id |
| 19:39.13 | Kamel | etc |
| 19:41.06 | Kamel | thinking i'm going for a cisco SPA3102 |
| 19:41.32 | drmessano | Do you have an analog line? |
| 19:41.40 | Kamel | i will |
| 19:42.00 | drmessano | Why would you want to do that? |
| 19:42.16 | Kamel | but like i was saying earlier, wont be used heavily for voice i'll be using primarily VOIP |
| 19:42.21 | raden_work | wow 21 people |
| 19:42.23 | raden_work | 31 |
| 19:42.32 | Kamel | to do what? get an analog line? |
| 19:42.36 | drmessano | Yes |
| 19:42.57 | raden_work | drmessano, he makes no sense |
| 19:43.20 | Kamel | lmao |
| 19:43.26 | Kamel | it does make sense |
| 19:43.36 | raden_work | Kamel we saturate our DSL connection with 8 - 10 voice calls all day long no issues so I dont know why you would need a pots line |
| 19:44.21 | Kamel | i want an analog line for a way to connect to the internet during down time, it's just a bare basic pots line, no frills at all, it's only used for data or as a backup if the internet is down |
| 19:44.35 | drmessano | LOL |
| 19:44.48 | drmessano | Where are you located? |
| 19:44.53 | Kamel | in florida |
| 19:45.06 | drmessano | How is dialup internet even a viable backup anymore for anything? |
| 19:45.15 | Kamel | it's better than nothing |
| 19:45.29 | drmessano | Save your money on the pots line and get service from another carrier if you need it that bad |
| 19:45.34 | drmessano | Nobody uses dialup for backup |
| 19:45.40 | Kamel | lol |
| 19:45.42 | ChannelZ | Maybe you just shouldn't have a computer or a phone |
| 19:46.34 | drmessano | By the time you get service from an ITSP and pay for the analog line, you can get a cheap backup line from the other provider around (or one of) |
| 19:46.42 | Kamel | i don't see what's wrong with my requirements, internet service providers go down, if i bundle my dsl service with a basic bare pots line it is basically free anyway |
| 19:47.02 | drmessano | If the ISP goes down why do you think the SAME LINE will work? |
| 19:47.36 | drmessano | Rarely is there a pure data outage on a DSL line.. unless your provider sucks |
| 19:48.16 | Kamel | well, you have a point there, but still if you're getting it for free, i don't see what the disadvantage would be |
| 19:49.15 | drmessano | So you get the line and the service for free? |
| 19:49.40 | Kamel | my local provider is AT&T, if i get dsl 6 meg service with a home phone line, it's $24.95 and they have other promotions etc, if i get it direct, it's $47.95/month |
| 19:50.13 | Kamel | the land line is $13/month for the basic (very) no frills phone line, no caller id, no voicemail, no long distance, etc |
| 19:50.35 | drmessano | and they give you dialup service? |
| 19:50.43 | Kamel | so that's $37.95 with a phone and dsl or $47.95 without |
| 19:50.49 | Kamel | yes, 20 hours per month free |
| 19:51.00 | Kamel | over that it will cost, but i wont be going over 20 hours |
| 19:51.01 | Kamel | lol |
| 19:51.09 | Kamel | unless the ISP blows, then i'd switch back to cable |
| 19:51.21 | drmessano | Do you know what the hell you're doing at all? |
| 19:51.24 | Kamel | but on cable right now and am spending $60/month for just the cable and it's an introductory thing |
| 19:51.48 | Kamel | of course, why do you ask? |
| 19:51.49 | drmessano | $24.95 is an introductory with AT&T |
| 19:51.56 | Kamel | this is also introductory but it's for a whole year |
| 19:52.03 | Kamel | i only have 1 month left on this plan |
| 19:52.12 | ChannelZ | and then after a year he will be back trying to figure out how to get something else cheaper |
| 19:52.16 | raden_work | AT&T has DSL FOR $19.95 |
| 19:52.19 | Kamel | the AT&T one is for 1 year, specifically |
| 19:52.33 | Kamel | ChannelZ: lmao, yea.... |
| 19:52.49 | drmessano | Not sure who your cable is with, but I pay $60 for Comcast Business 6meg service |
| 19:52.53 | Kamel | raden_work: but 768kbit sucks :( |
| 19:53.17 | Kamel | drmessano: it is comcast, maybe there's a better plan but i don't have cable tv in my home and no desire to |
| 19:53.24 | Kamel | maybe without tv service business is better |
| 19:53.25 | ChannelZ | I pay $60 for comcast business for 12mbit/2mbit |
| 19:53.26 | Kamel | i'm on res |
| 19:53.28 | drmessano | Thats without TV |
| 19:53.48 | Kamel | mine is 12/2 right now |
| 19:53.50 | drmessano | ChannelZ, same plan, they havent doubled the speed here yet |
| 19:53.52 | Kamel | but technically it's 6 |
| 19:54.04 | ChannelZ | drmessano: bitches! |
| 19:54.07 | drmessano | I get 20/2 with speedboost |
| 19:54.18 | Kamel | it's 6 with a possibility of 12 depending on how saturated the network is according to what the rep told me |
| 19:54.20 | drmessano | But thats short lived |
| 19:54.56 | Kamel | honestly having a large download isn't as important to me as having good up and down, which is a disadvantage to AT&T over comcast |
| 19:55.40 | Kamel | i was interested in the business 20 meg plan, but the upload was still just 2 megs which is kinda pointless, i'd take a 5/5 line over a 20/2 one any day |
| 19:56.11 | ChannelZ | yeah me too actually |
| 19:56.29 | ChannelZ | though Comcast has docsis 3 out here, they have a 100/10 plan for $200/mo |
| 19:57.08 | p3nguin_ | I've tried to explain to Charter that some people don't give a shit about the download speed when they are asking for business service. They wanted to convince me that 5/1 was "good." |
| 19:57.12 | Kamel | they have docsis 3 here too, but i am unaware of a 100mbit package, they have a 50mbit package i know, but it's still only 5mbit upload @_@ |
| 19:57.40 | Kamel | i mean 5mbit is nice, but for $135/month they quoted me |
| 19:57.42 | Kamel | and res |
| 19:57.50 | Kamel | which means it would have the 200gb limit still |
| 19:58.00 | drmessano | I take that back, I think we did get the speed increased now |
| 19:58.12 | Kamel | err, well 200-250, the rep explained 200gb was the soft cap where they started watching |
| 19:58.14 | ChannelZ | yay! |
| 19:58.22 | drmessano | I ran an entire speedtest at 2.5meg down |
| 19:58.40 | drmessano | Usually its first 1MB at boost, then throttled to 1 |
| 19:58.44 | drmessano | So thats different |
| 19:58.52 | p3nguin_ | curl http://speedtest2.eastlink.ca/superlarge.bin > /dev/null |
| 19:58.59 | *** join/#asterisk ManxPower-work (~EWieling@216.186.151.147) |
| 19:59.11 | ManxPower-work | ~answers |
| 19:59.12 | infobot | rumour has it, answers is Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt |
| 19:59.57 | p3nguin_ | Only getting 2060k today. |
| 20:00.20 | drmessano | I dont have curl.. Guess I will have to boot into Windows and use my GNUWin package :( SAD FACE |
| 20:00.43 | p3nguin_ | wget would be okay, too. |
| 20:01.09 | drmessano | So would an "lol" |
| 20:01.10 | p3nguin_ | You'll just end up with the file and have to delete it, where curl doesn't output a file unless you tell it to. |
| 20:01.37 | Kamel | well, you guys may think i'm crazy for ideas, and you may be right, but thanks for helping me, i learned a lot by our convo. who knows, maybe i'll check into comcast business service and see if they could give me a flat rate at the current speed. i'd be ok with that, comcast is a decent isp, i just don't much care for their monitoring excersizes etc |
| 20:02.06 | Kamel | i'm also very much so the type of person who will switch isp's out of principle, more people should be like that, then comcast wouldn't play their bs games |
| 20:02.38 | drmessano | I use what works.. Principle costs too much |
| 20:02.53 | Katty | wheeeeeeeeeeeeeeeeeeeeeee |
| 20:03.19 | Kamel | not when they end up sending you a bill for overage or somethind stupid like that |
| 20:03.25 | drmessano | I guess when you have time to piss, and don't care what works, that's fine.. But last thing I need is to switch ISPs every 6 months because Betanews says Comcast or AT&T is more evil |
| 20:03.38 | Kamel | lmao |
| 20:03.43 | Kamel | that's true |
| 20:03.52 | TJNII | Problem is you have lots of ISPs, but you still have to pay Comcast or Qwest for the actual wire to the building. |
| 20:04.00 | Kamel | drmessano: very curious, does comcast business block port 80 incoming? |
| 20:04.07 | ChannelZ | no |
| 20:04.12 | drmessano | Comcast BUSINESS |
| 20:04.12 | ChannelZ | or 25 |
| 20:04.16 | ChannelZ | or anything. |
| 20:04.28 | Kamel | ChannelZ: ah, nice, didn't know was always curious |
| 20:04.33 | *** join/#asterisk albertoandrade (~albertoan@201.22.33.3.dynamic.adsl.gvt.net.br) |
| 20:04.36 | drmessano | That means you run a work thingo off of one |
| 20:04.38 | drmessano | Not a home thingo |
| 20:04.45 | ChannelZ | honestly, I'm not sure why people would get comcast home anymore since the business price came down |
| 20:04.59 | drmessano | $59.95 includes gateway |
| 20:05.02 | drmessano | Cant beat it |
| 20:05.03 | Kamel | drmessano: i understand, but most businesses don't use their business connection to host their websites anyway |
| 20:05.17 | Kamel | i wouldn't |
| 20:05.19 | drmessano | Kamel, I guess you never heard of MS Exchange |
| 20:05.19 | ManxPower-work | Because most people are idiots and can't be bothered to learn enough to get the best value. |
| 20:05.20 | ChannelZ | I mean home is cheap if you have cable TV through them too but unless you're just a non-tech web surfing family, it's barely worth it |
| 20:05.22 | drmessano | or Webmail |
| 20:05.32 | carrar | Whats MS Exchange |
| 20:05.39 | carrar | Sounds like a FAIL |
| 20:05.40 | drmessano | Exactly |
| 20:05.57 | ManxPower-work | carrar: It's when you should exchange Microsoft for Linux. |
| 20:06.09 | carrar | haha |
| 20:06.14 | drmessano | Ports 80 and 25 are a drop dead requirement for many businesses |
| 20:06.17 | Kamel | drmessano: i guess, i'm not very business savvy i guess =\ |
| 20:06.24 | ManxPower-work | drmessano: but not for home users. |
| 20:06.39 | carrar | Sendmail, Pine & DaviCal!! |
| 20:06.42 | *** join/#asterisk iq (~iq@unaffiliated/iq) |
| 20:06.45 | ManxPower-work | If more ISPs blocked port 25 the world would be a better place. |
| 20:06.46 | p3nguin_ | If I'm going to pay for BUSINESS service, I'm going to run my BUSINESS stuff on it. |
| 20:06.48 | drmessano | ManxPower-work, except we were discussing Comcast BUSINESS service, but I guess I didnt use enough CAPS |
| 20:07.04 | p3nguin_ | helps drmessano with more caps |
| 20:07.19 | ManxPower-work | drmessano: actually someone asked why someone would get home instead of business, that's what I was respoinding to |
| 20:07.40 | Kamel | and btw i do know what ms exchange and webmail are, just when i think of businesses i think of places that have 5 computers connected to 1 cable line through a horrible router with some idiot they overpay to be the "IT guy" and basically isn't even used as much as many residential lines |
| 20:07.58 | Kamel | perhaps my idea of businesses is incorrect |
| 20:08.14 | Kamel | unless a large business, which wouldn't use comcast as their isp |
| 20:08.16 | drmessano | Not the couple hundred customers I used to support |
| 20:08.27 | drmessano | Lots of SBS out there |
| 20:08.40 | Kamel | shrugs |
| 20:08.54 | Kamel | clearly just not an area i know much about, what businesses use that is |
| 20:09.08 | ManxPower-work | At least you have options. |
| 20:09.32 | Kamel | i know what i use, but i never do things the way most places operate |
| 20:09.42 | ManxPower-work | At me cabin, the options are dialup, HughesNet, or Verizon EVDO/Cellular |
| 20:09.49 | Kamel | for better or worse lol |
| 20:10.00 | drmessano | Kamel, you're a home user.. that doesn't apply |
| 20:10.02 | p3nguin_ | And you chose which one? |
| 20:10.29 | ManxPower-work | p3nguin_: Me? Verizon EVDO |
| 20:10.30 | carrar | Packet over Smoke |
| 20:10.39 | Kamel | drmessano: you're right, but i do a lot of things which most home users dont, hosting sites from my home pc would be one of those things |
| 20:10.40 | carrar | wifi! |
| 20:10.48 | carrar | microwave |
| 20:10.55 | drmessano | Over a DSL line? |
| 20:11.01 | ManxPower-work | carrar: I've investigated them all. |
| 20:11.06 | drmessano | Youre kidding, yah? |
| 20:11.15 | p3nguin_ | Yes, over a Digital Subscribe Line line. |
| 20:11.16 | ManxPower-work | Cheapest I could find was $600/month for T-1 internet. |
| 20:11.35 | carrar | I have a T1 here and two bonded DSL's |
| 20:11.42 | carrar | Me loves T1 |
| 20:11.49 | Kamel | drmessano: me? no, i'm not kidding, but the sites i've hosted have been small and for specific purposes, not commercial or anything like that |
| 20:11.57 | carrar | and I don't care if it's only 1.544 |
| 20:12.11 | carrar | I control both ends |
| 20:12.19 | drmessano | Kamel, so you're a hobbyist then |
| 20:12.21 | p3nguin_ | 1.544 is good enough for a few people at once. |
| 20:12.34 | Kamel | drmessano: yes, that may describe me well |
| 20:13.02 | carrar | plus there is free wifi all around this city |
| 20:13.07 | drmessano | kamel, hobbyists annoy me.. Home user mentality, IBM aspirations |
| 20:13.32 | ManxPower-work | drmessano: don't forget "the budget of a homless person" |
| 20:13.37 | drmessano | HAHAHHA |
| 20:13.42 | Kamel | drmessano: well, i do have ambitions of doing some business, but on a small scale. my wife is a nail tech and would like to start a salon for example, which i may do something like create a simple website from my home connection for |
| 20:14.03 | drmessano | Kamel, or you could just have it hosted for THREE dollars a month |
| 20:14.08 | Kamel | mostly just to avoid paying for hosting services |
| 20:14.39 | drmessano | I should have ducked out of this a long time ago.. Hour of my life I will never get back :( |
| 20:14.46 | p3nguin_ | Who does it for $3/mo? |
| 20:14.48 | Kamel | drmessano: i guess, i was just giving example |
| 20:14.49 | ChannelZ | carrar: how does the bonded dsl work? I'm looking at switching telecom companies for my office because Qwest sucks, and the new company has a 25/2 bonded DSL product |
| 20:15.09 | drmessano | p3nguin_, I see ads in PC World all the time for $3 hosting places |
| 20:15.37 | Chainsaw | carrar: It's basically just two ADSL connections that are glued together on both the ISP end and your end. |
| 20:15.46 | drmessano | GoDaddy gives you free hosting with your domain reg |
| 20:15.49 | Chainsaw | carrar: (So you'll probably have to get a different line delivered/installed for it) |
| 20:15.55 | ManxPower-work | ChannelZ: In my experience "it works mostly well, except when the damn carrier can't keep all the pairs working" |
| 20:16.08 | Chainsaw | s/carrar/ChannelZ/ |
| 20:16.13 | drmessano | Limited, but I think Kamel Toes Nail Salon may not need more than 20GB a month transfer..Just sayin |
| 20:16.35 | Kamel | lmao |
| 20:16.43 | *** join/#asterisk atis_work (~atis_work@193.238.212.171) |
| 20:17.10 | Kamel | except when i host up the latest beta of windows on it and post it on pirate sites |
| 20:17.28 | Kamel | i'm only kidding... |
| 20:18.01 | drmessano | Oh, I am only going to be kidding when I recommend you leave Asterisk for the adults and buy a Skype phone from Wal Mart then |
| 20:18.15 | Kamel | but meh whatever. i like the flexibility of handling my own equipment, there's certainly some advantages to hosting yourself |
| 20:18.27 | drmessano | Is it LOL j/k :) or :) lol J/K ? |
| 20:19.17 | drmessano | Kamel, there's not, really.. running shit at home you can easily offload somewhere else for cheap is a waste of time.. having 5 servers at home and pretending to be Google gets old |
| 20:19.31 | drmessano | There's sunlight out there, really |
| 20:19.48 | *** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr) |
| 20:19.55 | p3nguin_ | I heard that sunlight burns! |
| 20:20.22 | Kamel | tbh i don't really care what you think of my ideas for doing the things i do, it's my decisions to make, i will be the one ultimately reaping the benefits or disaster that comes from them |
| 20:21.48 | drmessano | Ah.. the "I dont give a shit what you think" conversation begins.. Going to find that sunlight now.. CYA'S! |
| 20:22.09 | Kamel | later |
| 20:23.37 | Kamel | well i do care about your input, it's valuable to me. i just don't care about what judgements people pass about what i should do and how i should do it... it just doesn't have any effect and deters from beneficial conversation |
| 20:23.48 | Kamel | on that note, thanks for your help and teaching me |
| 20:24.07 | Kamel | anyway, i'm out too, going to the store |
| 20:29.20 | Katty | wtb shrimp recipe |
| 20:31.01 | nightrid3r | mmmmm shrimp :D |
| 20:31.47 | Katty | indeed |
| 20:31.52 | Katty | but i need a tasty way to prepare it |
| 20:40.35 | nightrid3r | don't ask me, i only eat the stuff |
| 20:46.30 | *** join/#asterisk ttwhy (~tekkno@p4FECFA10.dip.t-dialin.net) |
| 20:48.45 | *** join/#asterisk TimeRider (~steve@78.32.26.1) |
| 20:53.10 | *** join/#asterisk Circlefusion (~circlefus@74-132-116-73.dhcp.insightbb.com) |
| 20:54.32 | Kamel | Katty: i just steam my shrimp, then include it with any normal meal *shrugs* |
| 20:58.17 | p3nguin_ | If you aren't going to Return() at the end, is there any difference in Gosub() and Goto()? |
| 21:01.07 | nightrid3r | p3nguin_: you'll probably run out of stack space and crash |
| 21:01.37 | p3nguin_ | In what condition? |
| 21:01.49 | nightrid3r | gosub without return |
| 21:03.21 | nightrid3r | in the old 8bit day's precompilers used to check on that cos of the stack limits |
| 21:03.54 | nightrid3r | don't know how it works today but probably a similar system |
| 21:04.16 | *** join/#asterisk xmitter (~xmitter@c-24-21-213-242.hsd1.or.comcast.net) |
| 21:05.05 | nightrid3r | best is to avoid goto as much as possible to prevent your code from looking like spagheti |
| 21:05.49 | *** join/#asterisk Liqdfire (~panda@65.34.91.10) |
| 21:06.32 | Liqdfire | I am having an issue with realtime and odbc, anyone able to help ? |
| 21:06.57 | p3nguin_ | Well, I have had to use a Goto because I had a problem with more than one context having a 1 in it. Such as phones included call-return and blacklist contexts, both which had an option 1. So when I called the blacklist and pressed 1, it actually registered 1@call-return. |
| 21:07.15 | p3nguin_ | The solution was to change the includes to Goto()s. |
| 21:07.27 | p3nguin_ | Essentially creating IVRs out of those two contexts. |
| 21:09.43 | nightrid3r | sometimes its impossible to avoid goto's but try to limit them |
| 21:11.19 | Liqdfire | I have asterisk setup to read the configs using realtime over odbc from a SQL2008 server |
| 21:11.41 | Liqdfire | when I connect using a softphone it connects just fine |
| 21:11.43 | p3nguin_ | Macros are being deprecated in favor of Gosub. That was my reason for asking if a Gosub without a Return should instead be a Goto. |
| 21:12.10 | Liqdfire | however when registering my aastra phone, when it tries to update the sip config table it throws a sql exception |
| 21:12.57 | nightrid3r | yes, if you don't use return you have to use goto |
| 21:12.59 | Liqdfire | I noticed it was passing an empty string in for some of the columns that I had set to integers, so I changed them to varchars, but it is still crashing |
| 21:16.56 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 21:17.38 | norrec | does anyone know if you can add in ilbc support without recompiling? |
| 21:18.03 | ManxPower-work | norrec: Why don't you want to recompile? |
| 21:19.07 | norrec | ManxPower-work i installed though the repos =/ |
| 21:19.39 | Liqdfire | is it a production box ? |
| 21:19.53 | ManxPower-work | norrec: You begin to understand why we don't support packages. |
| 21:20.04 | ManxPower-work | norrec: the answer is: No!. |
| 21:20.15 | norrec | yeah, i'm realiseing the mistake of that one |
| 21:20.18 | ManxPower-work | You could rebuild the packages yourself. |
| 21:20.51 | ManxPower-work | just don't install a compiled from source asterisk on top of a packaged installed Asterisk |
| 21:20.57 | norrec | well, i'm thinking about just saving the config dir, and reinstalling from source |
| 21:21.11 | Liqdfire | do it |
| 21:21.25 | Liqdfire | especially if it is not a prod box yet |
| 21:21.25 | norrec | ManxPower-work: yeah, i figured that would be a bad idea |
| 21:21.31 | ManxPower-work | norrec: *nod* If you are going to be installing it on multiple servers, I would actually rebuild the package. |
| 21:22.11 | *** join/#asterisk albasheers (~basheer@188.116.235.226) |
| 21:22.12 | ManxPower-work | You might have a learning curve in package management, but it will serve you well in the future if you're going to be managing multiple servers. |
| 21:22.16 | norrec | well, sounds like thats what i will do then, does asterisk save any configs outside /etc/asterisk? |
| 21:22.37 | *** part/#asterisk albasheers (~basheer@188.116.235.226) |
| 21:22.48 | ManxPower-work | norrec: no, but you also have voicemail messages, etc. |
| 21:23.13 | ManxPower-work | One thing you could try is delete everything from /var/lib/asterisk/modules/* |
| 21:23.34 | norrec | ManxPower-work: yeah, i dont really have any in there, just test shit, but my configs are all setup already |
| 21:23.55 | norrec | ManxPower-work: what does that do? |
| 21:24.04 | ManxPower-work | find a way to "uninstall" the asterisk pakage without deleteing files. Then install Asterisk on top of that. Should work OK as long as you stick to the same major version of Asterisk. |
| 21:24.11 | *** join/#asterisk mnicholson (~mnicholso@nat/digium/x-kxrkcwlerladnloz) |
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| 21:24.40 | ManxPower-work | the most likely issue you would have is with the Asterisk modules |
| 21:24.46 | norrec | ah |
| 21:25.05 | ManxPower-work | If you stick to the same major version everything else should work fine. |
| 21:25.37 | norrec | so, with the source, how would i upgrade between 1.6, and 1.8 when it comes out? |
| 21:25.54 | norrec | or between 1.6.0 and 1.6.2 |
| 21:26.00 | ManxPower-work | norrec: there should be upgrade notes that covers that. |
| 21:26.08 | norrec | ah, alright |
| 21:27.59 | norrec | so for the souce install, does it try to use root or asterisk as the user by default? |
| 21:28.40 | ManxPower-work | norrec: That's a config file thing, so your source install should be the same user as your existing install. |
| 21:29.45 | ManxPower-work | when you install asterisk with "make install" also a "make config" which installs the "config" startup scripts, usually via init.d |
| 21:30.08 | norrec | alright |
| 21:30.10 | ManxPower-work | "make samples" will install the default set of Asterisk config files, overwriting your existing. |
| 21:32.06 | ManxPower-work | the init scripts are another potential point of problem, so reinstalling them is a good idea. |
| 21:32.23 | norrec | alright |
| 21:34.13 | *** join/#asterisk catojo (~catojo@189.24.48.139) |
| 21:34.37 | norrec | so would i do the ilbc install instructions after i install asterisk or at the same time? |
| 21:35.17 | Tim_Toady | before you compile it |
| 21:35.48 | Tim_Toady | you get the source, copy it in codecs/ilbc/ and then build asterisk |
| 21:36.24 | norrec | kk |
| 21:36.42 | tzafrir_laptop | Isn't there a script to download the source? |
| 21:36.50 | tzafrir_laptop | (of the ilbc codec) |
| 21:37.10 | Liqdfire | where is the realtime code for Sip channels? so i can see what it is doing whrn preparing the sql statements for odbc. |
| 21:37.19 | *** join/#asterisk adnc (~numer@unaffiliated/adnc) |
| 21:37.22 | bn-7bc | why didi the devs strip ilbc out of the standart distribution? |
| 21:39.49 | tzafrir_laptop | licensing issues |
| 21:42.45 | bn-7bc | ok |
| 21:43.48 | *** join/#asterisk voipmonk (~shido6@CPE002191f85581-CM001692568382.cpe.net.cable.rogers.com) |
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| 21:55.16 | Liqdfire | [Feb 21 16:55:07] WARNING[3699]: res_odbc.c:628 ast_odbc_prepare_and_execute: SQL Execute error -1! Attempting a reconnect... |
| 21:55.16 | Liqdfire | *** glibc detected *** asterisk: corrupted double-linked list: 0x00002aaaac4c09f0 *** |
| 21:55.35 | Liqdfire | that is what I am getting when any sip client tries to register |
| 21:57.08 | norrec | so, for ilbc, i wanted to have some of my peers use ilbc but my sip trunks for outbound/inbound are g711u, will asterisk auto convert, or what do i need to set up to convert between the two? |
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| 21:58.44 | voipmonk | asterisk will convert them for you norrec |
| 21:59.15 | norrec | oh good, thanks =D |
| 22:01.38 | *** join/#asterisk MindTheGap (~MindTheGa@189.59.199.118) |
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| 22:35.57 | norrec | is there a way to get more detailed information about a channel, like the codec being used and bandwidth info and such? |
| 22:36.04 | skrite | hey all, what is the CLI command that lets me see what channels there are available ( not the core show channels ) i need to know if my card is set up right |
| 22:36.29 | skrite | wait sorry, just found the help command :) |
| 22:38.00 | *** join/#asterisk etnos (~etnos@c-75-74-66-161.hsd1.fl.comcast.net) |
| 22:39.00 | p3nguin_ | core show channeltypes |
| 22:39.01 | *** join/#asterisk vk2dgy (~rossw@ali-syd-3.albury.net.au) |
| 22:39.49 | skrite | p3nguin_, hello again, you were the one helping me Friday. i did the show channeltypes and dahdi did not show up, should it have? |
| 22:39.59 | vk2dgy | hi good folks. Anyone here using Future-Nine as a SIP provider with Asterisk 1.4.18? |
| 22:42.12 | *** join/#asterisk mykhyggz (~col@evolone.org) |
| 22:43.30 | p3nguin_ | skrite: http://pastebin.com/d706b8a08 |
| 22:43.52 | p3nguin_ | skrite: If you have the dahdi channel driver loaded, it should show up. |
| 22:47.00 | *** join/#asterisk capitan (~captain@76.91.206.32) |
| 22:47.13 | skrite | thanks p3nguin_ here is what i get http://pastebin.com/d1b7177b2 the correct drivers seem loaded when i do an lsmod |
| 22:49.02 | p3nguin_ | skrite: kernel modules are not the same as asterisk modules. Asterisk modules are the channel drivers. |
| 22:49.43 | p3nguin_ | module load chan_dahdi |
| 22:49.58 | p3nguin_ | That should load the dahdi channel driver if there is no problem. |
| 22:51.58 | skrite | damn missing the /usr/lib/asterisk/modules/chan_dahdi.so |
| 22:53.21 | ManxPower-work | skrite: damn install dahdi before Asterisk |
| 22:53.38 | skrite | i did |
| 22:53.47 | skrite | but i think i did screw some things up |
| 22:53.56 | skrite | will check some apt info |
| 22:54.41 | ManxPower-work | Um, how do you screw up "make install"? |
| 22:55.23 | ManxPower-work | apt info? |
| 22:55.26 | nickaugust | [TK]D-Fender: ping! |
| 22:55.55 | ManxPower-work | You're wasting our time asking about packaged Asterisks? |
| 22:59.08 | skrite | ManxPower-work, wasting your time? sorry, channel did not look too busy. i have done a couple of installs not only of asterisk, but was trying to replace zaptel, think that is where it went wrong |
| 22:59.34 | skrite | still troubleshooting though |
| 23:01.35 | *** part/#asterisk pokui (~pokui@mbp-pjo.imul.com) |
| 23:02.53 | ManxPower-work | uninstall the packages, install zaptel and asterisk from source. |
| 23:04.44 | skrite | ManxPower-work, gotcha, thanks |
| 23:05.05 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-145.cablep.bezeqint.net) |
| 23:07.39 | *** join/#asterisk nickaugust (~anonymous@216-160-175-100.hlrn.qwest.net) |
| 23:35.32 | vk2dgy | so, do I presume nobody here uses future-nine for their sip provider? Does their reputation precede them or something? |
| 23:36.26 | vk2dgy | my drama is that I just can't seem to get my asterisk 1.4.18 to talk to them "properly" - it registers, but whenever I try to place a call, I get: |
| 23:36.31 | vk2dgy | sip_call: No audio format found to offer. Cancelling call to (xxxxx) |
| 23:40.25 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 23:42.21 | *** join/#asterisk angryuser_laptop (~angryuser@90-156-167-83.reverse.alphalink.fr) |
| 23:42.46 | voipmonk | what codecs do u have set for that provider vk2dgy ? |
| 23:45.06 | vk2dgy | I have the following (commented one or the other for testing) |
| 23:45.23 | vk2dgy | disallow=all |
| 23:45.24 | vk2dgy | ;allow=g729 |
| 23:45.28 | vk2dgy | allow=ulaw |
| 23:45.33 | ManxPower-work | you would not want to allow g729 |
| 23:45.44 | vk2dgy | so when I'm trying to use g729, I uncomment it and comment ulaw, and vice versa. |
| 23:45.59 | ManxPower-work | try g729 AFTER you get it working |
| 23:46.12 | vk2dgy | Manx- why not? They specifically state that G729 is their preferred codec. |
| 23:46.15 | skrite | ok, built the dahdi from source, and can modprobe dahdi and it is ok, but when i modprobe wctdm24xxp, i am getting this... http://pastie.org/836017 |
| 23:46.21 | vk2dgy | but I can't get it going with 711u either :( |
| 23:46.29 | *** join/#asterisk tzafrir (~tzafrir@212.179.75.202) |
| 23:48.27 | vk2dgy | "sip show peer (them)" confirms the codec too btw: |
| 23:48.34 | vk2dgy | <PROTECTED> |
| 23:48.35 | vk2dgy | <PROTECTED> |
| 23:52.23 | ManxPower-work | vk2dgy: because I doubt you know enough to make g729 passthru work at this point. |
| 23:52.44 | ManxPower-work | vk2dgy: do a sip debug, pastebin the results |
| 23:52.46 | ManxPower-work | ~pb |
| 23:52.46 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
| 23:53.16 | ManxPower-work | skrite: "service dahdi start". don't modprobe them manually |
| 23:53.29 | skrite | ok, thanks |
| 23:54.18 | adnc | hello, for matching local calls like 123456 would an entry like _N. be enough or do i need to consider anything else? |
| 23:54.32 | ManxPower-work | well N won't match 1 |
| 23:54.36 | vk2dgy | Manxpower - it doesn't work with g711u either, so it's not specifically a g729 issue. I had it all working with broadvoice, it's only when I changed to future-nine that the wheels have fallen off - hence my question about them specifically. |
| 23:54.42 | adnc | maxagaz, ohh |
| 23:54.50 | adnc | that is bad, so _X. is necessary? |
| 23:54.53 | ManxPower-work | vk2dgy: I'm wiating for the pastebin. |
| 23:55.16 | ManxPower-work | adnc: What are you trying to match? |
| 23:55.50 | adnc | manxPower-work i would like to match to local numbers and route them through a particular provider |
| 23:56.10 | adnc | local numbers in germany are from 4 to ... |
| 23:56.13 | vk2dgy | sip debug will flood you with stuff - there are a bunch of other trunks doing stuff at the moment. Will a debug of that trunk only help? |
| 23:56.29 | ManxPower-work | then just do a debug on that peer. |
| 23:56.31 | ManxPower-work | ~trunk |
| 23:56.32 | infobot | extra, extra, read all about it, trunk is a word with varying definitions. In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment."; in Ethernet a trunk carries more than one 802.1q VLAN. There is no such thing as a "SIP Trunk" -- Don't use the term. |
| 23:58.31 | adnc | ManxPower-work so is _X. enough? |
| 23:59.23 | *** join/#asterisk SaiSoma (~SaiSoma@adsl-074-167-136-030.sip.mob.bellsouth.net) |