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| 02:01.10 | nny | whats the command to check a global variable value from cli? |
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| 02:48.54 | prgmrchris | lol |
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| 06:18.35 | samuelsapps | hi guys, how we configure flash button in phone so it can be using with asterisk |
| 06:19.23 | [TK]D-Fender | What "phone"? What "flash button"? |
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| 06:29.16 | carrar | Click GO then start Phone and set Flash |
| 06:32.51 | [TK]D-Fender | Well it can't be an iPhone.... because Apple doesn't support Flash :p |
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| 06:44.16 | atan | lol. |
| 06:44.39 | atan | For the amount flash crashes my browser I think it was a smooth move on their part not to support it. |
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| 06:55.11 | samuelsapps | this button for transfer (if not my mistake) commonly exist in standard analog phone |
| 06:56.23 | WIMPy | That was the answer to the 2nd question. |
| 06:59.24 | [TK]D-Fender | samuelsapps: We were asking what you had. Not some hypothetical device someonelse else might have |
| 06:59.31 | [TK]D-Fender | someone else* |
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| 07:13.04 | ChannelZ | _is_ it an analog phone? |
| 07:15.29 | WIMPy | Wheere else do you find a flash? |
| 07:15.45 | WIMPy | Assuming it's a phone in the first place. |
| 07:16.12 | samuelsapps | sorry my english bad it is an analog phone (panasonic), from the reference i found in the net, for transfer just simply push flash key |
| 07:16.33 | samuelsapps | maybe it's flash key not button sorry my english bad |
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| 07:17.30 | WIMPy | Do you want to configure your phone or dahdi or some ATA? |
| 07:17.41 | [TK]D-Fender | samuelsapps: Asterisk is sofware... and you're talking about what you now claim to be an analog phone. You can't plug an analog phone into a piece of software.... |
| 07:18.15 | [TK]D-Fender | samuelsapps: Perhaps there is a piece of very important hardware in between you should be telling us about. |
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| 07:23.17 | samuelsapps | sorry guys don't be upset ok, (hope my English good), currently i'm trying to configuring transfer between extension, of course this can be done by configuring some extension using features.conf, in the some references from net this can be done using flash key/button (not flash in browser) |
| 07:23.31 | [TK]D-Fender | samuelsapps: Still not an answer |
| 07:23.43 | [TK]D-Fender | samuelsapps: You can't plug an analog phone into software. |
| 07:23.57 | [TK]D-Fender | samuelsapps: So clearly it is plugged into something else. |
| 07:24.04 | [TK]D-Fender | samuelsapps: And we're waiting to hear what it is. |
| 07:24.16 | WIMPy | samuelsapps: Your english is not a problem, but the missing information. |
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| 07:25.06 | samuelsapps | yes you're right, to connect this analog phone i already using voip gateway grandstream gxw 4108 |
| 07:27.04 | samuelsapps | but the problem is I want to configure transfer between extension and from PSTN |
| 07:28.30 | [TK]D-Fender | samuelsapps: Your gateway probably already uses "flash" for transfers |
| 07:28.35 | WIMPy | If you can call the destination, the Asterisk part is ok. |
| 07:28.42 | [TK]D-Fender | samuelsapps: Read its manual <=- |
| 07:28.53 | WIMPy | The transfer has to be done byt the Grangstream. |
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| 07:30.17 | WIMPy | Make sure the flash duration of the phone and the Grandstream match. |
| 07:32.02 | samuelsapps | the problem is when i press the flash key the phone is disconnected |
| 07:33.05 | WIMPy | Then the flash duration of the phone is probably too long. Configure them to match. |
| 07:33.49 | [TK]D-Fender | samuelsapps: Read your gatway's manual |
| 07:34.08 | [TK]D-Fender | samuelsapps: This is not an Asterisk issue |
| 07:35.58 | samuelsapps | hmmmm i see thanks lot for guidance friends |
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| 07:40.51 | [TK]D-Fender | And on that note... it's bed-time. |
| 07:40.53 | [TK]D-Fender | later all |
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| 08:32.00 | IsUp | morning |
| 08:32.46 | WIMPy | Hi IsUp |
| 08:35.40 | IsUp | howdy |
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| 09:43.37 | domi | is it possible to use templates for sip.conf when using realtime static? |
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| 10:15.59 | vfabi | Hi All, is it possible normally (without any issues) transcode from iLBC to g711a ? |
| 10:16.36 | WIMPy | Sure |
| 10:17.06 | vfabi | any ilbc_20\ilbc_30 ? |
| 10:17.39 | WIMPy | Don't know what the lib supports. |
| 10:18.05 | WIMPy | You need to get that seperately anyway. |
| 10:18.17 | vfabi | doc says only about ilbc_30 |
| 10:19.11 | vfabi | hmmm, let ask google ) |
| 10:20.35 | WIMPy | The website tells about 20 and 30 ms packets. |
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| 11:11.53 | atan | I don't suppose anyone here has something setup to allow you to use a cellular device to tx/rx calls? |
| 11:12.03 | atan | An android device even =\ |
| 11:12.15 | WIMPy | I guess many have. |
| 11:12.32 | WIMPy | You can't use adroid for that. |
| 11:12.53 | atan | WIMPy, just to be clear I do mean use the device to send the calls not so much act as a client. Perhaps I'm saying it wrong though, who knows |
| 11:14.20 | WIMPy | Not sure. |
| 11:14.44 | WIMPy | If you say not as client, I guess you want to abuse it as gateway? |
| 11:15.10 | atan | Abuse might be a strong word for it, but yes |
| 11:15.22 | WIMPy | Ok, that as I said. |
| 11:15.48 | kaldemar | atan: see chan_mobile and chan_datacard |
| 11:15.51 | WIMPy | You cannot access the voice data from android. |
| 11:16.34 | kaldemar | WIMPy: do you happen to know if that's the case with all versions of android? |
| 11:16.36 | WIMPy | Wasn't chan_datacard called can_stick now or something? |
| 11:16.51 | WIMPy | As far as I know: Yes. |
| 11:17.01 | atan | So to take advantage of this I presume I would need some device that would accept my SIP card and connect up to Asterisk then? |
| 11:17.32 | WIMPy | Yes. Like kaldemar said. |
| 11:18.15 | kaldemar | i don't know what a SIP card is though. |
| 11:18.25 | WIMPy | You can use a phone via BT or some via serial or you can use a data stick or a sip or an ISDN gatewaw. Lots of options. |
| 11:18.26 | atan | SIM, sorry |
| 11:18.29 | WIMPy | SIM? |
| 11:18.53 | atan | Well then... http://www.telecom-cafe.com/forum/attachments/month_1103/1103231436e326d16c50cafa4b.jpg I assume by the looks of this one needs a sticker thinger for each mobile number they plan to use then |
| 11:18.58 | kaldemar | atan: with chan_mobile, you can use a supported cellular phone to connect with bluetooth. |
| 11:19.03 | WIMPy | There are even POTS gateways. But that seems like a bad choice. |
| 11:19.21 | kaldemar | atan: chan_datacard is for 3G USB modems |
| 11:19.46 | WIMPy | Nice setup :-) |
| 11:21.02 | WIMPy | Much nicer than those SIM changers. |
| 11:24.52 | coppice | I wonder how much all those closely spaced radios desensitise each other |
| 11:26.25 | WIMPy | Do you think it might make a real difference? |
| 11:26.54 | IsUp | WIMPy: how can i reload 'manager' in 1.4? any idea? |
| 11:27.10 | IsUp | WIMPy: ive added an user for test |
| 11:27.42 | WIMPy | Currently it's 'manager reload'. |
| 11:27.44 | IsUp | WIMPy: ah okay 'reload manager' did it :) |
| 11:28.20 | coppice | when I used a bunch of GSM modems in the same room 10 years ago, we had to spread them out for good results |
| 11:29.02 | WIMPy | But there's nothing special with a room full of phones. |
| 11:29.31 | WIMPy | You will have at least as many phones as people. |
| 11:29.37 | IsUp | PRI GSM gateways are ok, its stable at least. |
| 11:29.45 | coppice | many phones screw each other up when you place them back to back |
| 11:30.13 | IsUp | i have 3x 2N StarGate gateways. running in elevator room. and 3 yagi antennas |
| 11:31.11 | WIMPy | Might have been cheaper for your operator to offer you the E1 on copper *lol* |
| 11:31.27 | WIMPy | How many of them do you use simultaneousely max? |
| 11:32.14 | IsUp | its almost full capacity between 18:00-03:00 hours |
| 11:32.29 | IsUp | our agent outbound calls are going over GSM gateways |
| 11:33.16 | IsUp | adult chat lines. so on weekends, we are reaching our max channels. im planning to add another gateway |
| 11:33.22 | WIMPy | You musthave a whole lot of BTSs for yourself then. |
| 11:33.59 | IsUp | yeah in this area we have enough BTS, and good reception |
| 11:35.53 | WIMPy | So that must be at least 6 channels at HR. |
| 11:40.15 | IsUp | our physical location is near to GSM provider's building. its great advantage |
| 11:40.34 | WIMPy | Is it some sort of unofficial use? |
| 11:40.50 | WIMPy | Otherwise I'd certainly ask them for a line. |
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| 11:40.59 | WIMPy | Would be cheaper for both sides. |
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| 11:44.58 | IsUp | it's official use. they are providing SIM cards for us. yeah E1 solution is better but we dont have agreement yet |
| 11:45.51 | IsUp | we are working with other GSM/Satellite providers. we are running SS7 with them |
| 11:45.59 | WIMPy | IIRC the stargate was some 28 grand when I looked some years backs. How much are they now? |
| 11:46.17 | coppice | how can I get a Nexus S running Android 4.0.3 to install 2.3.4? They've reformatted so much data, like the address book, that assume things will go horribly wrong if I try it |
| 11:46.51 | IsUp | it's around 12.000€ |
| 11:47.15 | WIMPy | That's a lot less then. |
| 11:47.31 | coppice | wouldn't getting a fibre have been less than 12k? |
| 11:47.32 | IsUp | yeah but its second hand :p |
| 11:47.39 | WIMPy | Ah, ok |
| 11:48.11 | WIMPy | He, was it you who had a shredded firmware some time ago? |
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| 11:55.25 | IsUp | me? :p |
| 11:55.39 | WIMPy | i guess that makes it a no. |
| 11:56.42 | IsUp | no it wasnt me, what firmware? :> |
| 11:57.14 | WIMPy | There has been someone here a while back with a Stargate, but shredded firmware. |
| 11:57.48 | IsUp | ah ok |
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| 12:24.20 | aways | Ola |
| 12:24.20 | aways | :D |
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| 13:20.55 | atan | Anyone know if the GigaSet One product from Siemens could be used in some way to act as a gateway? |
| 13:23.11 | IsUp | Network compatibility: PSTN | VoIP ATA | PBX : yes | yes | yes |
| 13:23.33 | IsUp | i dont know this device but whats your goal? |
| 13:24.24 | atan | IsUp to get a cellular device connected up to Asterisk remotely |
| 13:27.05 | IsUp | atan: so wheres the device? did you test anything? |
| 13:28.20 | atan | I haven't tried anything with it just yet :-) poking around for options |
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| 13:30.12 | IsUp | okay |
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| 13:49.45 | samuelsapps | hi all, it's possible dial dial number in asterisk cli |
| 13:50.44 | IsUp | samuelsapps: it's 'dial' or 'core dial' i cant remember but yes its possible |
| 13:52.38 | kaldemar | "console dial" |
| 13:52.59 | kaldemar | but only if you have chan_oss, chan_alsa or chan_console |
| 13:53.23 | kaldemar | "channel originate" may also be useful. |
| 13:53.54 | samuelsapps | well i'm using voip gateway i'll guess it cannot be done |
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| 14:05.32 | kaldemar | samuelsapps: what exactly? |
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| 14:20.58 | samuelsapps | my voipgateway is gxw4108 |
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| 14:24.27 | cusco | hi |
| 14:24.56 | cusco | I have a gtalk account in asterisk, and for some reason if I terminate asterisk the gtalk contact is still online |
| 14:26.02 | cusco | I entered gmail from 2 locations with that account and clicked in signiout all other locations |
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| 14:26.26 | cusco | thing is asterisk does not recieve calls made to that gtalk account |
| 14:29.00 | kaldemar | samuelsapps: and what do you want done? |
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| 15:53.23 | jaytee | Qwell, o/ |
| 15:54.05 | Qwell | ohai |
| 15:54.37 | jaytee | did ya have a nice Christmas? |
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| 15:55.39 | Qwell | meh |
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| 16:03.44 | wcselby | o/ |
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| 16:15.10 | wcselby | o/ [TK]D-Fender |
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| 17:02.24 | tdonahue | morning all |
| 17:02.41 | WIMPy | Good evening |
| 17:03.00 | tdonahue | i was looking for the sipsak code and noticed that sipsak.org seems to be gone... Does anyone know if someone has picked up maintaining the code? |
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| 17:28.16 | keith4 | odd. I can originate a call over PRI from the CLI, but trying the same thing via Dial() in the dialplan fails. I think I'm losing my mind |
| 17:29.49 | tdonahue | keith4, what does your Dial command look like? |
| 17:31.52 | keith4 | I have it whittled down to a 2-line test case. set the CID, then "Dial(DAHDI/g1/18007464726,30)", for example |
| 17:33.22 | tdonahue | what's happening when you dial? |
| 17:33.42 | tdonahue | are you seeing any errors in the logs? |
| 17:33.52 | keith4 | yah. verbose output with PRI debug on: http://pastebin.com/L6Q36Tww |
| 17:34.27 | keith4 | yet, I can originate a call to the same number via CLI... so confused |
| 17:36.13 | tdonahue | is it possible it doesn't like the CID that you are setting? |
| 17:36.48 | keith4 | excellent point. yes, I suppose it's entirely possible |
| 17:36.55 | tdonahue | -- Channel 0/1, span 1 got hangup request, cause 127 |
| 17:37.14 | tdonahue | that is the error that your PRI provider is sending for the call termination |
| 17:38.03 | keith4 | right. as far as I can tell, 127 is somewhat generic, though. ("Internetworking, unspecified", according to the wikipedia entrry for Q.931) |
| 17:38.59 | tdonahue | yeah, from what I understand it is generally used as a "Catch all" type of error for times when there isn't a standard code to send |
| 17:39.23 | keith4 | is this still a valid way to set CID? 'Set(CALLERID(all)="Keith Four" <8001234567>)' [but with real name/number, obviously] |
| 17:40.27 | Qwell | keith4: No quotes |
| 17:41.04 | keith4 | so... =Keith Four <8001234567> |
| 17:41.13 | keith4 | ? |
| 17:41.38 | tdonahue | I usually only set the number when dealing with PRIs |
| 17:41.58 | tdonahue | CALLERID(num)= |
| 17:42.14 | *** join/#asterisk ectod (~ectod@ectod.net) |
| 17:42.27 | keith4 | k, that might be it. I'll give it a shot |
| 17:42.42 | ectod | hi, i'm new to asterisk and i'm using it in conjunction with the SIP proxy's of my ISP which is Free ( french ISP ) |
| 17:42.43 | tdonahue | because the name string is going to get ignored by your carrier anyway |
| 17:42.53 | ectod | the thing is i can't receive incoming calls, i get a weird error |
| 17:43.05 | Qwell | tdonahue: Not necessarily. |
| 17:43.15 | *** join/#asterisk gravin (~gravin@17.34.49.60.brf01-home.tm.net.my) |
| 17:44.58 | ectod | it looks like when asterisk receives a call, it tries to identify it with the SIP proxy of my ISP, freephonie.net |
| 17:45.03 | ectod | and that makes no sense :( |
| 17:45.05 | keith4 | tdonahue: doesn't make a difference, it seems |
| 17:46.34 | ectod | anyone to help me understand this ? |
| 17:46.37 | Qwell | keith4: Did the telco that you're dialing out from give you the number that you're setting it to? |
| 17:47.05 | keith4 | Qwell: yes. this all worked last week, before I migrated this box from 1.4 to 1.6 finally ;-) |
| 17:47.17 | Qwell | Why even bother with 1.6? |
| 17:47.28 | Qwell | Both are now unsupported. |
| 17:47.55 | keith4 | i know. stuck with debian squeeze for the time being, though |
| 17:50.11 | tdonahue | keith4, have you tried calling your PRI provider' |
| 17:50.15 | tdonahue | s tech support? |
| 17:50.44 | keith4 | tdonahue: not yet. wanted to (try to) make sure it's not my fault, first |
| 17:52.27 | tdonahue | keith4, you can always block the caller id and to make sure it isn't a problem with the way its being set... Set(CALLERID(num)=) |
| 17:53.46 | keith4 | ha! that did it |
| 17:54.21 | keith4 | tdonahue: thank you! i was about to start pulling hair out |
| 17:54.50 | tdonahue | keith4, your welcome |
| 17:55.28 | ectod | anyone knows why asterisk wants to authenticate incoming calls with the carrier's Sip proxys ? and fails and drop the call to normal adsl phone line ? |
| 17:56.04 | keith4 | tdonahue: time to call the telco, i guess |
| 17:56.27 | tdonahue | keith4, have you tried sending the CID out with 11 digits? |
| 17:56.46 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 17:57.41 | keith4 | tdonahue: yah |
| 17:59.19 | *** join/#asterisk kikohnl (~kotis@72.253.138.39) |
| 18:00.16 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 18:09.32 | WIMPy | Is that the famous bad default for prilocaldialplan thing again? |
| 18:09.44 | WIMPy | keith4: Have you set that parameter? |
| 18:10.42 | keith4 | WIMPy: I've futzed with that and pridialplan quite a bit. currently it's back to default |
| 18:11.27 | WIMPy | Use "unknown" for both, unless you know what you're doing. |
| 18:12.01 | keith4 | so the comments in the sample file that suggest 'national' is a sane default are misleading? |
| 18:12.20 | WIMPy | Although the cause suggests that you hit a network that doesn;t know about these. |
| 18:12.34 | WIMPy | Definitely. |
| 18:14.42 | keith4 | setting both of those to 'unknown' doesn't seem to make a difference. sending 10- or 11- digit CALLERID(num) causes telco to reject |
| 18:15.03 | WIMPy | Someone else. |
| 18:15.24 | *** join/#asterisk anonymouz666 (~anonymouz@189.25.37.248) |
| 18:15.24 | keith4 | ah, wait. didn't blank the CALLERID(name) that time. hold on |
| 18:16.12 | WIMPy | Yes, that is also known to cause issues, but that would probably cause the call to be rejected immediately. |
| 18:16.51 | keith4 | correction: setting prilocaldialplan to unknown causes telco to accept call with 10-digit CID if the name is blank |
| 18:17.10 | WIMPy | Good |
| 18:17.34 | keith4 | indeed |
| 18:17.38 | keith4 | thank you |
| 18:17.54 | WIMPy | Unfortunatly I don't know if there's a parameter to switch that off. |
| 18:18.42 | WIMPy | But that Display IE is a protocol error. |
| 18:19.49 | WIMPy | I guess I should add those two to the usual suspects. |
| 18:23.55 | WIMPy | Is there any switch that will accept the caller names that way? |
| 18:24.28 | WIMPy | I mean, is that a very special feature or ONLY a bug? |
| 18:25.36 | *** join/#asterisk m0f0x (~elmofo@187.66.50.171) |
| 18:26.17 | keith4 | now that I think about it, my itsp is the same way. perhaps they have the same problem with their upstream switching? |
| 18:28.01 | keith4 | WIMPy: this is exactly the issue, yes? https://issues.asterisk.org/jira/browse/ASTERISK-18800 |
| 18:29.07 | *** join/#asterisk vader-- (vader@c-68-83-57-218.hsd1.nj.comcast.net) |
| 18:29.28 | WIMPy | That's one half of it, yes. |
| 18:29.50 | *** join/#asterisk eja (~user@173.219.119.226) |
| 18:29.53 | WIMPy | And according to the document I have, it's not undefined, but not allowed. |
| 18:30.19 | WIMPy | But at lest you find a parameter there to suppress it. |
| 18:30.28 | keith4 | I do see a "DISCONNECT (69)" come back, in pri debug output |
| 18:31.11 | WIMPy | {0x45, "Requested facility not implemented"}, |
| 18:31.22 | keith4 | right |
| 18:31.26 | WIMPy | I guess you could find other words for it. |
| 18:31.39 | eja | hello... what does this mean: ${EXTEN:-7:7} does that take the last 7 digits of $EXTEN? |
| 18:31.49 | keith4 | I wonder why this worked, for years, on 1.4 then |
| 18:32.04 | WIMPy | It had other defaults. |
| 18:32.16 | WIMPy | And that Display thing is pretty new, AFAIK. |
| 18:34.06 | keith4 | ah |
| 18:34.48 | WIMPy | It is good in NT mode, however. |
| 18:35.01 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 18:35.46 | ChannelZ | eja: yes |
| 18:37.02 | eja | channelz: thanks! so ${EXTEN:0} would just grab the first digit? |
| 18:38.21 | WIMPy | That would be :0:1. |
| 18:40.18 | eja | so what would ${EXTEN:0} do? i'm documenting some phone systems we recently acquired that are using asterisk. exten => _2XX,1,Dial(${TRUNK_STJ}/${EXTEN:${TRUNKMSD_LD}},30,r) |
| 18:40.47 | eja | i would think that should be ${TRUNK_STJ}/${EXTEN} |
| 18:40.54 | eja | $TRUNKMSD_LD = 0 |
| 18:41.07 | WIMPy | :0 does nothing. I.e. it returns the whole thing. |
| 18:41.25 | eja | ohh ok. so it is sending the XX then |
| 18:42.12 | WIMPy | If it is 0, it will send 2XX, if it is 1, XX. |
| 18:42.54 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 18:43.16 | [TK]D-Fender | ${EXTEN:0} = ${EXTEN} = should have just done the latter |
| 18:46.27 | ChannelZ | I would only assume TRUNKMSD_LD gets set to 1 or something in other cases to strip off leading 1s or whatever |
| 18:50.22 | eja | so would it send only the XX portion or the full 2XX? |
| 18:50.38 | *** join/#asterisk kamikazemicrowav (~kamikazem@seraph.contegix.com) |
| 18:51.58 | ChannelZ | if it were 1, yes |
| 18:52.14 | WIMPy | If it is 0, it will send 2XX, if it is 1, XX. |
| 18:52.31 | ChannelZ | ${EXTEN:start_digit:num_of_digits} |
| 18:52.50 | eja | ok gotcha. in the cisco world it sends only the matched portion iirc unless u tell it otherwise. |
| 18:52.55 | ChannelZ | if start_digit is negative, it starts at the end and counts backwards |
| 18:53.45 | eja | so what's the best way to tell if a system has an attached analog line? i see an entry in here for TRUNK=zap/g2 but no zap commands are available in the cli |
| 18:54.18 | [TK]D-Fender | eja: then chan_zap isn't even loaded |
| 18:55.21 | eja | so that's probably not even used anymore then |
| 18:55.45 | eja | or it should be used but not working since the module's not loaded lol |
| 18:56.33 | eja | lspci does show a tdm2400p. any way to tell what's attached to it? |
| 18:57.16 | [TK]D-Fender | eja: Why would you not know? |
| 18:58.29 | eja | the asterisk server is in arizona and i'm in california. i can't physically check. |
| 18:59.40 | [TK]D-Fender | eja: And you weren't told what was installed? |
| 19:00.45 | ChannelZ | and also Zap is now DAHDI so if you are reading an ancient dialplan but the system was updated..... |
| 19:00.50 | eja | nope. someone may have been when the system was acquired by us but it doesn't look like it was documented. |
| 19:01.01 | eja | ChannelZ: yeah they're running 1.4 |
| 19:01.37 | [TK]D-Fender | eja: dmesg should tell you what modules are installed. Configure all the channels and see what comes up red |
| 19:01.55 | [TK]D-Fender | eja: Once chan_zap successfully loads |
| 19:02.30 | *** join/#asterisk Azrael808 (~peter@dandlgreen.demon.co.uk) |
| 19:04.34 | *** join/#asterisk gandhijee_ (~akp@50.12.169.99) |
| 19:06.07 | eja | looks like chan_zap isn't compiled. dmesg shows port 13-24 Installed -- AUTO FXO (FCC mode). is there a command line utility to check if any of those ports have a live analog line w/o loading chan_zap? |
| 19:07.55 | [TK]D-Fender | eja: So the module is physically not present in the proper modules folder? |
| 19:08.30 | [TK]D-Fender | eja: I think zt_monitor might do it... I've never had to use any of the non-* debugging tools though... |
| 19:34.07 | gandhijee_ | eja, you can use the tool in asterisk and see if ther is voltage on the line |
| 19:35.07 | gandhijee_ | dadhi show regdump <chan number> granted this only works with the digium cards and the clones |
| 19:35.48 | *** join/#asterisk gandhijee_ (~akp@50.12.169.99) |
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| 19:55.13 | acidfoo | gandhijee_, namaste bapu |
| 20:02.58 | *** join/#asterisk Tim_Toady (~fuzzy@195.74.247.28.dsl.dyn.forthnet.gr) |
| 20:09.15 | *** join/#asterisk LordRain (~ItsRain@1800-lan.idsno.net) |
| 20:09.21 | LordRain | hello all |
| 20:09.37 | gandhijee_ | hmm, any chance anyone knows what files to dr up to get a polycom 550 with the 4.x firmware working with an old trixbox install? |
| 20:10.07 | Naikrovek | same as it would be for any other asterisk server |
| 20:10.09 | LordRain | can anyone give me a kick in the right direction on an upgrade? I am upgrading from 1.0.08CVS to 1.8 (yeah, hardcore old) and am curious to know what I'll need to know in terms of configuration file changes (syntax, etc) |
| 20:10.28 | Naikrovek | lots of changes, lord rain. easier to redo your config on a new system than upgrade, probably |
| 20:10.36 | LordRain | well, thats the plan |
| 20:10.44 | LordRain | i have a new box running 1.8 |
| 20:10.52 | LordRain | but im somewhat new to asterisk... |
| 20:11.18 | Naikrovek | aren't we all |
| 20:11.34 | LordRain | quite a few extensions in the extensions.conf, i.e.: exten => 504488xxxx,1,Dial,IAX2/pbx2:pbx2@x.x.x.x/504488xxxx |
| 20:11.47 | LordRain | but the syntax doesn't look exactly right for how it does in 1.8 |
| 20:11.48 | Naikrovek | yeah that syntax has changed a bit. |
| 20:12.10 | Naikrovek | 504488xxxx,1,Dial(IAX2/pbx2:pbx2@x.x.x.x/504488xxxx) |
| 20:12.13 | Naikrovek | note the parens |
| 20:12.41 | Naikrovek | you can also use 'n' after '1' in the priority list for an extension, though the numbers will still work. |
| 20:12.41 | *** join/#asterisk TimeRider (~steve@188-220-34-144.dsl.cnl.uk.net) |
| 20:13.13 | Naikrovek | so 504488,1,Dial... then |
| 20:13.24 | gandhijee_ | ok, well in that case is there some software from polycom to help update a cfg file? |
| 20:13.24 | Naikrovek | 504488,n,VoiceMail... or whatever |
| 20:13.34 | Naikrovek | gandhijee_: yes, polycom has a config file updater |
| 20:13.50 | gandhijee_ | O, on their website |
| 20:13.52 | Naikrovek | it converts from pre-3.3 to 3.3 or 4.0 config file structure |
| 20:13.59 | Naikrovek | yes |
| 20:14.10 | Naikrovek | polycom have produced many documents on how to do it |
| 20:14.17 | Naikrovek | they describe it better than any of us can |
| 20:15.09 | LordRain | so... |
| 20:15.14 | LordRain | the major changes are mostly syntax |
| 20:15.16 | LordRain | i.e. the parens? |
| 20:15.22 | LordRain | will it work without them? |
| 20:15.25 | LordRain | or they're required now? |
| 20:15.40 | LordRain | because in the 1.0 config, i see a mixture of usage of parens and not using them |
| 20:17.32 | Naikrovek | required now, i believe |
| 20:17.42 | Naikrovek | some delimiters changed, some didn't |
| 20:17.49 | Naikrovek | check the changelogs from 1.0.0 to 1.8 |
| 20:17.59 | Naikrovek | lots of reading, but you'll get your answers there |
| 20:18.49 | LordRain | very true on the lots of reading... |
| 20:18.53 | LordRain | good thing im a lit buff :P |
| 20:18.59 | LordRain | thank you, naikrovek :) |
| 20:19.00 | Naikrovek | :) |
| 20:19.02 | Naikrovek | np |
| 20:19.18 | Naikrovek | you read Game of Thrones series? |
| 20:19.23 | Naikrovek | mr. lit buff? |
| 20:20.34 | Naikrovek | A Song of Fire and Ice |
| 20:20.38 | Naikrovek | anyone... |
| 20:24.08 | keith4 | good books |
| 20:24.19 | keith4 | off topic, though |
| 20:25.49 | *** join/#asterisk kriegerod (~krieger@79.135.222.22) |
| 20:34.32 | *** join/#asterisk Nasga (~Nasga@AAmiens-157-1-124-77.w86-208.abo.wanadoo.fr) |
| 20:38.13 | LordRain | no, i have not... |
| 20:38.15 | LordRain | any good? |
| 20:44.31 | gandhijee_ | Naikrovek, any chance you can point me to this piece of software? |
| 20:44.39 | Naikrovek | one sec |
| 20:45.16 | Naikrovek | http://support.polycom.com/PolycomService/support/us/support/eula/ucs/UCConfig_agreement.html |
| 20:45.20 | gandhijee_ | thank you much |
| 20:45.23 | Naikrovek | click agree or whatever at the bottom |
| 20:45.48 | gandhijee_ | yep, thanks again |
| 20:56.31 | eja | hi guys, what exactly does this line do? Set(CALLERID(num)=${IF($["${CALLERID(num)}"=""]?$380:${CALLERID(num)})}) |
| 20:56.57 | eja | set callerid(num) to whatever the caller id number is if it's not empty and if it is empty set it to $380? |
| 20:58.49 | [TK]D-Fender | <PROTECTED> |
| 20:59.02 | [TK]D-Fender | that is an extra $ |
| 21:00.51 | eja | will it set callerid(num) to 380 or whatever's stored in $380? $380 doesn't look like a proper variable |
| 21:01.58 | *** join/#asterisk _Corey_ (~chatzilla@mail1.infradapt.com) |
| 21:04.33 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 21:05.48 | *** join/#asterisk treborsux (~IceChat77@75-144-117-117-Jacksonville.hfc.comcastbusiness.net) |
| 21:05.57 | treborsux | help |
| 21:06.14 | treborsux | zap channel dids has no effect |
| 21:08.01 | [TK]D-Fender | treborsux: #freepb |
| 21:08.03 | [TK]D-Fender | x |
| 21:08.05 | [TK]D-Fender | ^ |
| 21:08.13 | [TK]D-Fender | treborsux: This isn't a problem for here |
| 21:10.32 | *** join/#asterisk twodogs (~twodogs@kasai.biohazard.seattle.wa.us) |
| 21:28.53 | *** join/#asterisk nny (~Scott@cpe-174-107-223-014.sc.res.rr.com) |
| 21:29.13 | gandhijee_ | so on my 550's with the new firmware, the phone icons don't go to the full black, they are just the white outline |
| 21:30.11 | [TK]D-Fender | Not registered then |
| 21:30.29 | nny | dimensioning question, looking to create a scalable server that can support 1000 or so registrations and concurrent. I can either build it as a single solution or multiple servers, the latter seems smarter and simpler. I have read using SER in combination with * to handle it as well. With modern hardware (12 core servers, 32 + Gb of ram) what's the theoritical safe limit for a single solution without SER? |
| 21:30.36 | gandhijee_ | i can make and receive calls on them.... |
| 21:30.57 | gandhijee_ | so it kinda seems like it is |
| 21:30.58 | nny | this assumes minimal codec translation, no recording and no hardware lines |
| 21:31.12 | [TK]D-Fender | gandhijee_: Doesn't mean anything by itself like that |
| 21:31.47 | gandhijee_ | [TK]D-Fender, they show up registered on the asterisk side as well. |
| 21:32.47 | [TK]D-Fender | gandhijee_: It is mildly possible your icon definitions are messed up due to improper provisioning configs, but the likelyhood of that being the case and the rest of the phone functioning at all is extremely unlikely |
| 21:33.29 | gandhijee_ | [TK]D-Fender, i see |
| 21:33.47 | gandhijee_ | i guess thats what i get for being lazy and using this old ass copy of trixbox |
| 21:34.36 | [TK]D-Fender | gandhijee_: That doesn't actually mean anything by itself |
| 21:35.06 | gandhijee_ | what doesn't mean anything by itself |
| 21:35.19 | gandhijee_ | what do i need to do to get more details i guess |
| 21:36.29 | [TK]D-Fender | "Trixbox: does not telll me anything about what I should expect the condition of your phone to be. |
| 21:36.37 | gandhijee_ | ah |
| 21:37.12 | gandhijee_ | no i was saying i ended up with improper configs one the phone as i just shoved on trixbox then just enough files for the phones to stop boot looping |
| 21:37.26 | gandhijee_ | ie thats what i get for being lazy and shoving on trixbox |
| 21:46.13 | nny | [TK]D-Fender: what's the theoretical limit you'd consider running concurrently on one box, pretending that the hardware is some kind of clustered setup where hardware limits are higher than normal? |
| 21:46.38 | nny | [TK]D-Fender: my guts saying multiple servers, keep them sane in concurrent, but all the info I find is 4 years old or so |
| 21:47.27 | [TK]D-Fender | nny: I've never really pushed any limits with any system I've dealt with and most weren't more than around 200-300 in size |
| 21:47.38 | [TK]D-Fender | nny: Don't have better hard stats for you, sorry |
| 21:47.39 | nny | yeah that's my gut's limit too |
| 21:47.53 | nny | no problem, thanks. I can engineer it around known sanity, thanks |
| 21:48.00 | [TK]D-Fender | That wasn't an implied limit... othat than the limit of my experience, not the box |
| 21:48.04 | nny | same for me |
| 21:48.35 | [TK]D-Fender | 1000 is quite workable with nominal transcoding, but if you hit G.729 or similarly complex that can really stack up |
| 21:49.21 | [TK]D-Fender | Media alone isn't an issue for even a mild enthousiast-grade box |
| 21:49.32 | [TK]D-Fender | conversion could be though |
| 21:49.33 | nny | this client is looking at clustered dual quad-cores |
| 21:50.08 | [TK]D-Fender | nny: Dependsing how your deployment has to work, that certainly sounds healthy |
| 21:51.00 | nny | [TK]D-Fender: just passthrough (outbound calls only) and messages played to voicemails. The latter is the part I need to drill down on, I assume first is using native codecs on those files |
| 21:51.26 | [TK]D-Fender | nny: Media only then, you shouldn't need a 2nd box at all... |
| 21:51.29 | *** join/#asterisk jetlag (~jetlag@pool-71-168-240-26.cmdnnj.east.verizon.net) |
| 21:51.39 | [TK]D-Fender | other than safety's sake of redundancy |
| 21:52.00 | nny | [TK]D-Fender: wondering what the hit is to playabck g711 files to the channel, i assume disk read speeD? |
| 21:57.58 | [TK]D-Fender | nny: Playback is su-negligable |
| 21:58.16 | [TK]D-Fender | nny: that's a raw file read... not anything to blink at |
| 21:59.01 | nny | [TK]D-Fender: roger, thanks for the info. I'll probably just plan for both and adjust accordingly. Appreciate your input as always |
| 22:07.37 | *** join/#asterisk bbourdage (~bbourdage@75-150-221-205-Illinois.hfc.comcastbusiness.net) |
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| 22:38.36 | *** join/#asterisk Alborracho (c819d8de@gateway/web/freenode/ip.200.25.216.222) |
| 22:40.25 | Alborracho | how can i know if a TE420B card is broken? i los signalling form one day to another and i dont see any traffic with "ss7 dump" |
| 22:40.40 | Alborracho | dadhi_tool show the links ups fisically but no signalling |
| 22:41.04 | Alborracho | im using asterisk 1.4 |
| 22:41.13 | Alborracho | and chan_ss7 2.0 |
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| 22:49.27 | *** join/#asterisk drdru (~Adium@c-69-181-167-85.hsd1.ca.comcast.net) |
| 22:50.01 | drdru | how can I setup SIP Messaging with asterisk? it looks like it's in dialplan, but I'm struggling to find a sample |
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| 23:11.46 | ChannelZ | drdru: All I can offer is I believe you have to be running Asterisk 10 for that. |
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