| 00:12.13 | Rhomber | lol |
| 00:12.40 | Rhomber | so with the Dial() command, you can call a gosub or macro when it connects... can you call another application directly? like Agi? |
| 00:13.03 | Rhomber | I am calling the dial command from the agi and it would be good to not have to revert back to the dialplan |
| 00:24.42 | [TK]D-Fender | Rhomber: Who said that would exit your AGI? |
| 00:31.26 | Katty | hi fender. |
| 00:38.23 | [TK]D-Fender | Katty: Mew. |
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| 01:29.49 | Rhomber | [TK]D-Fender: What has that got to do with what I asked? |
| 01:29.58 | Rhomber | 'exiting' has nothing to do with the question :) |
| 01:31.02 | [TK]D-Fender | Rhomber: Depends on the interpretation of "revert" |
| 01:31.18 | [TK]D-Fender | Rhomber: And you left plenty of room |
| 01:31.29 | [TK]D-Fender | Rhomber: And nothing stops you from calling an AGI in that macro |
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| 01:54.04 | [TK]D-Fender | heads out for a while |
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| 02:13.04 | HolTech | why would a analog phone in an fxs port be replying a 404 Not found ? device setup or asterisk config ? |
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| 04:10.09 | Rhomber | [TK]D-Fender: Yes, calling an AGI in the macro.. meaning that it needs to have a macro.. which is inelegant when all I want is the AGI |
| 04:10.26 | Rhomber | though this is what I am currently doing... asterisk in general is inelegant though it does work well |
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| 05:25.44 | igcewieling | Rhomber: Dialing from inside an AGI is never elegant |
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| 05:35.09 | seik0 | Hi again, everybody. On friday I asked if something could help me, when I use "register" for outside sip-peer and get lost internet connection. Suggestion was to write down all dns-names to /etc/hosts. So I tried, but was no luck. Now I'm going to try dummy peer, which is always "up". |
| 05:40.57 | [TK]D-Fender | Rhomber: This is a 2-line global fix. Nothing to even blink at |
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| 08:32.49 | seik0 | A question for zaptel: we have working digium card (te120p). I wander, why ztscan shows "syncsrc=0", zttool shows "Internal clocking" for sync source. But in zaptel.conf span is configured as "span=1,1,0,ccs,hdb3,crc4", where second "1" means "slave" syncing. So the question - is it a "feature" or zaptel really thinks internal clocking is appropriate? How can I test it? |
| 08:34.28 | WIMPy | You know that you are several yaers out of date? |
| 08:34.45 | WIMPy | Apart from that: Is that port receiving sync? |
| 08:34.46 | seik0 | of course |
| 08:35.10 | seik0 | you mean, is port really should be slave? |
| 08:35.47 | seik0 | in fact, card most of the time workis ok |
| 08:36.47 | seik0 | so some kind of syncing is present |
| 08:36.50 | WIMPy | No is it connected to a working port? |
| 08:37.02 | seik0 | yes |
| 08:37.33 | seik0 | it's up just now - works ok, but for some reason shows "internal clocking" |
| 08:38.42 | WIMPy | Sounds like a bug. |
| 08:40.10 | seik0 | i'll try to set wrong timing source to see if something will change, but later in the evening |
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| 09:26.06 | zamba | i'm running asterisk on ubuntu.. and for each new kernel upgrade the dahdi module is broken.. is there anything i can do to automatically update the dahdi module whenever i upgrade the kernel? |
| 09:33.19 | WIMPy | It's a kernel driver that's not part of the kernel. That's just the way it is. |
| 09:34.10 | WIMPy | That't tha fine difference between hardware supported by Linux and Hardware with support for Linux. |
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| 09:58.33 | apurvtwr | hi . I am working with Asterisk 1.8 and have set up a dialplan where an inbound call is put queue of agents. These agents have a SIP number and a follow me. When the call comes in it is first made to the SIP umber which if not answered is forwarded to the follow me. |
| 09:59.15 | apurvtwr | this works fine, except that I want to know where did the agent pick up the call ( i.e. on his SIP or on his follow me number) |
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| 10:02.50 | apurvtwr | any pointers on how to achieve this? |
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| 12:33.41 | mbt | Anyone have any clue where the source code for the Digium phone firmware is? As I understand it, they run Linux, meaning I should be able to find its source somewhere... |
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| 12:36.16 | [TK]D-Fender | mbt, Says who? There is commercial code for linux you know.... |
| 12:36.36 | [TK]D-Fender | mbt, Just like there is OSS for Windows. |
| 12:36.52 | mbt | [TK]D-Fender: I understand that. |
| 12:37.06 | mbt | [TK]D-Fender: But if the Linux kernel is on the device, I should be able to get the source for _that_, as it is GPL. |
| 12:37.12 | mbt | [TK]D-Fender: That's all I am interested in. |
| 12:37.21 | mbt | [TK]D-Fender: I don't care about Digium's custom code. |
| 12:38.21 | mbt | [TK]D-Fender: Just because a commercial binary is running on top of the Linux kernel, doesn't mean the right to see the source of the kernel in a device is invalidated or revoked. :-) |
| 12:38.42 | coppice | who says the phones run Linux? |
| 12:38.49 | [TK]D-Fender | coppice, shh!!! |
| 12:40.53 | sekil | coppice: Linux run phones but don't tell |
| 12:41.10 | mbt | coppice: According to nmap, it is running a Linux kernel between 2.6.9 and 2.6.33 |
| 12:41.40 | mbt | Granted, that's based on fingerprinting, but that also matches what I've read about the hardware in the phone (who would use anything other than linux on the hardware inside the things, unless they're nuts?) |
| 12:42.11 | coppice | lots of phones run linux, but lots of others don't |
| 12:42.19 | mbt | So I guess to properly answer your question, coppice, the phone says it does. :-) |
| 12:42.51 | coppice | unless you own the phones there's no requirement for the supplier to give you access to the source code |
| 12:43.00 | mbt | coppice: That'd be correct. |
| 12:43.10 | mbt | coppice: That said, there was no notice in the box about GPL code. |
| 12:43.28 | mbt | coppice: And yes, I own a D50. |
| 12:43.29 | [TK]D-Fender | mbt, Or more precisely your extrapolation of what nmap says does... |
| 12:44.07 | [TK]D-Fender | mbt, Tried contact Digium for this yet? |
| 12:44.38 | *** join/#asterisk work (work@82.137.193.222) |
| 12:45.02 | mbt | [TK]D-Fender: I wouldn't know who to call there. I can't find any information on who would manage that stuff, and I'm not sure that the phone support people would know anything about the software stack at such a low level; they only seem to support the phone's functionality. |
| 12:45.33 | mbt | Anyway, I was just wondering if anyone knew off the top of their head... if not, I'll continue researching. |
| 12:47.40 | work | please I have asterisk + dahdi + ubuntu 12 + digium te405p connected to E1 with signaling PRI_CPE how to record all call on this E1 ? |
| 12:47.43 | *** join/#asterisk luckman212 (~luckman21@unaffiliated/luckman212) |
| 12:48.33 | [TK]D-Fender | work, "core show application monitor" <- |
| 12:49.03 | mbt | work: https://www.google.com/search?q=asterisk+record+all+calls |
| 12:49.27 | [TK]D-Fender | work, And I see in another channel that you mention "FreePBX". This is not the place to find support for it |
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| 12:57.20 | work | yes I new in this field and Iam setup those kinde of technology first time |
| 13:00.18 | work | should I use both asterisk and freepbx ? |
| 13:00.34 | [TK]D-Fender | work, Depends what you want to do |
| 13:00.51 | chatran | anyone know what is this: Unable to find a codec translation path from 0x100 (g729) to 0x8 (alaw) |
| 13:01.01 | mbt | work: I would recommend working with Asterisk itself, first, before using front-ends for it. |
| 13:01.12 | mbt | work: But that's just my own personal opinion. |
| 13:01.25 | work | I have e1 link and degim card with pri_cpe |
| 13:01.34 | [TK]D-Fender | chatran, It's exactly what it says. Asterisk can't translate between the 2 codecs |
| 13:01.38 | work | I want to record all calls ? |
| 13:01.58 | [TK]D-Fender | work, What calls? Calls don't exist yet. You haven't defined any PBX functionality |
| 13:02.43 | [TK]D-Fender | work, you need to look at what else your system will be doing and how |
| 13:03.09 | work | no it is project to forward all calls from exechange to my asterisk then record all calls and cdr |
| 13:03.20 | work | I have calls and log now |
| 13:03.25 | mbt | work: Does your Asterisk installation answer incoming calls to your E1? |
| 13:03.30 | [TK]D-Fender | chatran, G.729 is a patented and licensed codec. Have you purchased and installed your licenses from Digium's website? |
| 13:03.32 | work | yes |
| 13:03.39 | mbt | work: Okay, then I repeat from earlier: https://www.google.com/search?q=asterisk+record+all+calls |
| 13:03.52 | mbt | work: The first hit there shows you how to record calls automatically from the dialplan. |
| 13:03.58 | chatran | [TK]D-Fender well im conecting pap2t with an 4 port VoIP GW |
| 13:04.03 | [TK]D-Fender | mbt, He's using FREEPBX. |
| 13:04.03 | work | set debug span 1 is working and show me calls and numbers |
| 13:04.25 | [TK]D-Fender | chatran, Go install the paid codec or change what your device is using |
| 13:04.27 | mbt | [TK]D-Fender: Ah, I back out of that one, then. I don't use front-ends. |
| 13:05.01 | chatran | [TK]D-Fender i dont have license |
| 13:05.13 | [TK]D-Fender | chatran, Then change what your device uses |
| 13:05.24 | chatran | i cant change |
| 13:05.34 | [TK]D-Fender | chatran, ... because? |
| 13:05.35 | mbt | chatran: Then buy the codec license. :-) |
| 13:05.36 | work | yes please this is what I want extention dial plan is complex I need to organize freod optin for part of cdial plan FROM-PSTN |
| 13:06.07 | chatran | becose its an infraestruture that worked before... and is working now... just give this message |
| 13:06.32 | work | thankx mbt I will see |
| 13:06.55 | [TK]D-Fender | chatran, There is nothing to discuss here. Your device is asking for G.729 and your Asterisk server doesn't support it currently. One of them has to change. |
| 13:07.43 | chatran | but is working... |
| 13:08.00 | [TK]D-Fender | chatran> anyone know what is this: Unable to find a codec translation path from 0x100 (g729) to 0x8 (alaw) <- this does not look like "working" to me |
| 13:08.03 | chatran | even with no license the codec works ? |
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| 13:08.44 | work | mbt : this code |
| 13:08.45 | [TK]D-Fender | chatran, And you haven't shown us enough to prove where this is actually occurring in your call-flow |
| 13:08.47 | work | ; This fake context records all outgoing calls to /var/spool/asterisk/monitor in wav format. |
| 13:08.47 | work | [fake-outgoing-context] |
| 13:08.47 | work | exten => s,1,Answer() |
| 13:08.47 | work | exten => s,n,Monitor(wav,,b) |
| 13:08.47 | work | exten => s,n,Dial(DAHDI/g0/${EXTEN}) |
| 13:08.48 | work | exten => s,n,Hangup() |
| 13:08.51 | [TK]D-Fender | work, STOP |
| 13:08.56 | [TK]D-Fender | work, do NOT flood in here. |
| 13:08.57 | [TK]D-Fender | ~pb |
| 13:08.57 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood 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 install pastebinit with yum or aptitude. |
| 13:08.59 | [TK]D-Fender | ^^^ |
| 13:09.03 | work | should be in extention file |
| 13:09.05 | work | ? |
| 13:09.26 | mbt | work: Wow, please don't flood. The Monitor line is the important one. |
| 13:09.41 | [TK]D-Fender | work, that code is broken all by itself and does not tie into FreePBX in any way |
| 13:09.59 | [TK]D-Fender | ~book |
| 13:09.59 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 13:10.02 | work | sorry I do not want to annoy any one I am so sorry |
| 13:10.03 | [TK]D-Fender | work, ^^^ |
| 13:10.08 | mbt | work: Though I am pretty sure the mixmonitor is what you'd want. |
| 13:10.13 | chatran | [TK]D-Fender how i make good log ? asterisk -rvvvvvvvvvv ? |
| 13:10.28 | mbt | work: http://www.voip-info.org/wiki/view/MixMonitor |
| 13:10.32 | [TK]D-Fender | chatran, For starters. You may need SIP DEBUG on top of that |
| 13:10.35 | work | ): |
| 13:10.44 | [TK]D-Fender | mbt, Same thing |
| 13:10.48 | chatran | ok let me see sip debug |
| 13:11.04 | mbt | [TK]D-Fender: Oh. Didn't they used to do different things at one point? |
| 13:11.23 | [TK]D-Fender | go compare them |
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| 13:12.47 | mbt | [TK]D-Fender: Ah, they still do. Monitor by default records both legs in different files, MixMonitor by default records both legs in the same file. |
| 13:14.02 | [TK]D-Fender | mbt, And Monitor.... has an option to mix. And exec external scripts.... and.... |
| 13:14.40 | mbt | [TK]D-Fender: Yes it does. Admittedly much more versatile. I was merely trying to suggest the simplest solution, that's all. |
| 13:14.51 | mbt | [TK]D-Fender: My apologies if that is discouraged. |
| 13:15.23 | [TK]D-Fender | mbt, Nope, just nudging you towards information.... |
| 13:15.48 | [TK]D-Fender | mbt, I wonder if MixMonitor has any unique value at this point... |
| 13:16.18 | mbt | [TK]D-Fender: Just what you give it; a channel variable will do nicely. |
| 13:16.30 | chatran | ok [TK]D-Fender look: http://pastebin.com/ad4dny2F its a full log... |
| 13:18.10 | [TK]D-Fender | chatran, Clearly your H.323 end is insisting on G.729. Until you finx this it's going to keep dying like that |
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| 13:27.41 | chatran | [TK]D-Fender i had saw some g.729 plugins on net to asterisk that no need license. any works? |
| 13:29.05 | [TK]D-Fender | chatran, G.729 is a patented and licensed codec. www.digium.com <- |
| 13:30.20 | igcewieling | chatran: If Digium permits discussion of pirated g729 codecs, they could lose their license to the codec. Therefore pirated g729 codec discussion is prohibited here and could get you banned from the channel. |
| 13:30.37 | chatran | ok |
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| 13:36.56 | *** join/#asterisk as001 (~uros@82.117.198.142) |
| 13:37.24 | as001 | Hi is it possible to setup channel for Async AGI from Asterisk Manager in asterisk 1.8.20 ? |
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| 13:39.23 | as001 | When I put AGI(agi:async) in dialplan it hangs until I do break but I would like to do it during call with Asterisk Manager Interface. |
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| 13:42.01 | igcewieling | "do break" |
| 13:42.08 | igcewieling | is that anything like doing lunch? |
| 13:43.11 | igcewieling | as001: also that AGI call does not look valid. What is the ACTUAL AGI command? |
| 13:43.24 | coppice | isn't a do break a toilet break? |
| 13:43.33 | igcewieling | hey coppice! |
| 13:43.56 | as001 | I want to do EXEC PLAYBACK /var/lib/asterisk/sounds/en/busy-hangovers on sip channel. |
| 13:44.44 | igcewieling | I was referring to the AGI() command in the dialplan |
| 13:44.54 | as001 | AGI(agi:async) |
| 13:45.15 | *** join/#asterisk mnathani (~mnathani@198-84-231-11.cpe.teksavvy.com) |
| 13:45.43 | srp_ | [TK]D-Fender: hi. we had discuss about Linksys SPA 3102 voice gateway + router a few weeks back.. I wanted to know if Linksys PAP2 or PAP2T is comaptible with asterisk? |
| 13:45.53 | as001 | dialplan stops at that point until I do "ASYNCAGI BREAK" on channel from manger. |
| 13:46.43 | *** join/#asterisk igcewieling (~igcewieli@ip98-183-25-31.pn.at.cox.net) |
| 13:46.58 | [TK]D-Fender | srp_, Yes, but they are FXS only |
| 13:47.38 | chatran | ok [TK]D-Fender i do no have anymore that g729 messages... now when i call the people hear me... but when they call, they cant hear me. NAT problem again ? |
| 13:47.39 | igcewieling | as001: like I said that does not appear to be a valid AGI command. You need AGI(/path/to/your/script) Do a "core show application AGI" for syntax info |
| 13:48.07 | srp_ | [TK]D-Fender: oh.. okay.. |
| 13:49.26 | as001 | when I try EXEC PLAYBACK /var/lib/asterisk/sounds/en/busy-hangovers before BREAK before ASYNCAGI BREAK in manager I hear message but after BREAK I can see "command succesfully queues" but I can't hear sound on channel. I think call is ok its from manual. |
| 13:50.19 | as001 | I followed this instructions : http://www.scribd.com/doc/54892783/391/Async-AGI%E2%80%94AMI-Controlled-AGI |
| 13:55.32 | [TK]D-Fender | chatran, Can't tell for sure. You've masked IP's and haven't shown us any config or described what you've got set up |
| 13:57.03 | *** part/#asterisk igcewieling (~igcewieli@ip98-183-25-31.pn.at.cox.net) |
| 13:58.12 | srp_ | [TK]D-Fender: and one more small doubt.. on some sites I see "Linksys SPA3102 - Unlocked". Is it safe to use an unlocked device? |
| 13:59.20 | chatran | on sip debug i got: Content-Length: 0 ... its normal ? |
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| 14:01.06 | [TK]D-Fender | srp_, If you can get to the admin then you're fine. |
| 14:01.27 | srp_ | [TK]D-Fender: okay.. |
| 14:01.31 | as001 | What I need is ablility to do 'AGI(agi:async)' somehow from one AMI user event before I try that Exec Playback later in event and after that to do ''ASYNCAGI BREAK" . Is it possible ? |
| 14:02.59 | as001 | if I do just exec playback I get 'channel is not ready for Async AGI' as warning in cli. |
| 14:03.01 | work | please nice poeple how to name recorded file.wav as SOURCENUMBER_DATE_TIME.wav |
| 14:05.49 | work | extent=> s,n,mixmonitor (???????????????) |
| 14:06.09 | mbt | work: http://www.voip-info.org/wiki/view/Asterisk+variables -- use those for your filename. |
| 14:06.25 | [TK]D-Fender | ~book |
| 14:06.25 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 14:06.27 | [TK]D-Fender | work, ^ |
| 14:06.46 | [TK]D-Fender | work, "core show function STRFTIME" |
| 14:08.56 | mbt | work: Example, ${CALLERID(num)}_${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}) |
| 14:09.45 | *** join/#asterisk vfabi (~fabi@host-static-93-116-255-140.moldtelecom.md) |
| 14:10.16 | mbt | work: The predefined channel variables alone give you a _lot_ to work with. :-) |
| 14:11.38 | as001 | Core of question is can I start Async AGI from Asterisk Manager Interface ? |
| 14:11.50 | mbt | work: One other tip: searching google for "asterisk <concept>" or "asterisk <application>" will usually give you a link to voip-info.org as the first result. Sometimes the info there is a little out of date, but it is still a very useful resource nonetheless. |
| 14:12.07 | work | (mbt) thankx I hope I could send you some choclets for your answers big thankx for you and [TK]D-Fender |
| 14:12.26 | mbt | work: No worries, glad to help. |
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| 14:16.26 | work | then one query please for my project freePBX not needed or yes -- is asterisk play the same freepbx or I have to install both ? |
| 14:16.39 | xuw | we are experiencing echo over SIP. any way to avoid it ? |
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| 14:17.19 | [TK]D-Fender | work, FreePBX is a configuration front-end for Asterisk. |
| 14:17.54 | mbt | work: I haven't ever used FreePBX except in testing, and I found that I prefer configuring Asterisk directly; I think it is far more flexible that way, and it's also really useful to be intimiately familiar with the Asterisk configuration files, IMHO. |
| 14:18.05 | [TK]D-Fender | work, is it needed? No, of course not. |
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| 14:19.46 | work | ok then if I remove now freepbx I will lose web admin configuration (http://myip/admin) or no |
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| 14:20.53 | [TK]D-Fender | work, Yes. |
| 14:21.15 | [TK]D-Fender | wrokthat is what freepbx is. A Web admin interface that is expected to do your entire job for you. |
| 14:21.22 | mbt | work: Asterisk is directly configured through the use of configuration files in /etc/asterisk, or (if you configure it, or something like FreePBX configures it for you) through realtime configuration in databases and the like. |
| 14:21.24 | [TK]D-Fender | work, In its limited toaster way |
| 14:22.12 | work | I will try to get ride of using it step by step til be familier with asterisk file configration |
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| 14:36.19 | xuw | we are experiencing echo over SIP. any way to avoid it ? |
| 14:38.15 | [TK]D-Fender | xuw, Echo isn't from SIP. It's from a TDm endpoint |
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| 15:00.55 | as001 | ok it seems I can just break async agi from AMI and can not start it. I suggest to developers to add extra manager event Playback which will Playback some file on some channel. |
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| 18:01.20 | v0lZy | hey guys, I get asterisk make: *** [datafiles] Error 2 when compiling asterisk |
| 18:01.22 | v0lZy | any ideas? |
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| 18:12.28 | navaismo | nope can you pb the output |
| 18:12.46 | v0lZy | machine crashed :| |
| 18:12.52 | v0lZy | have to go through the entire process again.. |
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| 18:23.10 | drmessano | It "crashed"? |
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| 18:55.07 | edong23 | http://pastie.org/private/tyrqhtqpq3w9kd1guaqra |
| 18:55.16 | edong23 | im having an issue where music on hold isnt working... |
| 18:55.23 | edong23 | it was working before i upgraded |
| 18:55.55 | edong23 | that was verbose set at 100 i just get ringing... im a bit stuck... everything seems to be set this is simply asterisk's default hold music samples |
| 18:56.46 | edong23 | it could be unrelated to moh... perhaps i borked the queues? |
| 18:58.56 | navaismo | well your pb doesnt show much and the cli show that the moh was started then stopped. |
| 18:59.30 | navaismo | You can create one exten to test directly with Answer then MusicOnHold(default) |
| 19:00.20 | edong23 | i can pastie more im sorry, it just all looks right. ill do that though.. asterisk is running as root, so it shouldnt be permissions |
| 19:00.26 | edong23 | let me try the exten |
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| 19:04.10 | edong23 | navaismo: yeah, that works fine, i hear music... |
| 19:04.13 | edong23 | hm... |
| 19:05.09 | edong23 | i only have default and moh show files shows them in /var/lib/asterisk/moh ... |
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| 19:06.05 | edong23 | navaismo: i may see it... |
| 19:07.00 | edong23 | nope |
| 19:07.01 | edong23 | hm... |
| 19:07.11 | navaismo | so the last pb is for a queue? |
| 19:07.38 | edong23 | yeah, its a queue |
| 19:07.44 | edong23 | it was working before upgrade |
| 19:07.54 | edong23 | if you tell me what you need, i can pastie it... |
| 19:07.56 | navaismo | and there is an answer before the queue cmd? |
| 19:08.00 | v0lZy | i got it working guys |
| 19:08.04 | edong23 | im just not sure exactly |
| 19:08.08 | edong23 | answer... |
| 19:08.10 | v0lZy | apparently my error was from make 3.80 |
| 19:08.11 | edong23 | hold please |
| 19:09.39 | edong23 | navaismo: in extensions.conf under this context i answer() set callerid name to append a few letters then playback an ivr then queue |
| 19:09.56 | edong23 | i can obfuscate it slightly and pastie it if you need |
| 19:10.46 | navaismo | can you paste the complete cli output with a sip debug? |
| 19:11.23 | edong23 | ack... um... its a bit of a busy system, but let me see what i can do |
| 19:11.33 | edong23 | there will likely be lots of sip exchanges |
| 19:11.40 | edong23 | im not sure... you think this is a sip issue? |
| 19:12.32 | edong23 | it starts and stops music on hold pretty much immedately after a few "connected line changed. Saving it until answer" a few times |
| 19:15.07 | navaismo | how look the queue cmd? fancy options or only queue(queuename) |
| 19:15.50 | edong23 | n,Queue(all-queue,tR,,,120) |
| 19:15.57 | edong23 | i mean... is that fancy? |
| 19:16.01 | edong23 | it worked fine in 1.6 |
| 19:16.03 | edong23 | i think... |
| 19:16.09 | edong23 | maybe this could be an astdb thing... |
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| 19:16.22 | edong23 | im not sure... |
| 19:17.04 | *** join/#asterisk work (work@82.137.193.222) |
| 19:18.05 | edong23 | hm... |
| 19:18.19 | edong23 | when i do queue show and that queue, i see no calls |
| 19:18.20 | edong23 | even when i call it |
| 19:18.24 | edong23 | even when they answer |
| 19:18.28 | work | hello please PRI is not exist in asterisk CLI |
| 19:19.07 | [TK]D-Fender | Then you didn't stall libpri |
| 19:19.12 | [TK]D-Fender | install* |
| 19:20.22 | work | for sure I installed it from source code make and make install do you recommand me to reinstall it agian ? |
| 19:20.45 | edong23 | work: did you do it before asterisk? |
| 19:20.56 | work | yes |
| 19:21.24 | [TK]D-Fender | Show us that DAHDI is installed and configured |
| 19:21.25 | [TK]D-Fender | ~pb |
| 19:21.26 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood 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 install pastebinit with yum or aptitude. |
| 19:21.26 | [TK]D-Fender | ^^^^ |
| 19:21.52 | work | befoe asterisk as totorial dhadi then libpri then asterisk |
| 19:25.42 | work | o00h strange /etc/init.d/dahdi/status is working but nothing for dahdi on astererisk CLI |
| 19:26.23 | work | nothing for dahdi or pri |
| 19:27.09 | work | root@HP:~# /etc/init.d/dahdi status |
| 19:27.10 | work | ### Span 1: TE4/0/1 "T4XXP (PCI) Card 0 Span 1" (MASTER) HDB3/CCS/CRC4 |
| 19:27.10 | work | <PROTECTED> |
| 19:27.42 | edong23 | navaismo: hm... it would see that the music is working, just not as expected. after noone answers the queue for a period of time ( i hear ringing during this time) then i get music followed by "you are now first in line" or similar then hold music for a split second then ringing then possibly they answer |
| 19:27.46 | edong23 | is that normal? |
| 19:28.56 | work | on asterisk cli all time (No such command) |
| 19:29.11 | edong23 | work: that generally means the module didnt load |
| 19:29.16 | edong23 | if you did compile it |
| 19:32.42 | navaismo | edong23, your R option is: Ring instead of playing MOH when a member channel is actually |
| 19:32.44 | navaismo | <PROTECTED> |
| 19:33.49 | edong23 | well shit. |
| 19:34.43 | work | edong23 please advice me |
| 19:35.12 | igcewieling | work: I advice you run "module load chan_dahdi.so" in the CLI |
| 19:35.16 | edong23 | work: does dahdi work? |
| 19:35.17 | edong23 | yeah |
| 19:35.24 | edong23 | and ee if it gives you an error |
| 19:35.42 | work | yes dahdi work |
| 19:36.00 | [TK]D-Fender | work, Show us from * CLI both attempts |
| 19:36.01 | [TK]D-Fender | ~pb |
| 19:36.01 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood 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 install pastebinit with yum or aptitude. |
| 19:36.01 | [TK]D-Fender | ^^^ |
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| 19:37.05 | work | Command 'module load chan_dahdi.so' failed. |
| 19:37.05 | work | [2013-04-01 15:36:23] WARNING[3826]: loader.c:458 load_dynamic_module: Error loading module 'chan_dahdi.so': /usr/lib/asterisk/modules/chan_dahdi.so: undefined symbol: ast_channel_linkedid |
| 19:37.05 | work | [2013-04-01 15:36:23] WARNING[3826]: loader.c:848 load_resource: Module 'chan_dahdi.so' could not be loaded. |
| 19:37.05 | work | HP*CLI> |
| 19:37.20 | navaismo | PB!!!! |
| 19:37.44 | work | that is out put of load module on cli |
| 19:38.13 | [TK]D-Fender | work, Stop pasting all that junk in CHANNEL. use a PASTEBIN service as was linked to you many times today |
| 19:38.33 | [TK]D-Fender | work, And chan _dhadi is apparently NOT built right of your running DAHDI version. |
| 19:38.40 | [TK]D-Fender | go recompile and install * to pick up any changes |
| 19:39.35 | igcewieling | looks like an pre asterisk 10 chan_dahdi.so trying to be loaded in an Asterisk 10 or Asterisk 11 |
| 19:39.39 | work | sorry ): I am new in this -- so osrry |
| 19:40.00 | edong23 | work: thats ok, use www.pastebin.com or pastie.org or something |
| 19:40.17 | edong23 | its easier on the eyes |
| 19:40.57 | edong23 | navaismo: thanks for that. been a rough few days but ill admit... i didnt look at those options, just movd the configs |
| 19:42.28 | navaismo | np |
| 19:43.40 | edong23 | but... i dont remember this being the case before. it is HIGHLY possible there was just more call volume when i first walked into this build |
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| 19:49.23 | work | it is nice web site please see all data on http://pastebin.com/crNVF65e |
| 19:50.47 | work | load module + system.conf + dahdi status + chan-dahdi |
| 19:50.48 | [TK]D-Fender | work, Did you recompile and reinstall Asterisk as I suggested? |
| 19:51.35 | work | no |
| 19:52.13 | work | do you want me to do it now ? |
| 19:53.07 | [TK]D-Fender | No, I want you to do it THEN. But I'll settle for "now" |
| 19:55.25 | work | you put me in the corner man --- sorry I am new in this |
| 19:55.40 | work | I will recompile asterisk in make install now |
| 19:55.47 | edong23 | make clean probably first |
| 19:56.21 | work | ): how make clean ? |
| 19:56.27 | edong23 | make clean |
| 19:56.43 | edong23 | in the asterisk source directory you installed from |
| 19:57.03 | work | cd /usr/src/aste* then make clean ? |
| 19:57.11 | edong23 | if thats where it is |
| 19:57.11 | edong23 | yes |
| 19:57.23 | edong23 | <igcewieling> looks like an pre asterisk 10 chan_dahdi.so trying to be loaded in an Asterisk 10 or Asterisk 11 |
| 19:57.29 | edong23 | is this what happened? |
| 19:57.35 | edong23 | i wonder how... |
| 19:57.53 | edong23 | work: did you try to install asterisk/dahdi/libpri from a package manager at some point? |
| 19:58.10 | [TK]D-Fender | work, trash your extracted folder entirely, re-exstract, redo that part from scratch |
| 19:58.28 | [TK]D-Fender | work, and trash your modules folder. |
| 19:58.33 | work | <edong23> no |
| 19:58.46 | [TK]D-Fender | work, typically /usr/lib/asterisk/modules |
| 19:58.47 | [TK]D-Fender | ^ |
| 19:58.52 | igcewieling | [TK]D-Fender: make distclean cleared out the saved ./configure info the last time I tried it. Was quite shocked. |
| 19:58.53 | *** join/#asterisk sekil (~Ognjen@78.24.104.82) |
| 19:58.54 | [TK]D-Fender | (first) |
| 19:59.11 | work | <[TK]D-Fender> I will follow you |
| 20:00.15 | edong23 | isnt make distclean supposed to do that? |
| 20:02.30 | *** join/#asterisk wrbishop (~chatzilla@wrbishop.com) |
| 20:02.51 | work | root@HP:/usr/src# rm -rf asterisk * dahdi-li * libpri-1.4* |
| 20:03.08 | work | root@HP:/usr/lib/asterisk/modules# rm * |
| 20:03.25 | work | <[TK]D-Fender> then what ? |
| 20:04.15 | [TK]D-Fender | work, reextract your asterisk tarball and redo the installation process again |
| 20:04.38 | navaismo | cd / then rm -rf * <----april 1st |
| 20:04.43 | edong23 | um... with those spaces |
| 20:05.21 | edong23 | work: maybe reboot for good measure or at least unload any modules that are loaded |
| 20:05.28 | edong23 | before staring compile and install |
| 20:06.15 | work | ok reboot easier |
| 20:08.06 | work | ./configure && make menuselect && make && make install && make config && make samples is this ok ? |
| 20:10.53 | igcewieling | work: you do NOT want to make samples |
| 20:11.00 | igcewieling | that may overwrite your existing config |
| 20:11.20 | work | ok |
| 20:12.04 | work | what about make menuselect leave it as it ? |
| 20:15.39 | [TK]D-Fender | Yes, you have to redo that |
| 20:16.00 | [TK]D-Fender | packs up to head out |
| 20:18.01 | work | finish |
| 20:22.17 | work | it is work now I will sleep deeply |
| 20:22.32 | work | thanks very much for all |
| 20:23.54 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 20:24.53 | *** join/#asterisk IngeniousTechsLL (~ingenious@c-76-104-115-66.hsd1.va.comcast.net) |
| 20:25.54 | IngeniousTechsLL | Does anyone have time to help me with e FreePBX setup? Using SIP based phones, Arris 4 port FXS telphony modem for POTS lines? |
| 20:26.20 | *** join/#asterisk baronobeefdip (~baronobee@dynsec-efr6-26.tamucc.edu) |
| 20:26.20 | work | please I lost cdr-mysql module http://pastebin.com/53Xwfnkp |
| 20:26.50 | baronobeefdip | I have successfully installed and configured asterisk and I am able to call the phones but I don't hear anything from either the pbx playback or the other caller. |
| 20:27.17 | IngeniousTechsLL | This is a waste of time, no helpful people here AT ALL. |
| 20:27.37 | pabelanger | IngeniousTechsLL: thanks for stopping by |
| 20:27.53 | pabelanger | ~patience |
| 20:27.54 | infobot | i guess patience is a Godly attribute, or the solution for most things. |
| 20:30.33 | baronobeefdip | I have better luck at wait for days at the forums or going over the definitive guide again |
| 20:31.07 | IngeniousTechsLL | YTea I can see that. Can anyone help me? |
| 20:31.27 | pabelanger | ~book |
| 20:31.27 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 20:31.47 | pabelanger | IngeniousTechsLL: explain your issue |
| 20:32.06 | IngeniousTechsLL | I need a very simple question ansered. FXO and FXS, where do they need to go? I already have a 4 port FXS modem that comcast provided for the 4 lines I have through them. |
| 20:33.27 | pabelanger | what do you mean where do they go? |
| 20:34.08 | IngeniousTechsLL | I have 4 SIP based phones. I have Comcast Business class internet modem, with a Arris telephony modem for the phone lines. I had one line previously and one port from the Arris modem was being used. I had a phone line rj11 plugged in it directly from that to the phone. Now I want to use all 4 ports for all 4 lines. On the Arris modem 4 status lights show I have 4 active lines but nothing is |
| 20:34.08 | IngeniousTechsLL | hooked up. I need to utilize the existing cat6 ran thoughout the office for the new SIP phones. |
| 20:34.13 | IngeniousTechsLL | I am missing the FXO part. |
| 20:34.56 | sekil | is totally confused |
| 20:35.03 | IngeniousTechsLL | Lmao |
| 20:35.49 | pabelanger | you have a modem that provides an fxo interface, right? |
| 20:35.54 | sekil | fxo port is where you put your phone line...and fxs is where you put the phone itself or say PBX co port |
| 20:36.16 | IngeniousTechsLL | Ok,let me do this slow this time. I have a comcast business class modem that is providing the internet to the wifi router I have. I have a switch in there also, not sure if its connected to the modem or the wifi router, really doesnt matter. |
| 20:36.21 | sekil | and sip phone is for ..ah..none of that |
| 20:36.44 | IngeniousTechsLL | sekil.....So I bought SIP phones for nothing? |
| 20:37.12 | sekil | well if you have a sip server you can use then no |
| 20:37.25 | IngeniousTechsLL | I was under the impression I could utilize my POTS lines through the Arris modem with the 4 FXS ports and then hook the modem to my network and go from there...then I heard about the FXO. |
| 20:37.47 | IngeniousTechsLL | No SIP server....using POTS lines to Arris FXs to phones. |
| 20:37.49 | sekil | obviously your modem is also an ata |
| 20:38.01 | pabelanger | correct |
| 20:38.04 | pabelanger | http://my.digium.com/en/docs/misc/fxs_fxo_desc/ |
| 20:38.13 | pabelanger | IngeniousTechsLL: I think you need to read this |
| 20:38.15 | pabelanger | ^ |
| 20:38.39 | IngeniousTechsLL | Ive read alot, and I am somewhat of a IT Guru, but the phone stuff is messing with my head. |
| 20:38.50 | sekil | you probably have voip from upstream |
| 20:39.22 | IngeniousTechsLL | So I need to take the Arris 4 port FXS modem and put that in the middle of all the phones lines, then put a FXO thing in the closet? |
| 20:39.24 | pabelanger | The only reason to use FXS / FXO is if you want to use analog phones |
| 20:39.29 | pabelanger | other wisre, you don't need them |
| 20:40.25 | IngeniousTechsLL | Let me explain what I need and maybe you guys and throw some advice....I have 4 POTS lines from comcast, I need to be able to use those 4 lines on 4 different phones at all times. |
| 20:40.41 | IngeniousTechsLL | I really appreciate all the help. |
| 20:40.48 | sekil | those line come over copper cabling? |
| 20:40.59 | sekil | separately then the internet thingy |
| 20:41.00 | sekil | ? |
| 20:41.08 | IngeniousTechsLL | I have 2 ethernet ports at each station one is being used for the internet, I would like to use the other for the SIP phones using my POTS lines. |
| 20:41.23 | IngeniousTechsLL | Copper cabling ? |
| 20:41.36 | IngeniousTechsLL | Can anyone call me? |
| 20:41.51 | IngeniousTechsLL | I can explain miuch better pn the phone. |
| 20:42.09 | IngeniousTechsLL | 8048985679 |
| 20:42.13 | IngeniousTechsLL | PLEASE!!!!!!!!!!! |
| 20:42.51 | pabelanger | ~book |
| 20:42.52 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 20:43.11 | pabelanger | IngeniousTechsLL: ^ start here. You need a better starting point |
| 20:43.21 | pabelanger | it is all explained in the first few chapters |
| 20:44.03 | IngeniousTechsLL | I understand the basics |
| 20:44.04 | newtonr | IngeniousTechsLL: if i understood what you have said so far, you'll probably want a 4 FXO port TDM card in your FreePBX box to take in the 4 FXS lines from your Telco. Your SIP phones are going to talk over your network to Asterisk/FreePBX |
| 20:44.55 | newtonr | IngeniousTechsLL: if the telephony modem you are talking about is taking in analog lines and sending out SIP, then that just talks over ethernet to your Asterisk/FreePBX box as well.. Then you don't need any additional FXO ports anywhere |
| 20:45.12 | sekil | maybe the modem/ata already has some pbx/gw functionality |
| 20:45.16 | IngeniousTechsLL | So I need to get any 4 FXO card and put that where? I would like to use AWS EC2 with PIAF Green, already up and running, just need to take care of the hardware part. |
| 20:45.22 | *** join/#asterisk fkurkowski (~fkurkowsk@177.156.60.80) |
| 20:45.46 | pabelanger | If you need hardware, ec2 is not going to work for you |
| 20:45.50 | pabelanger | unless you get an ATA |
| 20:45.55 | pabelanger | which you already have it seems |
| 20:46.10 | pabelanger | so, why do you need to use FXS / FXO ports? |
| 20:46.16 | IngeniousTechsLL | The Arris that comcast bought me is http://www.speedguide.net/routers/arris-touchstone-tm604g-docsis-20-telephony-modem-1636 |
| 20:46.19 | pabelanger | and not just use SIP directly from an ITSP |
| 20:46.40 | newtonr | the 4 port FXO card goes in the PC/Server running whatever Asterisk implementation you use. |
| 20:46.55 | pabelanger | The TM604G delivers four lines of carrier-grade Voice over IP |
| 20:47.00 | pabelanger | who is your carrier? |
| 20:47.31 | IngeniousTechsLL | Comcast, and I have 4 regular POTS lines and Im guessing their digital because their using the moem...or maybe analog...not sure? |
| 20:47.32 | *** join/#asterisk tamiel (~tamiel@static-110-3.blueline.mg) |
| 20:47.43 | pabelanger | they are giving you VoIP |
| 20:47.56 | IngeniousTechsLL | Idk, are they? lol |
| 20:48.03 | pabelanger | that modem is doing the same thing asterisk would, with the FXO card you are looking to buy |
| 20:48.07 | IngeniousTechsLL | I am a IT tech for a compamy. |
| 20:48.08 | pabelanger | so, just by-pass the mode |
| 20:48.12 | pabelanger | modem* |
| 20:48.21 | pabelanger | and create a SIP trunk directing into asterisk |
| 20:48.40 | sekil | lol |
| 20:48.48 | IngeniousTechsLL | So either way I go will I hva to purchase 4 SIP trunks from a trunks provider? |
| 20:49.08 | IngeniousTechsLL | The Arris modem is only FXS |
| 20:49.14 | pabelanger | Yes, you will need to buy something from somebody if you want to connect asterisk to it |
| 20:49.27 | pabelanger | so, then you need FXO card in your asterisk box |
| 20:49.40 | IngeniousTechsLL | So Im not getting hoew this "IP PBX" is cheapoer than a regular pbx box? |
| 20:49.47 | drmessano | lol |
| 20:49.54 | v0lZy | lolz |
| 20:49.56 | pabelanger | who said it is |
| 20:50.20 | wrbishop | The OBi110 FXO/FXS interfaces are $50 each and work well. |
| 20:50.22 | IngeniousTechsLL | I dont get it, I am paying 100 for 4 lines plus 100 more for SIP trunks for each line? |
| 20:50.51 | pabelanger | you need to find a salesman to sell you some voip |
| 20:50.52 | newtonr | IngeniousTechsLL: pabelanger is saying get rid of the voip to analog device and replace it with just 4 pure sip trunks from your provider |
| 20:50.57 | pabelanger | ~itsp |
| 20:50.58 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 20:51.07 | pabelanger | IngeniousTechsLL: call one if these guys^ |
| 20:51.11 | wrbishop | Sorry, no experience with the Arris tech. |
| 20:51.11 | v0lZy | IngeniousTechsLL: Generally, its cheaper if u stick to VOIP only. Hardware wise computers are cheaper and more scalable compared to regular PBX boxes and voip calls usually cost less so the whole investment recycles itself much fater |
| 20:51.12 | drmessano | Its more like $100 for 4 lines or scrap them for $25/mo in SIP trunks |
| 20:51.35 | drmessano | Also without the need for a $400 card |
| 20:52.57 | wrbishop | I have some Voicetronix 4-line cards, but don't fit in my Raspberry Pi PBX ;-) |
| 20:53.08 | IngeniousTechsLL | 4 SIP phones, 4 offices, need to be able to transfer between phones and use the 4 lines on each phones. I have ethernet ports at each station. What do I need to do for the cheapest soltuion? |
| 20:53.24 | IngeniousTechsLL | wrbishop, will they work for me? |
| 20:53.31 | v0lZy | wrbishop: out of curiosity, any stress tests on the RPi? How many concurrent calls without transcoding? |
| 20:54.02 | wrbishop | The toughest part of the Pi install was configuring the OBi110; patience is always needed. |
| 20:54.05 | IngeniousTechsLL | This is a GC for Fort LEE, we wont be using the phones alot. So I can use a very small server. |
| 20:54.07 | v0lZy | IngeniousTechsLL: first you need to decide how many phone numbers you need and how many concurrent calls |
| 20:54.18 | v0lZy | (concurrent calls passing over the internet) |
| 20:54.25 | sekil | IngeniousTechsLL: start reading? |
| 20:54.28 | IngeniousTechsLL | I have 4 numbers, one will be used for fax for all stations. |
| 20:54.32 | wrbishop | I only have a few lines on the Pi, though they claim up to 10 lines; I have not "stressed" it yet, but so-far, so good. |
| 20:55.03 | v0lZy | IngeniousTechsLL: FAX and VoIP dont go well together, stick with classic telephony for fax or move to virtual fax services. |
| 20:55.05 | IngeniousTechsLL | Concurrent calls, not likely to happen but maybe once in a blue moon. |
| 20:55.16 | *** join/#asterisk killown (~killown@pdpc/supporter/student/killown) |
| 20:55.30 | v0lZy | IngeniousTechsLL: whats more important, outgoing or incoming calls? |
| 20:55.32 | killown | what's a better codec choice for voip? |
| 20:55.37 | IngeniousTechsLL | So I should use that line directly to a fax? |
| 20:55.48 | killown | I am using iLBC |
| 20:56.06 | IngeniousTechsLL | Hmm, neither really, I mean they barely use the phones here. |
| 20:56.07 | pabelanger | killown: all depends on bandwidth |
| 20:56.26 | IngeniousTechsLL | Probably both but on a very small dependecy. |
| 20:57.13 | killown | pabelanger, I have 2mbps, and it will have only two sip phone answering simultaneously |
| 20:57.35 | killown | I do not want to waste 64kb though |
| 20:59.16 | killown | I have only these codecs iLBC, GSM, g.729, g.711a e g.711u available in my voip service, what is a better choice? |
| 21:00.05 | sekil | killown: g729 |
| 21:08.32 | work | I cannot load module cdr_mysql ? |
| 21:09.22 | citywok | is the right version installed via addons (* 1.6 and lower) |
| 21:10.49 | work | no Asterisk 11.3.0 |
| 21:11.41 | navaismo | select the module when you do the make menuselect in section addons |
| 21:12.42 | work | what the name of modules |
| 21:14.42 | navaismo | its a joke right? |
| 21:15.18 | navaismo | I quote you: "<work> I cannot load module cdr_mysql ?" |
| 21:15.40 | work | yes |
| 21:20.05 | *** join/#asterisk fkurkowski (~fkurkowsk@177.156.60.80) |
| 21:26.19 | *** join/#asterisk b0ot (~DynamicFa@147.177.61.218) |
| 21:26.30 | b0ot | Anyone know of some quality cheap cellular gateways? |
| 21:30.20 | work | <navaismo> please which module name in menuselct ? |
| 21:33.42 | navaismo | ¬¬ what module you cant load? |
| 21:35.04 | work | cdr_mysql |
| 21:36.36 | navaismo | ok now go to make menuselect and check it, its under ADDONS section |
| 22:01.59 | WIMPy | When do we get an equalizer? |
| 22:06.28 | drmessano | ~happyclownpbx |
| 22:06.28 | infobot | [HappyClownPBX] is currently in closed beta, approaching 12GB in size, uses Asterisk for its core, it pwns, is also now compatible with the Diahatsumashiniriki Keyotason 200LP-A11 SIP phone |
| 22:07.05 | eirirs | 1april... |
| 22:07.10 | eirirs | :P |
| 22:08.04 | work | make[2]: *** No rule to make target `../main/modules.link', needed by `asterisk'. Stop. |
| 22:08.05 | work | make[1]: *** [main] Error 2 |
| 22:08.05 | work | make[1]: Leaving directory `/usr/src/asterisk-11.3.0' |
| 22:08.05 | work | make: *** [_cleantest_all] Error 2 |
| 22:08.43 | work | this happened during recompie asterisk |
| 22:14.48 | edong23 | work: i just got home from work. i had 0 issues installing dahdi/libpri/asterisk and then sangoma and it all worked flawlessly. so lets talk about your setup |
| 22:14.52 | edong23 | what distro? |
| 22:15.59 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 22:16.19 | work | ubuntu +degium te405p + asterisk + e1 from PSTN |
| 22:16.44 | work | my aim to collect all csr and recorde all calls |
| 22:16.55 | edong23 | and you have installed the list of dependancies first, correct? |
| 22:17.09 | edong23 | collect "cdr"? |
| 22:17.14 | edong23 | and record ALL calls? |
| 22:17.19 | work | yes |
| 22:18.28 | work | it was problems in cdr_mysql mofules and now I am recompiling asterisk |
| 22:19.36 | work | my project to receive E1 with pri_cpe |
| 22:20.16 | work | calls forwarded to my server in order to archive them in database wuth there cdr |
| 22:21.04 | work | my problem now in cd-mysql modules |
| 22:21.43 | [TK]D-Fender | ODBC is the current way |
| 22:21.56 | work | how ? |
| 22:22.27 | edong23 | work, do you have mysql-dev installed |
| 22:22.37 | edong23 | or whatever its called in the *buntus? |
| 22:25.06 | *** part/#asterisk sekil (~Ognjen@78.24.104.82) |
| 22:30.29 | *** join/#asterisk ectospasm (~ectospasm@unaffiliated/ectospasm) |
| 22:31.20 | work | <edong23> no I have not it |
| 22:32.15 | edong23 | isnt that what will cause that error? |
| 22:32.27 | edong23 | it doesnt have the development files |
| 22:32.35 | edong23 | i think this is true, but im not sure |
| 22:33.03 | work | cdr show status |
| 22:33.24 | work | only I have |
| 22:33.26 | work | <PROTECTED> |
| 22:33.26 | work | <PROTECTED> |
| 22:33.48 | work | I need mysql? how ? |
| 22:37.24 | navaismo | argh |
| 22:37.25 | [TK]D-Fender | ~book |
| 22:37.26 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 22:37.34 | [TK]D-Fender | work: Read. The. BOOK. |
| 22:37.38 | edong23 | yah |
| 22:37.45 | navaismo | again you need to select from the modules below addons section |
| 22:37.55 | [TK]D-Fender | work: A lot of good people did a lot of hard work to write that book which you can read online for FREE. |
| 22:40.17 | work | ok I agree with you |
| 22:41.18 | work | it needed especially I a, new in this but for now I am just need a hint to collect my cdr in database |
| 22:41.47 | edong23 | work: you are unable to complete a build that works.... you need more than a hint |
| 22:41.56 | edong23 | i mean no offense at all |
| 22:42.17 | edong23 | you are undertaking a task that for someone new, can seem daunting, but that book is well written |
| 22:42.17 | work | ): |
| 22:42.51 | [TK]D-Fender | work: Everything is "just a hint for you". |
| 22:42.55 | WIMPy | wonders if it makes sense to use Asterisk to collect CDR data. |
| 22:43.23 | [TK]D-Fender | work: You're starting from scratch.... |
| 22:44.02 | edong23 | work, things will go much smoother forever if you get a more than "i got it working" understanding of the technology you are using |
| 22:44.12 | edong23 | that book can really really help |
| 22:44.18 | *** join/#asterisk n3hxs (~ed@pool-108-16-94-10.phlapa.fios.verizon.net) |
| 22:44.31 | edong23 | youll find out that many of your questions will just go away |
| 22:44.43 | work | ok man |
| 22:44.48 | work | ok men |
| 22:45.15 | work | I am fonna to read in spit of my english is not good |
| 22:50.51 | edong23 | work, what language is your native language? |
| 22:51.42 | work | if I tell you you will cik me out |
| 22:52.44 | work | kick |
| 22:53.04 | work | any way it is ARABIC |
| 22:56.56 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-ljopvbknzuihjzrx) |
| 23:01.13 | navaismo | always be proud of your nation |
| 23:02.15 | newtonr | work: there is a lot of international representation in here :) |
| 23:03.13 | work | oooh man which nation poeple kill others like wolfs it is very sham |
| 23:03.46 | work | we are like Erou in medetrian age |
| 23:05.19 | work | it is look like multy carbage source code without compiler ): --(: |
| 23:05.34 | ectospasm | wat? |
| 23:06.04 | work | it ia a jok about arab do not care |
| 23:15.19 | *** join/#asterisk blee (~blee@adsl-74-162-89-48.mco.bellsouth.net) |
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| 23:43.06 | *** join/#asterisk Minotaur01 (~minotaur0@S01060018e7f9c7df.hm.shawcable.net) |
| 23:43.47 | *** join/#asterisk xibalba (~xibalba@unaffiliated/xibalba) |
| 23:46.13 | *** join/#asterisk HolTech (~IceChat77@pool-74-107-104-2.bltmmd.fios.verizon.net) |
| 23:47.30 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 23:49.37 | *** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn) |
| 23:49.46 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
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