| 00:00.08 | kazaa_lite | ummm... i was just curious whether they have embraced the value of sofswitches or not |
| 00:00.11 | bcrisp | instead of waiting in the queue, the caller can record a msg and hangup |
| 00:00.46 | bcrisp | dlynes: the msg stays is played to the agent once their turn comes up |
| 00:00.54 | *** join/#asterisk dkirker-openmobl (n=dkirker@adsl-75-15-141-6.dsl.snlo01.sbcglobal.net) |
| 00:01.03 | dlynes | bcrisp: that should be possible using dialplan in asterisk 1.6.2 (originate() is a dialplan application in 1.6.2) |
| 00:01.48 | bcrisp | ok orignate was only in AMI as far as i know in 1.6.1.11 |
| 00:02.23 | [TK]D-Fender | Or you could use CLI. Or call-files... or... whatever |
| 00:02.33 | bcrisp | right call files was my first attempt |
| 00:02.54 | bcrisp | which i may retry if this h extension thing doesnt work |
| 00:05.27 | bcrisp | [TK]D-Fender, what would be the effect of calling Queue() within the h extension? |
| 00:05.32 | bcrisp | world destruction? |
| 00:05.56 | [TK]D-Fender | bcrisp: Whats the point? |
| 00:06.17 | bcrisp | i think im trying to replicate originate() |
| 00:06.28 | [TK]D-Fender | bcrisp: huh? |
| 00:06.35 | bcrisp | user records message and hangs up |
| 00:06.41 | bcrisp | want a call to go back into the queue |
| 00:06.45 | jblack | avatar _rocked_ |
| 00:06.49 | bcrisp | when its answered, have the playback of the message to the agent |
| 00:07.03 | bcrisp | via the queuemacro |
| 00:07.14 | Katty | jblack: orly? |
| 00:07.25 | [TK]D-Fender | bcrisp: Doesn't work. You'll have to originate |
| 00:07.36 | bcrisp | so i should upgrade to 1.6.2 i suppose |
| 00:07.47 | jblack | Yeah. See this one. And see it in 3d. |
| 00:08.10 | Katty | 3d? ^_- |
| 00:08.38 | Katty | i didn't know it came out in 3d |
| 00:08.45 | jblack | 3d these days isn't the 3d from 20 years ago. Nowadays they use polarized lenses instead of red/green |
| 00:09.14 | TSM | or they use blanking |
| 00:09.34 | [TK]D-Fender | bcrisp: I wouldn't use 1.6.2 yet |
| 00:09.41 | [TK]D-Fender | bcrisp: Certainly no need |
| 00:09.52 | jblack | The 3d movies I have been to all used polarized lenses. |
| 00:09.59 | drmessano | But but |
| 00:10.21 | drmessano | We dont need Dahdi anymore |
| 00:10.23 | drmessano | YAY |
| 00:10.34 | drmessano | DING DONG THE WITCH IS DEAD |
| 00:10.39 | drmessano | THE WITCH OH WITCH |
| 00:10.41 | TSM | drm: why? |
| 00:10.45 | drmessano | THE WITCH IS DEAD |
| 00:11.14 | jblack | who killed your dahdi? |
| 00:11.17 | drmessano | Well, if you didnt have TDM/Analog cards, only needed DAHDI for meetme mixing |
| 00:11.28 | drmessano | Now we dont need it for that anymore |
| 00:11.32 | drmessano | DING DONG THE WITCH IS DEAD |
| 00:12.03 | TSM | drm: oneday mabey sangoma will be able to work without dahdi |
| 00:12.29 | drmessano | Doesnt bother me any |
| 00:12.44 | drmessano | Analog = Balls |
| 00:12.45 | Katty | i wonder how much sangoma has sponsered the development of the asterisk project |
| 00:13.09 | TSM | Katty: dont know, but they seem to be a major player |
| 00:13.30 | drmessano | They do? |
| 00:14.24 | Katty | something to at least thank about |
| 00:14.31 | Katty | and think (= |
| 00:15.16 | TSM | i like their gurantee that they will 100% gurantee to make their card work with your system and asterisk/callwever/etc.. |
| 00:15.29 | bcrisp | [TK]D-Fender, ok, originate in 1.6.1.11 ... researching |
| 00:15.43 | Katty | they are very helpful. i've called them a few times. |
| 00:16.26 | TSM | who do you speak to, marc, i think he is the only one |
| 00:16.34 | Katty | not sure. |
| 00:30.43 | bcrisp | [TK]D-Fender, if i use System(asterisk -rx \"originate .......... within my dial plan, will the new call be able to read channel variables or _, __ channel variables? |
| 00:31.25 | bcrisp | basically i need a way to persist variable information between the dialplan, originate, and the target context |
| 00:32.04 | [TK]D-Fender | bcrisp: then use call files / AMI originate |
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| 00:32.24 | bcrisp | reading more into call files |
| 00:35.06 | LemensTS | what is the vmgain default value? |
| 00:36.43 | *** join/#asterisk jebba (n=jebba@host150.190-231-143.telecom.net.ar) |
| 00:37.18 | jebba | Connected to Asterisk 1.6.2.0-2 currently running on Nokia-N900-42-11 (pid = 1692) |
| 00:46.17 | *** join/#asterisk friendly12345 (n=friendly@ppp118-209-31-140.lns20.mel4.internode.on.net) |
| 00:55.12 | [TK]D-Fender | LemensTS: ) <- |
| 00:55.14 | [TK]D-Fender | 0 |
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| 01:06.28 | *** part/#asterisk ruben23 (n=AGENT@122.55.48.243) |
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| 01:14.53 | *** part/#asterisk ruied (n=ruied@bl9-252-182.dsl.telepac.pt) |
| 01:17.38 | bcrisp | [TK]D-Fender, im so close (i think) |
| 01:20.15 | LemensTS | TK: thanks |
| 01:24.12 | LemensTS | TKD-Fender: vmgain was "", i changed it to "0" and it is still complaining... -- Executing [s@macro-vm:2] Set("SIP/voipstreet_in-08368bb0", "VMGAIN=g(0)") in new stack |
| 01:24.12 | LemensTS | [Dec 19 19:22:45] WARNING[9717]: func_db.c:58 function_db_read: DB requires an argument, DB(<family>/<key>) |
| 01:25.32 | friendly12345 | Hi there. What's the best way of sending SNMP traps when a variable changes? At the moment the most elegant method seems to be an app which connects to the asterisk manager and then parses events |
| 01:39.43 | TSM | if i had a redundant system with DRDB between two nodes, would this be good enough to keep even the astdb and all the config files in sync or is there any issues with asterisk |
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| 02:34.07 | mchou | Is there a way to obtain rtp latency stats in asterisk? |
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| 03:55.35 | Mango | mchou: ${SIPPEER(peer,status)} perhaps? |
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| 04:11.39 | comradeb14ck | sup everyone |
| 04:12.11 | Mango | waves |
| 04:13.25 | Mango | I saw a video demonstrating some software that works with Asterisk that allows you to control the PBX using an ajax app. You could do things like see the calls in a queue, drag and drop them to any agent, drag and drop calls into a conference, etc. |
| 04:13.29 | Mango | Anyone know the name of it? |
| 04:13.39 | comradeb14ck | there's a few |
| 04:13.44 | comradeb14ck | fop is probably the one you saw |
| 04:13.47 | comradeb14ck | 'flash operator panel' |
| 04:14.00 | comradeb14ck | see: http://fop2.com |
| 04:15.39 | Mango | Nifty; thanks :) That wasn't the one I saw before but it looks cool. |
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| 04:59.45 | p3nguin | I've got a Linksys SPA-3102, and I'm wondering if anyone knows the setting to make it where any call that comes in on the PSTN can automatically be routed out to the Ethernet/SIP. |
| 05:13.48 | ChannelZ | yeah |
| 05:14.06 | ChannelZ | search the forums on asterisk.org there's a write-up about it. Too many steps to list here |
| 05:16.21 | p3nguin | You don't happen to have the URL for it, do you? |
| 05:17.08 | p3nguin | I've followed some steps I found, but the device never picks up the call... so that's why I was looking for someone that is familiar with it. |
| 05:18.49 | ChannelZ | http://lmgtfy.com/?q=linksys+spa-3102+%22asterisk+configuration+howto%22 |
| 05:20.59 | bcrisp | i have another one of those questions where i cant find the right words |
| 05:22.22 | bcrisp | i need a call file to call into a queue. i'm not sure what channel i should set up. I supply Queue as the application with Data being the queue name |
| 05:23.48 | bcrisp | using Local/exten@context doesnt seem to work |
| 05:27.14 | ChannelZ | p3nguin: once configured right it looks like any other sip device - when my phone rings, it comes into the dialplan |
| 05:27.32 | p3nguin | I think I figured out what I was missing, but I have to test it still. |
| 05:28.51 | ChannelZ | probably the line wasn't enabled or the 'VoIP-To-PSTN Gateway' wasn't on |
| 05:29.28 | ChannelZ | (or the other way around - or both :) |
| 05:29.53 | p3nguin | Still isn't working. |
| 05:30.18 | ChannelZ | so you have it registering to your * under 'PSTN Line' ? |
| 05:30.49 | p3nguin | I thought it was the dial plan setting that was keeping it from working, but it didn't help when I adjusted it. |
| 05:31.38 | ChannelZ | So when the phone rings you just never see anything on the * side at all? |
| 05:31.45 | p3nguin | The device has been in service for a while acting as a "splitter," where if you pick up the phone (which is connected to line 1), you can dial out through * or the PSTN depending on what pattern you dial. |
| 05:32.24 | p3nguin | It has also been configured to ring through to line 1, so that it works like a normal passthrough device so the phone can be answered when a PSTN call comes in. |
| 05:33.08 | p3nguin | So I think the problem is that I am registering line 1 to * rather than registering PSTN Line to * |
| 05:33.13 | *** join/#asterisk dlynes (n=dlynes@bas5-hamilton14-1096762521.dsl.bell.ca) |
| 05:33.32 | dlynes | Is dahdi still in release candidate mode? |
| 05:33.40 | dlynes | I thought it was a final release product? |
| 05:34.08 | p3nguin | I was hoping it wouldn't be so difficult to change the operations of it, because this is only going to be a temporary configuration and then it will need to go back to the way it has been all along. |
| 05:34.11 | ChannelZ | p3nguin: You can do both |
| 05:34.53 | p3nguin | I figured I could. I just have to do some additional configuration on both sides of things. I was expecting less work, but I'll dig in in just a moment. |
| 05:34.53 | ChannelZ | I have one registering on port 5060 as one user and the other on 5061 as another user, though I'm not sure they necessarily even need to be separate port numbers |
| 05:34.58 | Corydon76-dig | dlynes: we have release candidates prior to making full version releases |
| 05:35.28 | Corydon76-dig | dlynes: so, yes, there are full releases, but the next release is in candidate status right now |
| 05:36.53 | dlynes | Corydon76-dig, ah...where are the final releases then? I only see release candidates on downloads.digium.com |
| 05:37.15 | Corydon76-dig | Look in the releases directory |
| 05:37.36 | p3nguin | channelz: So you think I can use 5060 on both and everything will be okay? |
| 05:37.58 | ChannelZ | I just reset mine and they're both registering OK.. not sure if it works though |
| 05:38.09 | ChannelZ | I don't actually have a handset hooked up to mine, just the line |
| 05:38.15 | dlynes | Corydon76-dig, i.e. downloads.digium.com/pub/telephony/dahdi-linux-complete/releases |
| 05:38.29 | dlynes | Corydon76-dig, and the directory above that is all release candidates also |
| 05:38.44 | Corydon76-dig | dlynes: correct |
| 05:38.56 | ChannelZ | I'm not sure if it will get confused about the traffic having 2 things using the same UDP port number on a single device |
| 05:39.06 | dlynes | Corydon76-dig, so even the releases folder is all release candidates...no final releases |
| 05:39.53 | Corydon76-dig | dlynes: look closer |
| 05:39.58 | Corydon76-dig | http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete/releases/ |
| 05:40.20 | Corydon76-dig | http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete/releases/dahdi-linux-complete-2.2.0.2+2.2.0.tar.gz |
| 05:40.37 | dlynes | Corydon76-dig, yeah...nvm...one or two files in there are releases...everything else is rc's |
| 05:40.41 | dlynes | Corydon76-dig, thanks |
| 05:40.59 | Corydon76-dig | Yes, there are many more release candidates than actual releases |
| 05:41.01 | dlynes | Corydon76-dig, i was going cross-eyed from all the rc spam |
| 05:41.40 | dlynes | Corydon76-dig, I guess there's no libpri for asterisk 1.6, either? |
| 05:41.54 | dlynes | Corydon76-dig, or do you use libpri-1.4.10.2 for 1.6? |
| 05:42.31 | Corydon76-dig | Correct |
| 05:42.54 | dlynes | Corydon76-dig, thanks |
| 05:42.55 | Corydon76-dig | We don't bump version numbers for libpri just to make them match anymore |
| 05:43.26 | m0e | does anyone have any pointers to docs/hw sites for simboxes? |
| 05:43.34 | dlynes | Corydon76-dig, ah...just haven't had to do any pri work with asterisk for quite a while |
| 05:50.21 | p3nguin | Finally, I got it to work. |
| 06:00.14 | dlynes | m0e, simboxes? |
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| 06:11.10 | *** part/#asterisk Tech_Travis (n=Travis@mail.techglia.com) |
| 06:12.09 | Mango | Huh |
| 06:12.15 | Mango | For some reason I can't get the echo test working. |
| 06:12.25 | Mango | exten => s,1,Answer |
| 06:12.26 | Mango | exten => s,n,Echo |
| 06:13.14 | Mango | The CLI says it's starting the echo test, but no RTP data happens. |
| 06:14.06 | dlynes | You have to create the rtp data from the calling end |
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| 06:14.21 | dlynes | Mango, It needs something to echo back |
| 06:14.48 | Mango | for some reason it is not receiving it. |
| 06:15.11 | Mango | It works on the local network but not over the PSTN. |
| 06:15.14 | Mango | scratches his head. |
| 06:16.51 | dlynes | Mango, you mean it works for sip, but not for dahdi? |
| 06:17.43 | Mango | Sorry, I did not explain that well. It works with SIP over the local network but not when I call with a regular phone, to a SIP trunk that the Asterisk is registered to. |
| 06:19.12 | dlynes | Mango, ah |
| 06:20.25 | dlynes | ah...no idea offhand, then |
| 06:20.46 | dlynes | and i've gotta head to bed, so I can't really help you much with it tonight |
| 06:20.58 | Mango | Thanks anyway :) |
| 06:25.14 | m0e | dlynes, something like an fxo but with a cellular sim |
| 06:25.28 | p3nguin | This is ridiculous. I changed the settings back, and now the phone can't dial out the PSTN. It's like there's no matching dial pattern, but I didn't even touch the dialplans. |
| 06:27.17 | ChannelZ | did you accidentally set it to the wrong one/ |
| 06:27.18 | ChannelZ | ? |
| 06:27.53 | dlynes | m0e, yeah...for those things, I think the only docs are the ones that come with them |
| 06:28.00 | dlynes | m0e, I've tried looking around, myself |
| 06:29.34 | m0e | ok cool.. thanks :) |
| 06:36.00 | p3nguin | This is retarded. The fucking 3102 will no longer make calls out the PSTN even with PSTN fallback enabled. |
| 06:41.46 | ChannelZ | what does it do |
| 06:47.30 | m0e | dlynes, i found this http://www.voptech.com/product_view.asp?tid=47&id=169 |
| 06:47.40 | m0e | seems like what we are looking for |
| 06:48.27 | p3nguin | channelz: If I dial a number for internal phones, it hits * correctly. If I dial a number that is to be routed over the PSTN, it give a fast busy signal. |
| 06:54.26 | p3nguin | It acts like I have a bad dialplan on line 1, but I didn't touch the outgoing dialplan at all. |
| 06:55.46 | p3nguin | I configured the SIP stuff on PSTN Line so that I could enable the PSTN-to-VoIP gateway, where incoming calls would be routed to * instead of rining the phone. It worked fine. Then I undid those things and now it can't dial OUT. How are these related? |
| 06:56.11 | p3nguin | I can't figure it out, so I'm probably going to have to try to find a reset for the entire device and reconfigure it all tomorrow. |
| 06:57.09 | ChannelZ | you turned off the PSTN-to-Voip gateway? |
| 06:57.13 | p3nguin | yes |
| 06:57.33 | ChannelZ | but it's still sending calls to * |
| 06:57.45 | p3nguin | But even if I left it on, that shouldn't affect the calls that go from line 1 to gw0. |
| 06:58.12 | p3nguin | The dialplan is configured to send some calls to * and others out gw0. |
| 06:58.36 | p3nguin | 9202, for example, is routed to * correctly. |
| 06:58.52 | p3nguin | 18004444444 should go out gw0, but instead gives me a fast busy. |
| 06:59.36 | p3nguin | It's like I lost the ability to use the PSTN Line for OUTGOING calls. |
| 06:59.58 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 06:59.59 | p3nguin | This is the first time I ever enabled the PSTN-to-VoIP gateway. |
| 07:00.20 | p3nguin | first time I even tried to configure it, because I never needed it before. |
| 07:01.10 | p3nguin | But now that I have configured it once and got it to work, I unconfigured it and put it back like it was before. |
| 07:02.05 | p3nguin | And now it's 1AM and I'm really irritated with it. There is no reasonable explanation that I can come up with as to why it can't call OUT the PSTN anymore. |
| 07:02.45 | p3nguin | If I unplug the power to the device, the phone calls out the PSTN like it's supposed to. |
| 07:02.58 | p3nguin | ALL CALLS go out the PSTN with no power on it, though. |
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| 07:28.37 | friendly12345 | Hi there. What's the best way of sending SNMP traps when a variable changes? At the moment the most elegant method seems to be an app which connects to the asterisk manager and then parses events |
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| 08:43.50 | trumee | guys, i am upgraded from * 1.4 to 1.6. Now i am unable to load thedigium G729a codec. |
| 08:44.03 | trumee | <PROTECTED> |
| 08:44.13 | trumee | loader.c: Module 'codec_g729a.so' could not be loaded |
| 08:44.31 | trumee | is what i get on asterisk -rx "module load codec_g729a.so" |
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| 08:47.06 | trumee | uh oh, i need the 1.6 version. sorted! |
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| 09:02.34 | DelphiWorld | hi all |
| 09:02.39 | DelphiWorld | anyone in didx? |
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| 09:34.34 | devyll | hello all. Can you please advise on the best solution for having remote queue members logged in queues ? Usually I'm using sip phones connected dirrectly to the asterisk machine, but for the next 2 weeks I will need to have mobile phone connected in the queues. Somehow they will not be able to be connected via sip to the asterisk machine but I will still need them to act as logged in sip users. |
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| 10:35.44 | ChannelZ | so are 1.6.0, 1.6.1 and 1.6.2 all separate branches? |
| 10:45.50 | mbrevda | yup |
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| 12:31.20 | steve745 | can anyone tell me how to use bridge() in a dialplan correctly |
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| 13:04.38 | gavimobile | folks, what I believe I have a working asterisk install which I would like to start setting up and testing. is there a way I can start testing my asterisk setup without buying anything |
| 13:22.03 | *** join/#asterisk angryuser (n=angryuse@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
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| 13:33.07 | Gugge | gavimobile, that really depends what you have now |
| 13:33.22 | Gugge | you would need a pc with a netcard to run asterisk |
| 13:35.17 | gavimobile | Gugge: im guessing that's not the same as a network card |
| 13:38.04 | gavimobile | Gugge: I would like to set up something so basic just to test with. |
| 13:40.58 | Gugge | yes, i ment a nic |
| 13:41.24 | Gugge | but im guessing you have a pc, since you are inhere .... offcause that could be with a cellphone or something |
| 13:42.15 | Gugge | for me to know if you need to buy something you would have to tell me what you want to test, and what you allready have |
| 13:48.44 | gavimobile | Gugge: I have a pc with a nic, 56k v90 modem and asterisknow install |
| 13:49.58 | gavimobile | and I would like to test by calling myself and listening to an "ivr" that I will create |
| 13:58.36 | Gugge | then you would either need a hardware phone, a softphone, or a phoneline |
| 13:58.50 | gavimobile | I have a phoneline |
| 13:58.59 | gavimobile | not sure what a softphone is |
| 13:59.13 | gavimobile | and I have a telephone of course |
| 14:01.31 | *** join/#asterisk voipmonk (n=shido6@66.49.128.103) |
| 14:02.24 | Gugge | you would need some FXO hardware to connect the asterisk box to the PSTN |
| 14:02.37 | *** join/#asterisk Carlos_Tico (n=grillo@c-98-201-56-25.hsd1.tx.comcast.net) |
| 14:05.51 | gavimobile | what can I do without purchasing anything else from what I have |
| 14:08.49 | voipmonk | what do you have? I came in late :) |
| 14:10.22 | wonderworld | too - much - S N O W ! |
| 14:12.01 | HaMF | hehe |
| 14:13.14 | gavimobile | voipmonk: I have a pc with a nic, 56k v90 modem and asterisknow installed, a phoneline and a phone |
| 14:13.34 | voipmonk | and what would you like to do gavimobile ? |
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| 14:25.37 | gavimobile | I would like to start by creating an answering machine or a voicemail |
| 14:25.43 | gavimobile | or something along thoese levels |
| 14:27.01 | gavimobile | voipmonk: * |
| 14:27.59 | voipmonk | you'll need to buy an fxo card or a sip did |
| 14:28.20 | *** part/#asterisk cracoucas83 (n=lynxblan@210.174.192-77.rev.gaoland.net) |
| 14:28.25 | gavimobile | aren't there free sipdid;s? |
| 14:28.49 | gavimobile | voipmonk: * |
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| 14:29.02 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 14:29.57 | gavimobile | voipmonk: I signed up to a website called ipcomms.net.. they gave me a sip host username and password, also some trunk config... |
| 14:30.15 | gavimobile | they even gave me a phone number |
| 14:30.48 | voipmonk | great |
| 14:30.52 | voipmonk | you're 1/2 way there |
| 14:31.01 | voipmonk | have you read the asterisk book? |
| 14:31.27 | gavimobile | does this sound normal, that companies give free did's? |
| 14:31.40 | voipmonk | yes , because when ppl call them they get paid |
| 14:31.44 | gavimobile | I am not sure what I've been reading |
| 14:32.18 | gavimobile | but I have been reading up online.. |
| 14:36.41 | TSM | ~book |
| 14:36.43 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 14:41.26 | *** join/#asterisk mpe (n=mpe@0x4dd624b2.adsl.cybercity.dk) |
| 14:47.32 | dlynes | voipmonk, how do you charge other people/companies for calling your dids? |
| 14:48.05 | voipmonk | its a CLEC thing |
| 14:48.15 | voipmonk | I will rant about it on my blog shortly |
| 14:48.17 | dlynes | voipmonk, We are a CLEC |
| 14:48.24 | voipmonk | I will have jeremy rant about it , rather |
| 14:48.57 | dlynes | voipmonk, where is your blog, then? |
| 14:49.38 | dlynes | voipmonk, also, I'm guessing you need to be running ss7 or something to take advantage of this? |
| 14:50.08 | dlynes | voipmonk, or are pris sufficient? |
| 14:58.04 | TSM | does us have sdh, i would have thought a clec is connected like that |
| 14:58.26 | TSM | ~sdh |
| 15:11.28 | bcrisp | mornin |
| 15:14.54 | *** part/#asterisk m0e (n=m0e@41.196.154.175) |
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| 15:48.53 | bcrisp | grrrrr |
| 15:50.44 | bcrisp | i try setting variables prior to calling Queue() - once the call is connected those variables are gone |
| 15:51.25 | [TK]D-Fender | brcShould follow standard inheritence rules. |
| 15:51.54 | bcrisp | hm |
| 15:54.43 | bcrisp | [TK]D-Fender, http://pastebin.ca/1721380 |
| 15:56.28 | [TK]D-Fender | bcrisp: Go read CHANNELVARIABLES doc again. There is another format to force inheritence for it. |
| 15:56.37 | bcrisp | ok thank you will do |
| 15:57.00 | [TK]D-Fender | Other optiosn : mod callid as "key" |
| 15:57.04 | [TK]D-Fender | callerid* |
| 15:57.10 | bcrisp | oh interesting |
| 16:01.27 | *** join/#asterisk Alagar (n=Administ@122.164.103.31) |
| 16:03.24 | bcrisp | so besides bla, _bla, and __bla , do you mean using the SHARED function? |
| 16:07.59 | bcrisp | shared worked |
| 16:08.17 | bcrisp | although i think it may become a problem in my scenario |
| 16:13.00 | *** join/#asterisk hwt (i=hwt@curb.thorkildssen.com) |
| 16:13.35 | hwt | ahoy. is there a way to give the user a second dialtone (using playtones?), and stop the tones when the user starts typing with dtmf? |
| 16:13.51 | hwt | and when input is finished, go to extension. |
| 16:17.35 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 16:17.40 | [TK]D-Fender | hwt: Go make a recording of dialtone andBackground() it in an IVR |
| 16:18.54 | hwt | allright, but I may also use the DISA application, can't I? this is a two-stage dialing system. |
| 16:21.00 | [TK]D-Fender | hwt: Yes, but that creates extra CDR's, doesn't have a way to compensate illegal matches, etc |
| 16:21.13 | [TK]D-Fender | hwt: Also crap for controlling timeouts |
| 16:21.21 | hwt | [TK]D-Fender: i'm not doing CDRs on Asterisk. |
| 16:23.58 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 16:28.38 | *** join/#asterisk voipmonk_ (n=shido6@67.204.37.228) |
| 16:33.16 | voipmonk_ | password protect an outbound route from an inbound route like a calling card... bbl |
| 16:33.22 | *** join/#asterisk rethus (n=contio@p50876D51.dip.t-dialin.net) |
| 16:33.42 | voipmonk_ | using and extension as a password.... (hax0r hax0r hax0r) |
| 16:33.52 | voipmonk_ | and a Goto |
| 16:34.03 | rethus | how can i optimize my asterisk-upstream? |
| 16:34.18 | rethus | now i can only handle 6 persons with a 6 Mbil-DSL |
| 16:36.31 | bcrisp | [TK]D-Fender, awesome, modifying the callerid worked! |
| 16:39.39 | *** join/#asterisk ttl- (n=patrick@d5153A420.access.telenet.be) |
| 16:44.32 | [TK]D-Fender | rethus: What is there to "optimise"? |
| 16:45.24 | rethus | i have 6 Mbit dsl, and ony can handle 5 till 6 listener,,, i wan't to have the best sound quality for listener but so many listener handling as possible |
| 16:45.46 | rethus | now i have only G726 allow |
| 16:46.01 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 16:46.52 | [TK]D-Fender | rethus: "6 Mbit" isn't the complete number... |
| 16:47.08 | Katty | morning |
| 16:47.13 | rethus | number of what? |
| 16:47.17 | [TK]D-Fender | Katty: Mew. |
| 16:47.23 | [TK]D-Fender | rethus: that is HALF of the picture |
| 16:47.31 | rethus | i don't understand |
| 16:47.45 | Katty | 5 squirrels in the yard this morning (= |
| 16:47.51 | [TK]D-Fender | rethus: Of course now, because you are only thinking in ONE DIRECTION |
| 16:48.27 | Katty | one direction? TOWARDS LUNCH |
| 16:48.40 | rethus | what u try to say me? |
| 16:49.05 | Katty | maybe he doesn't understand english very well |
| 16:49.11 | Katty | rethus: what is your primary language? |
| 16:50.46 | Katty | ooh, 6 squirrels now |
| 16:50.55 | Katty | 7! |
| 16:51.44 | [TK]D-Fender | rethus: 6mbit is only your DOWNLOAD. WAKE UP. |
| 16:52.12 | [TK]D-Fender | rethus: You are limited by your UPSTREAM as well. |
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| 16:54.10 | *** join/#asterisk RobH (n=RobH@cpe-173-169-30-118.tampabay.res.rr.com) |
| 16:55.08 | voipmonk | rethus: go to http://www.pingtest.net/ and tell me what you get |
| 16:55.18 | voipmonk | click on begin test |
| 16:56.24 | voipmonk | then go to www.speedtest.net and begin the test and tell me wha tu gt |
| 16:57.37 | voipmonk | it will tell you an upload speed - whats it say? |
| 16:58.09 | voipmonk | or click on copy direct line and paste it here :) |
| 16:58.25 | voipmonk | direct link |
| 16:58.28 | voipmonk | i cant read nor type today |
| 16:58.44 | voipmonk | 91% keyboard... 91% efficiency :) |
| 16:59.09 | voipmonk | pets his 1035nr running snow leopard... its ok... I still wuv u |
| 17:02.01 | rethus | [TK]D-Fender: i know, that was not my question |
| 17:02.27 | rethus | i know, only 10% is upstream |
| 17:02.36 | [TK]D-Fender | rethus: You aren't showing everything that is involved and wondering why you have quality issues |
| 17:02.36 | rethus | but please reat my question exactly |
| 17:03.08 | [TK]D-Fender | rethus: You also didn't mention ptorocols, and where those calls are going to exactly. |
| 17:03.20 | rethus | i only ask for tunning-tips to handle as much connection as possible with &Mbit-DSL |
| 17:03.43 | *** join/#asterisk ecto (n=weechat@user-24-236-95-118.knology.net) |
| 17:03.46 | [TK]D-Fender | rethus: The 6mib isn't what is screwing you. |
| 17:03.55 | rethus | all calls coming form phones (ISDN, analoge) |
| 17:04.09 | [TK]D-Fender | rethus: What don't you get? 600 mBIT would be enough for your 6 calls. |
| 17:04.41 | rethus | You don't understand me.. please step back and let me explain... forget all postings before |
| 17:05.49 | rethus | i have 6 Mbit and want to connect many many many connections. Now i still can handle 6 connection, cause the upstream put me down. I use G726 as codec and dissallow all other codecs. My Question |
| 17:06.22 | rethus | Are3 there some Asterisk-Tricks which make me able to handle more connections, but with the same qualitiy of audio-signal? |
| 17:06.27 | rethus | (or better) |
| 17:06.38 | *** join/#asterisk nighty^ (n=nighty@tin51-1-82-226-147-104.fbx.proxad.net) |
| 17:06.53 | [TK]D-Fender | rethus: IAX2 Trunk Mode |
| 17:07.14 | [TK]D-Fender | rethus: And you still didn't answer my question about what PROTOCOLS, and WHERE the calls are going. |
| 17:07.17 | *** join/#asterisk manxpower (n=ewieling@35.sub-70-223-46.myvzw.com) |
| 17:07.56 | rethus | Protocols: G726 (no ulaw, no gsm and so on) or what you mean with protocols? |
| 17:08.39 | [TK]D-Fender | rethus: that is a CODEC |
| 17:09.00 | *** join/#asterisk doolittlework (n=d@196.211.34.2) |
| 17:09.02 | rethus | can u give me an example? |
| 17:10.19 | rethus | i have asterisk running on my server... and one Talker is connected via "asterisk dial 100". The signal is connected to sipgate, where i have a phonenumber... all listeners connecting with ther Phones (no softphones) to this number (Thats my architecture) |
| 17:10.45 | [TK]D-Fender | rethus: SIP, IAX2, MGCP, H.323 are PROTOCOLS. |
| 17:11.20 | rethus | ok, i only have sip |
| 17:11.44 | [TK]D-Fender | rethus: then you can either change codecs or change your packetization rate. |
| 17:11.46 | rethus | connect to sipgate.de and the listeners call the sipgate-phonenumber (routet by sipgate to my asterisk) |
| 17:12.13 | rethus | how can i do this? Can u give me a hint? |
| 17:12.33 | rethus | which codec to choose? How changing packetzation rate? |
| 17:13.55 | manxpower | rethus: You need to read the Asterisk Book. |
| 17:13.57 | manxpower | ~answers |
| 17:13.58 | infobot | hmm... 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 |
| 17:15.11 | *** join/#asterisk sebbl (n=Momofu@109.192.162.148) |
| 17:18.16 | rethus | [TK]D-Fender:which codec should i choose if i only handle sip-connections (SIP-Protocol)? |
| 17:20.17 | *** join/#asterisk voipmonk (n=shido6@67.204.37.228) |
| 17:22.05 | *** join/#asterisk Badrobot- (n=badrobot@cpe-76-173-229-89.socal.res.rr.com) |
| 17:22.06 | [TK]D-Fender | rethus: whatever fits into your bandwidth |
| 17:25.07 | *** join/#asterisk verywiseman (n=khaled@unaffiliated/verywiseman) |
| 17:26.09 | Katty | hi |
| 17:27.09 | Katty | hi? |
| 17:27.39 | chuckf | hi! |
| 17:27.55 | Katty | hi :> |
| 17:28.48 | [TK]D-Fender | echo-cancels #asterisk |
| 17:29.53 | Katty | birdy birdy everywhere |
| 17:33.18 | rethus | [TK]D-Fender: so i have searched a bit. Sipgate.de don't support iax but ilibc. Maybe i try to switsch to ilibc for first step |
| 17:35.04 | *** join/#asterisk wopsy (n=wopsy@digi00332.digicube.fr) |
| 17:37.39 | Tim_Toady | ilibc? new c lib made by apple? |
| 17:38.04 | tzafrir | rethus means ilbc |
| 17:38.10 | Tim_Toady | i know (: |
| 17:38.24 | tzafrir | rethus, are both sides under your control? |
| 17:39.01 | *** join/#asterisk jtodd (i=w4hm08e3@asterisk/community-director-and-tie-dye-shirt-lover/jtodd) |
| 17:39.01 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 17:39.01 | rethus | tzafrir: what u mean? I have an asteriskserver here (under my C) and the sipgate-account i can handle too |
| 17:39.24 | tzafrir | Hmm... but I guess you don't control sipgate |
| 17:39.33 | tzafrir | What codecs are supported by sipgate? |
| 17:39.38 | rethus | :D no, sipgate not my company ;) |
| 17:39.46 | rethus | http://www.sipgate.de/faq/index.php?do=displayArticle&article=302&qw=iax |
| 17:41.45 | p3nguin | voipmonk: Do you think this is sufficient? http://www.pingtest.net/result/5995964.png and http://www.speedtest.net/result/658253359.png |
| 17:42.47 | tzafrir | gsm is also reasonable. It gives lower audio quality, but requires less CPU time |
| 17:42.52 | chuckf | goes to move more snow out of the way of the car |
| 17:43.45 | [TK]D-Fender | tzafrir: Yup, but iLBC is lower BW w/ higher quality (using 30ms/15 kbps standard) |
| 17:45.08 | p3nguin | How many ITSPs are supporting iLBC, though? |
| 17:46.25 | *** join/#asterisk Mango (n=Mango@d154-20-90-13.bchsia.telus.net) |
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| 18:01.16 | bcrisp | grr |
| 18:01.40 | dlynes | bcrisp, rrg |
| 18:01.46 | bcrisp | GRRrr |
| 18:02.01 | bcrisp | damn queues |
| 18:02.11 | dlynes | bcrisp, chill..... |
| 18:02.15 | bcrisp | grrrrrrrrrrr |
| 18:02.16 | bcrisp | heh |
| 18:02.18 | dlynes | bcrisp, hommmmmmmmm |
| 18:02.50 | bcrisp | call file calls into a test context and is sent to Queue(whatever) |
| 18:02.55 | bcrisp | but it abruptly just ends |
| 18:03.03 | bcrisp | stops playing moh cuz the user hung up? |
| 18:03.57 | bcrisp | that happens when i use a localchannel in the call file |
| 18:04.24 | bcrisp | if i specify some test SIP/bla .. it sits and rings rather than just answering and queueing... i must be missing something obvious |
| 18:04.48 | bcrisp | ill pastebin |
| 18:05.01 | dlynes | bcrisp, did you take a look at that free code on voip-info.org for hang up, hold place in queue, and call back? |
| 18:05.18 | bcrisp | didnt see hold place in queue but the others ya |
| 18:06.08 | dlynes | bcrisp, this one: http://www.voip-info.org/wiki/view/Asterisk+Queue+Callback? |
| 18:07.12 | bcrisp | ya |
| 18:07.24 | dlynes | it holds the place in queue |
| 18:07.33 | bcrisp | right thats not my problem |
| 18:07.43 | bcrisp | let me pastebin something real quick |
| 18:08.06 | dlynes | bcrisp, Thought you were still trying to pave your own road, instead of using someone else's road? |
| 18:10.10 | bcrisp | no |
| 18:10.37 | bcrisp | http://pastebin.ca/1721490 |
| 18:12.47 | rethus | does asterisk work on a 64 Bit ubuntu installation? |
| 18:13.05 | bcrisp | dlynes: can you kind of follow the logic in my pastebin? |
| 18:13.38 | dlynes | bcrisp, yes...the logic is easy enough to follow |
| 18:13.47 | bcrisp | k, the call file works |
| 18:13.53 | bcrisp | (albeit it rings twice into the test context) |
| 18:14.00 | dlynes | bcrisp, and then you go into the local context, it drops the caller into the queue, and then the line disconnects? |
| 18:14.04 | bcrisp | yes |
| 18:14.16 | bcrisp | it actually rings that extension twice |
| 18:14.22 | dlynes | bcrisp, it sends two calls into the test context for every call file? |
| 18:14.24 | bcrisp | since the channel name is Local/1@context |
| 18:14.30 | [TK]D-Fender | rethus: Yes |
| 18:14.40 | rethus | sounds good. |
| 18:14.42 | bcrisp | dlynes: ya |
| 18:15.31 | bcrisp | dlynes: i tried using Local/1@test with an Application:Queue Data:queueName but it never queues the call |
| 18:16.02 | dlynes | bcrisp, have you tried to find out why it's sending two calls for every call file? |
| 18:16.03 | manxpower | do you have an exten => 1 in extensions.conf [test] ? |
| 18:16.16 | bcrisp | manxpower ya |
| 18:16.27 | bcrisp | i understand the reason why its ringing twice but dont know how to work around it |
| 18:16.33 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 18:16.38 | dlynes | bcrisp, why is it ringing twice, then? |
| 18:16.39 | manxpower | Why is it ringing twice? |
| 18:16.40 | bcrisp | if 1) i dont supply an application, theni have to supply a channel, context, and extension |
| 18:16.52 | [TK]D-Fender | bcrisp: exten => 1,n,System(echo -e "Channel:Local/1@test\\nSet:LEAVEMSG=${UNIQUEID}\\nContext:test\\nExtension:1\\n" > /tmp/${UNIQUEID}.call ) <- both ends point to the same place |
| 18:17.10 | bcrisp | yes fender.. |
| 18:17.12 | [TK]D-Fender | bcrisp: YOU are sending A to A |
| 18:17.32 | [TK]D-Fender | bcrisp: Perhaps you should think of pointing one end somewhere else |
| 18:17.32 | bcrisp | im saying i dont know how to set up a channel that just relays the c all into the queue |
| 18:17.38 | dlynes | oh yeah...I wasn't thinking |
| 18:17.41 | manxpower | How avant guarde |
| 18:17.50 | [TK]D-Fender | bcrisp: F-ing well point to to an EXTEn that CALLS QUEUE :p |
| 18:17.57 | *** join/#asterisk Alagar (n=Administ@122.164.103.31) |
| 18:18.02 | dlynes | bcrisp, Get rid of the 'Channel:Local/1@test' out of the call file |
| 18:18.14 | bcrisp | i need some channel declaration there |
| 18:18.19 | bcrisp | i dont know what to put in there! |
| 18:18.55 | [TK]D-Fender | bcrisp: F-ing well point to to an EXTEN that CALLS QUEUE :p <------------------------------ |
| 18:19.06 | manxpower | bcrisp: most call have two enpoints |
| 18:19.15 | bcrisp | ya i understand.. |
| 18:19.21 | bcrisp | if i supply SIP/Blabla |
| 18:19.23 | *** join/#asterisk mnt_real (n=sinan@bas12-montrealak-1167974851.dsl.bell.ca) |
| 18:19.28 | bcrisp | it doesnt automatically answer |
| 18:19.37 | [TK]D-Fender | bcrisp: Oh and there is the issue of it playing instantly since it isn't going to sit around waiting to be "answered" by a queue member |
| 18:20.08 | *** join/#asterisk geneticx (n=geneticx@adsl-144-252-204.mia.bellsouth.net) |
| 18:20.08 | dlynes | bcrisp, Don't you want to reconnect the caller to the queue? |
| 18:20.14 | bcrisp | yes of course |
| 18:20.44 | dlynes | bcrisp, Why not something like Channel:SIP/mytrunk/guythatusedtobeinmyqueue? |
| 18:20.56 | dlynes | bcrisp, and then the rest would be the same |
| 18:21.16 | bcrisp | thats the part im not sure how to configure |
| 18:21.43 | bcrisp | i set up a SIP/tester interface in sip.conf.. but if i supply that as the Channel in call file it rings him.. |
| 18:21.52 | [TK]D-Fender | dlynes: Clearly he doesn't want to restrict to calling ONE PERSON |
| 18:21.53 | dlynes | bcrisp, very good |
| 18:22.01 | dlynes | bcrisp, it's supposed to |
| 18:22.04 | bcrisp | ....fk |
| 18:22.11 | dlynes | bcrisp, how else are you going to get him back in the queue, unless you call him? |
| 18:22.18 | bcrisp | huh? |
| 18:22.21 | bcrisp | its not a callback |
| 18:22.31 | [TK]D-Fender | dlynes: Go caffeinate before you confuse the locals... |
| 18:22.46 | *** join/#asterisk elliot98 (n=elliot@unaffiliated/elliot98) |
| 18:22.50 | elliot98 | hello all! |
| 18:22.51 | dlynes | bcrisp, ok, so you're just wanting to hold the place in the queue...thought you had already done that |
| 18:22.59 | manxpower | bcrisp: um, .call files are for creating calls. Looks like you are not wantin g to greate calls |
| 18:23.16 | bcrisp | User dials 1 while in queue, exits into queue context where the user records a message. Then a call file is generated that calls a context, this call should be placed in the queue |
| 18:23.30 | bcrisp | (the user hung up) |
| 18:23.32 | dlynes | bcrisp, it's not a call, though |
| 18:23.38 | dlynes | bcrisp, a call needs two legs |
| 18:24.06 | [TK]D-Fender | manxpower: Go join him for that coffee :) |
| 18:24.09 | bcrisp | i want to originate a call that takes the place of the user |
| 18:24.09 | elliot98 | I am looking into * features...I am wondering if I can enable blind transfer for only a limited amount of devices |
| 18:24.27 | bcrisp | when that call is connected it plays back the msg that the prior user left |
| 18:24.28 | [TK]D-Fender | bcrisp: Neither of these 2 seems to be on track for your issue |
| 18:24.47 | bcrisp | i wish i could draw in IRC heh |
| 18:25.01 | bcrisp | user leaves msg, hangs up. I need to originate a call that comes into the queue again |
| 18:25.09 | bcrisp | (like a robo call IN to the queue) |
| 18:25.10 | [TK]D-Fender | draws the bestest stick-people EVAR |
| 18:25.14 | dlynes | bcrisp, Channel:Local/2@test, where you drop exten 2 into an infinite moh just to hold the call |
| 18:25.39 | dlynes | bcrisp, then when you're ready to call the person back, call them back, and reconnect them to that call, booting out the moh |
| 18:25.51 | bcrisp | cries |
| 18:26.05 | bcrisp | im not calling them back |
| 18:26.23 | bcrisp | i want to generate a robo call that calls into the queue ON BEHALF of the person hehe |
| 18:26.24 | dlynes | bcrisp, oh....you're still on that old issue, where you want to play a recording back to the operator? |
| 18:26.32 | bcrisp | yes and it works |
| 18:26.37 | bcrisp | except for the duplicate call |
| 18:26.40 | dlynes | bcrisp, ah |
| 18:27.10 | bcrisp | it does actually play back the message and hangup |
| 18:27.12 | dlynes | bcrisp, in the Channel:... field, you'll need it to go into a context that can detect when the operator's accepted the call |
| 18:27.23 | bcrisp | no |
| 18:27.31 | dlynes | bcrisp, when it's detected that the operator's picked up the call from the queue, it plays back the recording |
| 18:27.43 | bcrisp | i already have all that working |
| 18:27.46 | bcrisp | using queuemacro |
| 18:27.56 | bcrisp | member macro |
| 18:28.01 | dlynes | you must not, or you wouldn't have it ringing an extension twice |
| 18:28.14 | bcrisp | after the user records the message and hangs up |
| 18:28.22 | bcrisp | CLI - show queue shows 2 callers |
| 18:28.40 | bcrisp | the agent answers the first and it plays back the message and hangs up |
| 18:28.44 | bcrisp | 2nd call is exactly the same |
| 18:29.33 | bcrisp | ill pastebin that shows the queue macro too |
| 18:29.40 | bcrisp | maybe that will clear up my sorcery |
| 18:29.49 | dlynes | bcrisp, ok, I think I know what you're doing now |
| 18:29.55 | bcrisp | k |
| 18:30.19 | dlynes | bcrisp, Under Channel: in the call file and context/exten in the call file you have them both going to the same place, and that's why you've got two calls...we've already established that |
| 18:30.30 | dlynes | bcrisp, Make one of them go to a Noop() and a Hangup() instead |
| 18:30.36 | bcrisp | ok |
| 18:30.43 | dlynes | bcrisp, Or, failing that working (it might hang up both ends) |
| 18:30.55 | bcrisp | the other thing i notice is that if a user records message via Record() but DOESNT hit #, the rest doesnt happen |
| 18:31.01 | dlynes | bcrisp, Make one of them go to a Noop(), monitor the other leg, wait for it to finish, and then hangup() |
| 18:31.15 | bcrisp | ya i did that too |
| 18:31.32 | bcrisp | i had Local/1@context and then had Context:context Extension: 2 |
| 18:31.53 | bcrisp | ill give it another whirl |
| 18:32.33 | dlynes | bcrisp, yeah...so wait in the one context until the other leg completes, before hanging up, then |
| 18:32.52 | bcrisp | hmm |
| 18:33.57 | dlynes | bcrisp, the question is, how do you determine that the other leg has completed... |
| 18:35.56 | dlynes | [TK]D-Fender, couldn't he just solve his problem by use of a DeadAGI() script? ... I'm not familiar with AGI, but I believe that's what DeadAGI() is supposed to be for....where a caller has probably already disconnected from the channel, right? |
| 18:36.53 | *** join/#asterisk ttl- (n=patrick@d5153A420.access.telenet.be) |
| 18:38.12 | *** join/#asterisk crazybyte (n=crzp@unaffiliated/crazypenguin/x-000001) |
| 18:41.46 | [TK]D-Fender | dlynes: Dead channels aren't going to play audio to a live caller, and if its sitting in queue it sure as hell isn't going to do it in a DEAD STATE |
| 18:50.05 | elliot98 | does anyone have experience with addqueuemember? |
| 18:50.24 | elliot98 | gives a big to wave to the Asterisk enthusiests |
| 18:51.33 | elliot98 | when I add a ZAP agent (a BRI line) using addqueuemember, is there any way to track down the number * called...all that shows up in "show channels" I think is a ZAP/1-1 listing |
| 18:58.51 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.229.58) |
| 18:59.57 | *** join/#asterisk ethicx (n=geneticx@adsl-144-252-204.mia.bellsouth.net) |
| 19:02.32 | *** join/#asterisk ttl- (n=patrick@d5153A420.access.telenet.be) |
| 19:07.09 | elliot98 | I have a phone registered behind NAT...it can call, but it does not receive phone calls |
| 19:07.15 | elliot98 | I set the phone to NAT |
| 19:07.16 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 19:07.21 | elliot98 | but no go |
| 19:08.34 | [TK]D-Fender | ~sipnat |
| 19:08.35 | 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 |
| 19:08.37 | [TK]D-Fender | ^^^^^ |
| 19:12.05 | rethus | i search the german sound-files for ubuntu (asterisk) anyone know an adress? In "Asterisk-Buch" only old URL's |
| 19:16.29 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:18.22 | Mango | I AM NOT CIRCUIT BUSY DAMMIT. |
| 19:18.25 | p3nguin | If Playback() is the first command (first priority on the exten), is it recommended to leave off the explicit Answer() since Playback() will answer the channel? |
| 19:19.01 | p3nguin | I know it's okay to leave it off, but I want to know if it is recommended to include it anyway or skip it. |
| 19:23.28 | rethus | ilbc is not anymore in asterisk 1.6, can i anywhere get a precompiled module for ubuntu? |
| 19:26.15 | *** join/#asterisk voipmonk (n=shido6@67.204.37.228) |
| 19:26.35 | *** join/#asterisk Godfather_ (n=Godfathe@62.43.134.46.dyn.user.ono.com) |
| 19:29.07 | *** join/#asterisk puzzled (n=patrick@535335AA.cable.casema.nl) |
| 19:34.02 | *** join/#asterisk wam (i=wam@unaffiliated/wam) |
| 19:44.45 | lesouvage | Is it possible to use the value of a variable as the name of another variable and use it to get the value of that other variable. The variable holding part of the name: NUMBER=1001 name of other variable: NR1001 So it might look something like $[NR${NUMBER}]. |
| 19:45.21 | lesouvage | But that isn't working. Is this possible anyway? |
| 19:45.50 | *** join/#asterisk ayrjola (n=ayrjola@a88-115-162-170.elisa-laajakaista.fi) |
| 19:45.57 | *** join/#asterisk DarkRift (n=dark@65.92.170.77) |
| 19:49.12 | dlynes | [TK]D-Fender, ok...didn't know how DeadAGI() worked... |
| 19:49.27 | dlynes | Mango, ? |
| 19:50.16 | Mango | I got it solved, thank you :) |
| 19:51.50 | dlynes | rethus, sure it is |
| 19:52.13 | rethus | in extensions.conf is it right to write exten => TIMEOUT(response)=5 ? |
| 19:52.22 | rethus | have version 1.6 |
| 19:52.33 | dlynes | rethus, so do i, and ilbc is definitely in 1.6 |
| 19:52.47 | dlynes | rethus, exten => Set(TIMEOUT(response)=5) |
| 19:53.14 | fenrus | anyone know of a modern and active project creating graphs of mysql-cdr's? |
| 19:53.32 | dlynes | fenrus, a2billing? |
| 19:53.54 | rethus | dlynes: thx, but ilbc is able in 1.6, but not in it.. must insert manualy |
| 19:54.04 | fenrus | dlynes, thanks - will have a look |
| 19:54.13 | dlynes | rethus, no...it's in it...what are you talking about? |
| 19:54.26 | dlynes | rethus, Are you installing it from a binary package? |
| 19:54.39 | dlynes | rethus, if so, that's part of your problem right there |
| 19:54.57 | dlynes | rethus, the package maintainer chose not to include ilbc, then...nothing to do with asterisk |
| 19:55.32 | dlynes | fenrus, a lot of the open source billing packages out there for asterisk are very continent-specific, too |
| 19:56.05 | dlynes | fenrus, i.e. they're only good for north america and would require a lot of work to get them to work in europe, or they're written with europe in mind, and require a lot of work to work in north america, ... |
| 19:56.49 | fenrus | <PROTECTED> |
| 19:56.52 | fenrus | all right |
| 19:57.02 | fenrus | but i just want the statistics-part of it |
| 19:57.04 | dlynes | fenrus, and they usually have multi-currency support added in as an afterthought |
| 19:57.19 | dlynes | fenrus, and so the multi-currency support sucks at best |
| 19:57.27 | fenrus | ;) |
| 19:57.32 | dlynes | fenrus, ok...then any of them should suit you just fine |
| 19:57.36 | dlynes | fenrus, even mcc |
| 19:57.45 | dlynes | fenrus, mcc has a pretty good reporting section |
| 19:57.53 | fenrus | nice |
| 19:58.25 | fenrus | at the moment my cdrs are in mysql, since i figured i'd go write my own reporting-application |
| 19:58.36 | fenrus | but i guess that thre's support for that too |
| 19:58.51 | dlynes | fenrus, if you decide to use mcc, they're very european-centric: http://www.kolmisoft.com/ |
| 19:59.09 | dlynes | fenrus, You could also try Crystal Reports or JasperReports, too |
| 19:59.10 | fenrus | that's nice since i'm from europe |
| 19:59.33 | dlynes | fenrus, seeing as how you're just looking for reporting |
| 20:00.54 | dlynes | fenrus, JasperReports (http://jasperreports.sf.net/) - (http://jasperforge.org/website/jasperreportswebsite/trunk/index.html?group_id=252) |
| 20:01.04 | *** join/#asterisk voipmonk (n=shido6@67.204.37.228) |
| 20:01.08 | dlynes | fenrus, it uses Java, but has won a number of awards |
| 20:01.42 | fenrus | the samples look lice, i think i'll give it a try |
| 20:02.37 | dlynes | fenrus, and it looks like it's got a report designer now, too (iReport) |
| 20:02.53 | dlynes | fenrus, So, theoretically, it could even be a good base for a billing system now |
| 20:06.56 | fenrus | neat |
| 20:09.24 | *** join/#asterisk Ad-Hoc (n=nimbus@62.1.163.240.dsl.dyn.forthnet.gr) |
| 20:09.35 | *** join/#asterisk JAMMAN2110 (n=james@unaffiliated/jamman2110) |
| 20:21.22 | rethus | dlynes: jes i installed it from ubuntu binarys |
| 20:21.32 | rethus | but no problem, i didn't need it anymore |
| 20:23.09 | rethus | how are the soundfiles sorted? |
| 20:23.14 | rethus | some has prefix vm- |
| 20:23.17 | rethus | whta this means |
| 20:24.26 | rethus | and how can i use extension.ael instead of extensions.conf? |
| 20:26.09 | voipmonk | already looking at ael and dont know the basics |
| 20:27.19 | dlynes | rethus, vm- means voicemail |
| 20:27.35 | dlynes | rethus, and they're sorted however you feel like sorting them |
| 20:27.45 | dlynes | rethus, type 'ls --help' at the command line |
| 20:27.52 | rethus | dlynes: thanx. Is there a autio for "Please enter Pin?" |
| 20:28.21 | dlynes | rethus, Just use the Authenticate() application...it automatically plays that audio back |
| 20:30.26 | rethus | dlynes: k, it works. but its in eglish, i need german. |
| 20:30.35 | rethus | have installed the german files |
| 20:30.50 | rethus | ubuntu in folder /usr/share/asterisk/sounds/de |
| 20:30.56 | rethus | but system anser in english |
| 20:37.48 | dlynes | rethus, Set(LANGUAGE=de) |
| 20:44.26 | rethus | dlynes: and how can i preset it for whole asterisk? |
| 20:45.06 | rethus | in sip.conf i have on all [xyz] set language=de |
| 20:51.11 | dlynes | rethus, you don't: exten => 1,1,Set(LANGUAGE()=de) ; Set language to deutsch |
| 20:51.30 | *** join/#asterisk rejohn_ (i=415b7422@gateway/web/freenode/x-bshgbgbwdtaehaon) |
| 20:52.17 | dlynes | rethus, btw...why wouldn't you do language=de in the sip general section, instead of defining it for every sip context? |
| 20:55.11 | rethus | i wan't to be save that it was used, but it wasn't |
| 20:55.19 | rejohn_ | <PROTECTED> |
| 20:58.31 | rethus | dlynes: any idea why it is still in english? |
| 21:02.07 | *** join/#asterisk JAMMAN2110 (n=james@unaffiliated/jamman2110) |
| 21:04.46 | dlynes | rethus, no idea...never used multilingual support in asterisk |
| 21:05.58 | dlynes | brb |
| 21:06.26 | *** join/#asterisk dlynes (n=dlynes@bas5-hamilton14-1096762521.dsl.bell.ca) |
| 21:07.02 | rethus | can i set maxcalls for asterisk via agi ? |
| 21:07.28 | rethus | so if it set to 5 and 6th call, * should dissallow connection for the 6th |
| 21:09.33 | *** join/#asterisk Tech_Travis (n=Administ@cpe-76-87-9-130.socal.res.rr.com) |
| 21:10.01 | rethus | what is the option internal_timing in asterisk.conf? |
| 21:16.00 | dlynes | rethus: probably to use a software timer, instead of dahdi |
| 21:23.03 | rethus | k. thanks |
| 21:24.04 | *** join/#asterisk crazybyte (n=crzp@unaffiliated/crazypenguin/x-000001) |
| 21:24.34 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 21:24.55 | *** join/#asterisk aidinb (n=Aidin@24-182-32-138.static.lnbh.ca.charter.com) |
| 21:58.51 | rethus | don't get the german language to work :(# |
| 22:02.43 | *** join/#asterisk grandpapadot (n=nonya@99-175-248-81.lightspeed.brhmal.sbcglobal.net) |
| 22:02.50 | *** part/#asterisk grandpapadot (n=nonya@99-175-248-81.lightspeed.brhmal.sbcglobal.net) |
| 22:02.58 | *** join/#asterisk lowtek (n=nonya@99-175-248-81.lightspeed.brhmal.sbcglobal.net) |
| 22:03.45 | lowtek | Hi all, I'm installing a new * box and want to go with 1.6, but there are three seperate versions (1.6.0, 1.6.1, 1.6.2) and apparently three different branches. How do I know which one to use and why the split? We're running 1.4 in production now. |
| 22:04.35 | rethus | if i look "core show settings" language still on "en" instead i have set it to "de" |
| 22:04.50 | rethus | where can i look for the problem |
| 22:05.02 | rethus | in sip.conf [general] i have language=de |
| 22:05.16 | Katty | hellloooo |
| 22:08.02 | lowtek | Hi all, I'm installing a new * box and want to go with 1.6, but there are three seperate versions (1.6.0, 1.6.1, 1.6.2) and apparently three different branches. How do I know which one to use and why the split? We're running 1.4 in production now. |
| 22:08.25 | Katty | jblack: ping |
| 22:09.11 | ChannelZ | lowtek: it's a good question and the * site is pretty crap, not explaining *anything* |
| 22:09.45 | ChannelZ | lowtek: it's basically just feature sets. If you're building from scratch I suppose there is no reason not to go with the newest |
| 22:09.48 | *** join/#asterisk smooth_penguin (n=smoove@59.95.6.173) |
| 22:10.47 | ChannelZ | lowtek: I think they do so many branches because of the nature of * being somewhat a 'mission critical' app so lots of new features get a new version so as not to disrupt existing installations with changes but still fix bugs in them |
| 22:11.07 | lowtek | Right, I figured that part out, but I really can't find a list of branch version features ... |
| 22:11.51 | lowtek | There must be reasons for the split(s) ... |
| 22:12.57 | ChannelZ | http://svnview.digium.com/svn/asterisk/trunk/CHANGES?view=markup |
| 22:13.29 | ChannelZ | that's about the closest I could find last night |
| 22:14.23 | ayrjola | lowtek: if you want to use 1.6.x use 1.6.1... I have seen some stability issues in 1.6.0 and 1.6.2 was released.. I think yesterday :) |
| 22:15.41 | ayrjola | 1.6.2 seems quite good, and we have run Rc in small production enviroments with no problems |
| 22:15.41 | rethus | i have installed german language-pack. if i rename /usr/share/asterisk/sounds/en to en_old and rename de to en, i got the german language... but not if i have only the folder de under sounds. Any idea why |
| 22:15.46 | rethus | have set sip.conf to language=de |
| 22:16.17 | rethus | and asterisk.conf to languageprefix = yes |
| 22:16.20 | *** join/#asterisk WindBack (n=quassel@200-122-74-15.cab.prima.net.ar) |
| 22:16.54 | lowtek | Does the 1.6 branch still require libpri? |
| 22:17.23 | WindBack | I'm going to reinstall asterisk on the same box. I have a g729 licences purchased. There is any trouble to reinstall g729 module? |
| 22:17.45 | lowtek | WindBack: Nope, just backup your .LIC files just in case. |
| 22:18.00 | ayrjola | lowtek: depends on what you use * |
| 22:18.33 | lowtek | I want to use meetme, so I'll need dahdi_dummy, but noticed that dahdi was XXX out in menuconfig, which prompted me to ask ... |
| 22:18.44 | lowtek | Also noted that libpri stops at ver 1.4.x on downloads.digium.com |
| 22:18.54 | ayrjola | dahdi needs libpri |
| 22:19.02 | *** join/#asterisk MaliutaLap (n=biteme@96.53.148.53) |
| 22:19.13 | WindBack | lowtek: ok, when i will run make unistall. The licences will be erased from /var/lib/asterisk/licences? |
| 22:19.23 | ayrjola | that is libpri vwrsion number |
| 22:19.34 | ayrjola | nothing to do with * version |
| 22:19.59 | lowtek | Well, dahdi compiled and installed just fine without libpri ... and I was able to install dahdi_dummy ... |
| 22:20.18 | lowtek | ... and it loads -> dahdi_dummy 3204 0 |
| 22:20.46 | lowtek | WindBack: No, it won't remove anything it didn't put there. |
| 22:21.07 | WindBack | lowtek: thans |
| 22:21.13 | ayrjola | I have always installed libpri, just in case if you need PRI hardware later |
| 22:21.30 | ayrjola | then you dont need to recompile dahdi |
| 22:21.47 | lowtek | hmm.. but the dahdi cahnnel driver is XXX'd out in menuconfig ... |
| 22:22.01 | *** join/#asterisk mpe (n=mpe@0x4dd624b2.adsl.cybercity.dk) |
| 22:23.05 | ayrjola | well, install libpri ja recompile dahdi... takes 10 minutes :) theres no harm to libpri |
| 22:23.18 | ayrjola | or * |
| 22:23.33 | lowtek | Right, just trying to understand ... |
| 22:23.34 | lowtek | Thanks |
| 22:25.50 | *** join/#asterisk danj1980 (n=dan@91.110.29.19) |
| 22:26.27 | danj1980 | Hi, does anyone have a cheap source for Polycom in the UK? |
| 22:26.59 | Han | ebay? |
| 22:30.15 | danj1980 | Nothing much on ebay |
| 22:31.07 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 22:31.28 | *** join/#asterisk erqzor (n=quassel@217.139.119-80.rev.gaoland.net) |
| 22:33.36 | StevenR_ | danj1980: voipon.co.uk ?(never used them, but they *seem* cheap) |
| 22:39.06 | *** join/#asterisk bleger (n=ladorval@c-68-45-58-209.hsd1.nj.comcast.net) |
| 22:41.48 | TSM | danj1980: voipon too, i only got cheeper from insight UK though special deal |
| 22:45.45 | *** join/#asterisk rethus1 (n=contio@p50874086.dip.t-dialin.net) |
| 22:47.33 | danj1980 | Ive used voipon, but wanted to see if I can find cheaper. |
| 22:47.45 | danj1980 | Computer2000 dont stock polycom |
| 22:50.17 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 22:52.35 | rethus1 | have an complicated authentication-question |
| 22:53.38 | rethus1 | i will authenticate a caller by his pin... so if the pin 123 is enterd, * should search in the db for this pin, and if it exist, it give back the name of the caler.... how can i do this? |
| 22:54.00 | *** join/#asterisk jtodd (i=hnd7beyp@asterisk/community-director-and-tie-dye-shirt-lover/jtodd) |
| 22:54.00 | *** mode/#asterisk [+o jtodd] by ChanServ |
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| 23:04.22 | *** join/#asterisk afink (n=afink@204.26.87.226) |
| 23:04.24 | TSM | danj1980: call insight, speak to mark sharpe |
| 23:04.39 | *** join/#asterisk tzafrir (n=tzafrir@212.179.75.202) |
| 23:17.52 | danj1980 | Thanks |
| 23:18.02 | danj1980 | Whats his position in insight? |
| 23:19.27 | danj1980 | Also, we are thinking about leasing polycom phones to our clients. Does anyone know the safest way to safe guard our investment? We dont want our clients to be able to use their polycom handsets if they run off. |
| 23:19.59 | danj1980 | Is there any way to lock the provisioning server and/or admin password? |
| 23:21.22 | p3nguin | When I call my cell phone's voicemail from my Cisco phone on my * system, the dtmf for the password isn't being sent reliably. I tried both rfc2833 and inband settings, but the dtmf still isn't good enough for the entry to be read as being correct. What could be the issue and what should I look at to try to figure out how to get dtmf to be reliable? |
| 23:22.26 | TSM | danj1980: not sure how well you can lock the pollys down, there is a boot password that can be cahnged, you can encrypt the FTP/TFTP boot files etc.. |
| 23:23.04 | p3nguin | The outbound call from * to the cell phone voicemail system is over an IAX2 trunk, and there doesn't seem to be any dtmf settings for IAX2. |
| 23:23.29 | *** part/#asterisk rethus1 (n=contio@p50874086.dip.t-dialin.net) |
| 23:24.53 | *** join/#asterisk simplydrew (n=simplydr@adsl-074-171-198-201.sip.owb.bellsouth.net) |
| 23:29.05 | p3nguin | Using inband on the sip side (phone to *), and trying both ulaw and g729, it says invalid password, even though I am typing it in correctly. Using rfc2833 on the sip side, the dtmf doesn't even register at all. The system keeps telling me to enter the password, even though I did. |
| 23:29.47 | jblack | Did someone attempt to contact me here? |
| 23:29.57 | jblack | katty: pong |
| 23:30.25 | *** join/#asterisk coppice (n=chatzill@25.176.64.202.dyn.pacific.net.hk) |
| 23:31.11 | *** join/#asterisk voipmonk (n=shido6@67.204.37.228) |
| 23:36.23 | *** join/#asterisk seanbright (n=sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 23:38.46 | *** join/#asterisk sirrus (n=sirrus@p4FFF0F7F.dip.t-dialin.net) |
| 23:38.52 | sirrus | Hi there! |
| 23:39.13 | sirrus | I've got a strang issue with chan_capi and avm C2 |
| 23:39.48 | sirrus | Is there anyone who can give me tips? |
| 23:42.08 | sirrus | The problem is - 1st call to meetme works fine, second call gets the audio stream from the first call... |
| 23:48.07 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 23:50.12 | *** join/#asterisk astrutt (n=astrutt@67.93.60.229) |
| 23:51.04 | *** join/#asterisk voipmonk (n=shido6@67.204.37.228) |
