| 00:01.27 | nny | [TK]D-Fender: ok almost done, got a good chunk of the successful registration |
| 00:03.03 | nny | [TK]D-Fender: http://pastebin.com/4HMRsa7Y |
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| 01:00.36 | lopz | hi |
| 01:00.40 | lopz | Good evening, I'm building my IVR and I have a problem when I directly typing the option extension, eg 120, when digit 1, I sent to the extension 1, I can not complete the 120, I tried retracing time between tones using: |
| 01:01.12 | lopz | Set(TIMEOUT(digit)=5) or featuredigittimeout = 1000 and transferdigittimeout => 3 in features.conf |
| 01:01.15 | lopz | any idea ? :( |
| 01:02.10 | lopz | *CLI> core show version Asterisk 1.6.2.21 built by root @ arch1 on a i686 running Linux on 2012-05-25 00:13:07 UTC |
| 01:02.20 | [TK]D-Fender | lopz: idea : pastebin your entire IVR context so we can see. |
| 01:02.22 | [TK]D-Fender | ~pb |
| 01:02.22 | 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. |
| 01:02.23 | [TK]D-Fender | ^^^ |
| 01:03.09 | lopz | This weekend started with Asterisk, I have very clear how it works |
| 01:05.38 | lopz | hey [TK]D-Fender -> http://pastebin.com/hXF8LhTb :) |
| 01:06.18 | [TK]D-Fender | lopz: You don't have a 120 in there you can dial |
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| 01:07.53 | lopz | uhmmm my idea is that you can type any number of context [sales] or I have to put one at all? or there is a _120X, ....? |
| 01:12.26 | [TK]D-Fender | lopz: You didn't provide anything for it to do that witrh |
| 01:12.52 | lopz | ohh yeah, [TK]D-Fender I agree exten => _12XX,1,Goto(sales,${EXTEN},1) same => n,Hangup() , its correct? |
| 01:12.54 | [TK]D-Fender | lopz: You can use "include => context" to allow that context to see the contents of another if you want them to have everything in there as well |
| 01:13.10 | [TK]D-Fender | "include" is the proper way when possible |
| 01:13.50 | lopz | wait.. testing.. |
| 01:15.21 | lopz | [TK]D-Fender: http://pastebin.com/TUj8ceFJ |
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| 01:16.19 | [TK]D-Fender | yup |
| 01:18.13 | lopz | [TK]D-Fender: Uhm .. but now you can access voicemail from a caller outside the company: ( |
| 01:18.49 | [TK]D-Fender | lopz: then you should split your contexts up a little bit MORE so that you have a chunk of JUST sales in that one... |
| 01:18.58 | [TK]D-Fender | and make another that inclides vmail, etc |
| 01:19.08 | [TK]D-Fender | time to split things up nicers |
| 01:19.11 | [TK]D-Fender | - |
| 01:19.13 | [TK]D-Fender | -s |
| 01:19.43 | [TK]D-Fender | ok, I'm off for a while... |
| 01:19.54 | lopz | It is a practice for the University of the field of VoIP, and they asked that we implement a call center in a small business, but must meet certain safety measures, etc. |
| 01:20.20 | lopz | [TK]D-Fender: thanks |
| 01:20.42 | [TK]D-Fender | Yes, so you should hav a context to include that has only the parts you want in it. |
| 01:20.53 | [TK]D-Fender | if you want that + a few more things... make another context. |
| 01:20.57 | [TK]D-Fender | this is a heirarchy... |
| 01:21.14 | [TK]D-Fender | break it dow into the separate bits and make useful "container " contexts |
| 01:21.19 | [TK]D-Fender | is actually off now... |
| 01:21.48 | lopz | ok |
| 01:22.09 | lopz | [TK]D-Fender: thanks ;) |
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| 03:31.12 | dandate2 | does g729 coded use more cpu on the softphone's computer than ulaw? |
| 03:32.00 | dandate2 | we've been having quality problems on some really old computers ever since we switched from ulaw |
| 03:33.01 | dandate2 | and i imagine its their old computers trying to compress real voice into g729 |
| 03:33.20 | dandate2 | because the issue is that they can hear the caller just fine, but the caller cannot hear us too well |
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| 04:15.05 | *** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.4.1 (2012/05/29), 1.8.12.1 (2012/05/29), DAHDI-linux 2.6.1 (2012/04/20), DAHDI-tools 2.6.1 (2012/04/20), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 04:18.39 | carrar | dandate2 |
| 04:18.55 | carrar | dandate2, remember the cpu is also decoding g729 also |
| 04:19.03 | carrar | so it should be both ways in theory |
| 04:19.31 | carrar | But, yes, g.711u uses less cpu the g729 |
| 04:19.35 | carrar | then |
| 04:23.43 | dandate2 | i know encoding and decoding is taxing on the PBX |
| 04:23.55 | dandate2 | but is g729 also taxing on the softphones computer? |
| 04:26.13 | carrar | I wouldn''t think it's very much |
| 04:26.21 | carrar | since you are decoding it just fine |
| 04:26.41 | carrar | Why not switch to g.711u and see if it makes any difference |
| 04:27.08 | carrar | ... and then report back!! |
| 04:27.42 | carrar | detail decoding cpu POST!! |
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| 06:12.29 | din3sh | hello all |
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| 06:14.52 | din3sh | how does avaya fare up with asterisk? |
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| 06:54.39 | jacc0 | din3sh: what type of avaya are you talking about? |
| 06:55.59 | jacc0 | newer types (call manager) only allow incomming sip packets using tcp (or you need an extra license) |
| 06:56.15 | jacc0 | it's dull , I know |
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| 07:12.12 | jacc0 | so you have to set transport=tcp for the trunk |
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| 07:37.31 | din3sh | hey jacc0, was busy sorry |
| 07:37.48 | din3sh | i didnt mean trunking actually |
| 07:38.02 | din3sh | i meant pbx comparison, asterisk v/s avaya |
| 07:38.25 | din3sh | how can client be persuaded that asterisk is better? |
| 07:38.44 | din3sh | i mean avaya have released some fancy collaboration tool called avaya fare |
| 07:38.56 | din3sh | avaya flare |
| 07:40.23 | din3sh | http://www.youtube.com/watch?v=KW_uzYiquug |
| 07:50.11 | jacc0 | closed source VS opensource should be all you need to convince |
| 07:51.46 | jacc0 | the time where you trust a developer on his word without being able to verify his words/work is far behind us. |
| 07:52.21 | jacc0 | trust is good. verification is better |
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| 08:02.10 | jacc0 | definitely when talking about goverment use |
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| 08:08.21 | pithagoraians | <PROTECTED> |
| 08:09.41 | din3sh | jacc0: avaya is opening up, sip phones can be used instead of proprietary avaya phone |
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| 08:20.20 | stix | What can cause the "invalid" state of a queue-member? SIP/3017 (dynamic) (Invalid) has taken no calls yet |
| 08:23.09 | jacc0 | pithagoraians: exten = 012345678,1,dial(sip/100) |
| 08:23.31 | jacc0 | where 012345678 is the DDI and 100 is your internal sip number |
| 08:24.22 | ChannelZ | device name |
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| 08:29.32 | jacc0 | pithagoraians: you need to add that to your extensions.conf |
| 08:32.37 | din3sh | jacc0: how can we restrict transfers within a particular context? |
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| 08:37.45 | jacc0 | no clue |
| 08:39.47 | din3sh | i'll ask google then |
| 08:39.47 | din3sh | :p |
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| 08:40.56 | pithagorians | jacc0: i get WARNING[4455]: app_dial.c:2218 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) |
| 08:40.56 | pithagorians | <PROTECTED> |
| 08:43.16 | jacc0 | what did you set in your dialplan? is the phone online? |
| 08:43.25 | jacc0 | and registerd? |
| 08:44.38 | jacc0 | type 'sip show peers' in Asterisk CLI to see whitch phones are registerd |
| 08:48.15 | pithagorians | <jacc0> ah sorry, it's my fault. the sip phone handling this number was not connected |
| 08:50.04 | jacc0 | ;) |
| 08:50.42 | pithagorians | <jacc0> any clue why i get such statuses in asterisk cli http://pastebin.com/URGdzuu6 ? |
| 08:52.35 | jacc0 | asterisk is lookingup DNS records to find out the ip that belongs to the name _sip._udp.sipconnect.sipgate.de |
| 08:55.16 | jacc0 | http://www.voip-info.org/wiki/view/Asterisk+SIP+srvlookup |
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| 10:51.31 | pithagorians | i'm using such an extension to send called number with ++ to the sipgate trunk - exten => _++XXXXX.,1,Goto(outbound-e164,${EXTEN:2},1) but aster claims [May 30 12:49:12] NOTICE[4417]: chan_sip.c:22650 handle_request_invite: Call from 'Company_phone-7003' (10.1.2.240:3123) to extension '00+490396945050' rejected because extension not found in context 'phones'. |
| 10:52.02 | pithagorians | while external calls goes into the context outbound-rules |
| 10:52.35 | pithagorians | any clue? |
| 10:57.40 | kaldemar | pithagorians: _++XXXXX. does not match 00+490396945050. the latter begins with two zeroes. |
| 10:58.20 | jacc0 | exten => _00+XXXXX.,1,Goto(outbound-e164,${EXTEN:2},1) |
| 10:58.20 | pithagorians | <kaldemar> i'm calling it like ++490396945050 |
| 10:58.40 | kaldemar | pithagorians: your phone is sending 00+... |
| 10:58.44 | jacc0 | then your phone replaces the first + with 00 |
| 11:00.26 | pithagorians | yes, linphone has such function - replace + with 00. now works. thanks ;) |
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| 11:24.31 | v0lZy | hello |
| 11:24.45 | jacc0 | hi |
| 11:25.21 | pithagorians | sipgate trunk has such an option called "fallback sender number". all my external calls go via this number and in some cases i Unknown caller. any clue? |
| 11:25.32 | v0lZy | I am using Draytek VigorPhone 350 and i've locked the phone settings for the users. Unfortunately, those locked settings include an absolute redirect for calls... |
| 11:25.54 | v0lZy | So now a user cant redirect their phone line to another user and im thinking |
| 11:26.04 | v0lZy | is it possible to script this in asterisk in such a way that a user for example enters |
| 11:26.25 | v0lZy | #48 and all calls to his line are redirected to internal number 48 |
| 11:27.33 | v0lZy | im thinking like #48 to engage redirect |
| 11:27.46 | v0lZy | and as soon as a user picks up the handset, demolish the redirect. |
| 11:27.50 | v0lZy | but i have no idea how to go about something like this. |
| 11:31.25 | jacc0 | 'as soon as a user picks up the handset' : asterisk will have no notion of that, only on incomming calls or when an outgoing call is made |
| 11:31.47 | v0lZy | hm |
| 11:31.56 | v0lZy | ok |
| 11:31.58 | v0lZy | how about this |
| 11:32.16 | v0lZy | when u dial #43 |
| 11:32.32 | v0lZy | hm... |
| 11:32.36 | v0lZy | thought i had it |
| 11:32.38 | v0lZy | dang. |
| 11:32.54 | v0lZy | if the user would be put into an application that waits for hangup |
| 11:33.10 | v0lZy | it could work |
| 11:34.02 | v0lZy | ah, i got it |
| 11:35.26 | v0lZy | user dials #43 and is connected to an application that catches all these #XX numbers. As long as that line holds, incoming calls to the users number are redirected to number 43 |
| 11:36.20 | v0lZy | and the chan there is muted or something so no sound is passed around |
| 11:36.23 | v0lZy | would that work? |
| 11:36.33 | wdoekes2 | pithagorians: you're probably sending the wrong cli? and/or the wrong mechanism? (see sendrpid= sip.conf option and the your sipgate trunk documentation about cli format (leading +? leading 00?)) |
| 11:43.58 | *** part/#asterisk atan (~atan@unaffiliated/atan) |
| 11:52.29 | WIMPy | pithagorians: Oh I just came across that the other day. For the trunk plan you need a p-preferred-identity header. |
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| 11:54.19 | v0lZy | hi WIMPy |
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| 12:14.28 | v0lZy | how do i reload the voicemail.conf? |
| 12:14.46 | WIMPy | You don't have to. |
| 12:15.35 | v0lZy | hm |
| 12:15.44 | v0lZy | i dont get it, i dont want asterisk to send me notification emails |
| 12:15.52 | v0lZy | so i deleted my e-mail address from voicemail.conf |
| 12:15.58 | v0lZy | but i still keep getting notifications |
| 12:16.52 | kaldemar | v0lZy: voicemail reload |
| 12:16.57 | v0lZy | thanks |
| 12:17.01 | WIMPy | Hmm. I thought that one was read on demand, but maybe that's (no longer) the case. |
| 12:17.03 | v0lZy | ok, i didnt get it this time |
| 12:17.09 | v0lZy | lets see if it is stable like that now |
| 12:18.20 | v0lZy | ok |
| 12:18.22 | v0lZy | that worked |
| 12:18.31 | v0lZy | can u help me out with another thing? |
| 12:18.42 | v0lZy | i described it above |
| 12:19.10 | v0lZy | what i want is that when a user dials #43, the useris connected to an application that catches all these #XX numbers. As long as that line holds, incoming calls to the users number are redirected to number 43 |
| 12:19.22 | v0lZy | when the user cuts the line |
| 12:19.33 | v0lZy | the redirection is killed |
| 12:19.35 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 12:21.29 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:21.30 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 12:21.33 | v0lZy | of course, the application is muted or something |
| 12:21.39 | v0lZy | so no sound is passed back or forth |
| 12:26.35 | *** join/#asterisk Dale_Noll (~dale@kamino.mcts.org) |
| 12:28.04 | v0lZy | Im thinking like this: http://pastebin.com/DXv2Rqc0 |
| 12:28.29 | v0lZy | but i dont know about the wait for hangup thing... |
| 12:28.53 | *** join/#asterisk JamKo (~JamKo@173-160-6-202-naples.hfc.comcastbusiness.net) |
| 12:29.17 | [TK]D-Fender | v0lZy, "wait for hangup from the user..." <- there is no "wait". It is immediate. |
| 12:29.38 | v0lZy | sorry, i ment this: http://pastebin.com/bwctf304 |
| 12:29.43 | [TK]D-Fender | v0lZy, and it isn't coming from the user, * hangs up |
| 12:30.22 | [TK]D-Fender | v0lZy, Why are you trying to set a value and then erasing it right after? |
| 12:30.23 | v0lZy | [TK]D-Fender: what i want to do is to call **XX |
| 12:30.29 | v0lZy | and as long as that call is holding |
| 12:30.43 | v0lZy | calls to the caller are forewarded to XX he inputed |
| 12:30.48 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 12:30.59 | v0lZy | well |
| 12:31.23 | [TK]D-Fender | v0lZy, You are not doing anything to keep the call sitting around.... |
| 12:31.23 | v0lZy | My phone has this ability to 'always forward' 'forward when busy' or 'forward when no pickup for X seconds' |
| 12:31.35 | v0lZy | but this is in the same menu as other settings which i dont want users to access |
| 12:31.41 | [TK]D-Fender | v0lZy, and if you're waiting for hangup, then you should actually using the hangup extension. |
| 12:31.45 | v0lZy | but the users do need a way to forward their phone when not at their desk |
| 12:32.17 | v0lZy | so what i need is not to hangup once the forwarding is activated |
| 12:32.26 | v0lZy | because i want that 'hangup action' to deactive the forwarding. |
| 12:32.49 | v0lZy | is that possible? do something on asterisk when a call is finished? |
| 12:32.59 | [TK]D-Fender | <[TK]D-Fender> v0lZy, and if you're waiting for hangup, then you should actually using the hangup extension. <---- |
| 12:33.29 | v0lZy | I'm relatively new to this |
| 12:33.54 | [TK]D-Fender | "asterisk standard extensions" <- very important section to read up on. |
| 12:35.33 | *** join/#asterisk e7e5 (~rudenko@188.134.2.33) |
| 12:35.57 | v0lZy | whats a hangup extension? |
| 12:36.13 | v0lZy | googling asterisk standard extensions |
| 12:36.15 | [TK]D-Fender | "h" <- |
| 12:37.32 | v0lZy | ah ok |
| 12:37.34 | v0lZy | so |
| 12:38.17 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 12:38.58 | v0lZy | exten => _**.,h,DBdel(CFI/${CALLERID(num)}) |
| 12:39.00 | v0lZy | ? |
| 12:39.30 | [TK]D-Fender | Nope. Keep reading... you just put that in the PRIORITY slot. |
| 12:40.34 | [TK]D-Fender | http://www.voip-info.org/wiki/view/Asterisk+standard+extensions |
| 12:40.46 | v0lZy | i've read that |
| 12:41.01 | v0lZy | but i think i lack other general knowledge to know where it applies to |
| 12:41.32 | *** join/#asterisk Z_God (~julius@2001:888:141f:0:d038:fba4:88e5:f22f) |
| 12:43.23 | [TK]D-Fender | http://www.voip-info.org/wiki/view/Asterisk+h+extension <--- first line says it pretty blatantly |
| 12:44.43 | v0lZy | i've read it |
| 12:44.51 | v0lZy | but i have no idea how to use it |
| 12:45.01 | [TK]D-Fender | When the call dies it goes there AUTOMATICALLY |
| 12:45.16 | *** join/#asterisk mazpe (~mazpe@c-174-61-76-85.hsd1.fl.comcast.net) |
| 12:45.23 | v0lZy | yeah but how do i tell it that when it gets there, it does DBdel(CFI/${CALLERID(num)}) |
| 12:45.46 | [TK]D-Fender | because that what you put in your exten........ |
| 12:47.04 | v0lZy | uh... |
| 12:47.57 | v0lZy | so |
| 12:48.08 | v0lZy | instead of _**.,h,DBdel(CFI/${CALLERID(num)}) |
| 12:48.14 | v0lZy | h,DBdel(CFI/${CALLERID(num)}) ? |
| 12:48.39 | [TK]D-Fender | exten => extension,priority,application |
| 12:48.41 | [TK]D-Fender | ^^^ |
| 12:50.02 | v0lZy | http://pastebin.com/qjz0WSHT |
| 12:50.03 | v0lZy | like so? |
| 12:50.07 | v0lZy | h,1,DBdel... |
| 12:50.09 | v0lZy | ? |
| 12:50.16 | v0lZy | ah |
| 12:50.22 | v0lZy | insted of exten? |
| 12:50.27 | v0lZy | huh.. no, wait. |
| 12:50.47 | [TK]D-Fender | YES |
| 12:51.25 | v0lZy | yes to instead of exten? |
| 12:51.29 | v0lZy | or yes to what i pasted? |
| 12:51.30 | pithagorians | <WIMPy> i add it like http://pastebin.com/JCmK8v7U where 069175372981 the DDI mapped to one of the internal numbers. so if i understand correctly it must have as caller the 069175372981 number but i still receive the Fallback one |
| 12:52.01 | [TK]D-Fender | now I'd make sure to do all of this is a separate context, because any time a call ends * looks for "h" relative to wherever the caller is in the dialplan and if this is in the open all you calls will hit it |
| 12:52.11 | [TK]D-Fender | exten => h,1,DBdel(CFI/${CALLERID(num)}) <- correct |
| 12:53.05 | v0lZy | it put this into a separate application |
| 12:53.18 | v0lZy | i only want it to be triggered with **XX |
| 12:53.21 | v0lZy | so |
| 12:53.23 | v0lZy | **01 |
| 12:53.27 | v0lZy | **00 etc |
| 12:53.44 | v0lZy | but only if the XX is valid. |
| 12:54.05 | WIMPy | pithagorians: Wrong format. The help clearly states that all caller IDs have to be unformatted. Use 4969... |
| 12:54.13 | [TK]D-Fender | do your holding & hangup exten in another context that you jump to then. |
| 12:54.31 | v0lZy | so like |
| 12:54.34 | v0lZy | **.. |
| 12:55.22 | v0lZy | [TK]D-Fender: hm... if i just make a context like this |
| 12:55.24 | v0lZy | hold on |
| 12:55.39 | *** join/#asterisk wtfitsme (~WTFitsME@asams.mserve.com) |
| 12:57.22 | v0lZy | http://pastebin.com/sRCWdLR8 |
| 12:57.24 | v0lZy | like this? |
| 12:57.47 | [TK]D-Fender | v0lZy, http://pastebin.com/2Kh3Pzwx |
| 12:58.16 | [TK]D-Fender | "h" needs to be really isolated from the rest of your dialplan |
| 12:58.27 | WIMPy | v0lZy: Have you thought about implementing CF in your dialplan? |
| 12:58.50 | [TK]D-Fender | WIMPy, That looks like exactly what he is already doing... |
| 12:59.07 | [TK]D-Fender | WIMPy, He just wants it so you have to leave the phone off the hook to be in effect |
| 12:59.48 | v0lZy | hm... |
| 13:00.12 | v0lZy | so this is basically 2 applicatiosn |
| 13:00.15 | WIMPy | Kind of. He needs to have an ongoing call. I was more thinking of forwarding at no answer. |
| 13:00.46 | v0lZy | WIMPy: i have people that go to the conference room |
| 13:00.53 | [TK]D-Fender | WIMPy, This is how he wants it to work though... certainly not a common approach |
| 13:00.54 | v0lZy | and or go on a break |
| 13:01.04 | v0lZy | and want to force the call immidiately to be forwared |
| 13:01.07 | v0lZy | not just on no answer. |
| 13:01.13 | [TK]D-Fender | "neglect protection" |
| 13:01.28 | v0lZy | neglect protection? |
| 13:01.44 | v0lZy | as in u mean anyone could walk by and hangup and destroy that forwarding? |
| 13:02.06 | [TK]D-Fender | v0lZy, No, this seems to be what your approach is for,. |
| 13:02.16 | *** join/#asterisk LiuYan (~LiuYan@222.125.129.197) |
| 13:02.28 | v0lZy | The point of this approach is in my users. |
| 13:02.30 | [TK]D-Fender | v0lZy, No other forwarding schemes I've ever seen require you to sit on a call |
| 13:02.41 | [TK]D-Fender | v0lZy, So you seem to compensating for twits |
| 13:02.48 | v0lZy | I have an app already here that if u dial 000023, it asks u to input the internal u want to forward to |
| 13:02.56 | v0lZy | and when u dial 000023 the second time, it disables the forward. |
| 13:03.03 | WIMPy | Yes, a button with indicatior seems like a nice idea. |
| 13:03.11 | v0lZy | but my users are literally too ignorant to remember that they have call forwarding on. |
| 13:03.32 | v0lZy | so then they are like |
| 13:03.46 | v0lZy | 'why is my coworkers phone ringing when they call me?!' |
| 13:04.07 | v0lZy | if i leave the phone on hook |
| 13:04.15 | v0lZy | but use a speaker |
| 13:04.27 | v0lZy | then i get a display on the phone as to where its forwarded to |
| 13:04.28 | v0lZy | like |
| 13:04.41 | v0lZy | **43 on the screen would mean its forwarded to 43 |
| 13:04.46 | v0lZy | or |
| 13:04.49 | v0lZy | i can do |
| 13:04.49 | [TK]D-Fender | v0lZy, Mayde you should just let users do forward on their phones.... |
| 13:05.00 | [TK]D-Fender | and let the phone deal with it |
| 13:05.04 | v0lZy | ***43 and it should forward to *43 which would be 43 voicemail. |
| 13:05.22 | kaldemar | you can send "desktop messages" to some phones, e.g. an indicator that shows on the phone display that a forward is active. |
| 13:05.33 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:1563:4b54:aa66:81d0) |
| 13:05.36 | v0lZy | i don tthink my phone supports that |
| 13:06.01 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:06.03 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:06.05 | v0lZy | [TK]D-Fender: thats the thing.. in this firmware, they put it in the same menu context as settings like username, pass etc |
| 13:06.10 | *** join/#asterisk zerohalo (~zerohalo@74-94-141-204-ma.hfc.comcastbusiness.net) |
| 13:06.29 | [TK]D-Fender | v0lZy, What crap phone is that? |
| 13:06.51 | WIMPy | And doesn;t the phone differ between user and admin stuff? |
| 13:07.31 | v0lZy | WIMPy, [TK]D-Fender : The VigorPhone 350 from DrayTek allows locking the iterface with an administrator password |
| 13:07.46 | v0lZy | and that blocks access to 'phone settings' and 'system settings' together |
| 13:07.53 | [TK]D-Fender | v0lZy, yeah, from the makers of crap roters.... |
| 13:07.58 | [TK]D-Fender | routers* |
| 13:08.10 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-xhdspkzvphsulajq) |
| 13:08.10 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 13:08.19 | v0lZy | actually the interface stuff is crap, the router itself is good by my judgement |
| 13:08.20 | v0lZy | anyway |
| 13:08.26 | v0lZy | it locks 2 menus at the same time |
| 13:08.27 | WIMPy | Hmm. Actually I don't know anyone who makes better plastic routers. |
| 13:08.35 | v0lZy | so if i unlock it.. it unlocks the sip settings unfortunately |
| 13:08.46 | WIMPy | But I have no idea about their routers voice capabilities. |
| 13:08.48 | v0lZy | draytek has very good routers if u ask me. |
| 13:08.56 | v0lZy | but i just have the phone here |
| 13:08.57 | [TK]D-Fender | v0lZy, Well if your users are dumb and so is your equipement... then you'll do what you have to do... |
| 13:08.58 | v0lZy | no router. |
| 13:09.24 | v0lZy | [TK]D-Fender: the online thing is simillar to what they've always had on ISDN here |
| 13:09.27 | v0lZy | some bosch system |
| 13:09.30 | v0lZy | Integral phones |
| 13:09.42 | v0lZy | there if u configure a redirect |
| 13:09.45 | v0lZy | then pick up the phone |
| 13:09.53 | v0lZy | it cancels the redirect |
| 13:10.09 | v0lZy | had to set it each day individually for a coworker that got fired |
| 13:10.30 | v0lZy | because the remaining person in that room kept bumping it with her ass and knocking it off hook |
| 13:10.43 | puzzled | lol |
| 13:10.46 | *** join/#asterisk pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger) |
| 13:10.46 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 13:10.48 | v0lZy | thus terminating the redirect .D :D |
| 13:10.49 | WIMPy | Pull the plug? |
| 13:11.02 | WIMPy | But a nice feature otherwise. |
| 13:11.12 | WIMPy | And again one that's impossible with SIP. |
| 13:11.14 | v0lZy | it has its pros i suppose |
| 13:11.16 | v0lZy | and its cons |
| 13:11.22 | v0lZy | depends on how skilled the user :D |
| 13:11.27 | *** join/#asterisk nighty- (~nighty@static-68-179-124-161.ptr.terago.net) |
| 13:11.41 | WIMPy | Sure |
| 13:11.50 | v0lZy | well one thing with this draytek phone |
| 13:11.57 | v0lZy | it has a speaker button and a headset button |
| 13:12.08 | v0lZy | and u enter the number then pick up the phone to call |
| 13:12.14 | kaldemar | WIMPy: that's not impossible with SIP. |
| 13:12.15 | v0lZy | or u pick up the phone then enter the number then press dial |
| 13:12.19 | v0lZy | or |
| 13:12.22 | v0lZy | u pres speaker and dial |
| 13:12.25 | v0lZy | anyway |
| 13:12.31 | [TK]D-Fender | v0lZy, everything we take for granted with a doze other makers... |
| 13:12.34 | v0lZy | i figure that i can apply that to redirects and activating voicemail |
| 13:12.38 | [TK]D-Fender | v0lZy, All of whom would be preferable.... |
| 13:13.11 | [TK]D-Fender | v0lZy, Congratulations ... your woes make Grandstream look GOOD. |
| 13:13.27 | v0lZy | :D |
| 13:13.49 | v0lZy | well... vigor350 is a pretty 'dumb' piece of equipment comapred to the aastra we have at the reception |
| 13:13.52 | v0lZy | now that phone i love. |
| 13:14.00 | v0lZy | the interface is more difficult |
| 13:14.08 | v0lZy | but its administration bliss |
| 13:17.28 | pithagorians | <WIMPy> when i use the format http://pastebin.com/fVEmt61h my call goes unknown. probably i don't understand what precisely have to do |
| 13:19.03 | v0lZy | hm |
| 13:19.05 | v0lZy | whats the code |
| 13:19.07 | v0lZy | like |
| 13:19.08 | WIMPy | pithagorians: Err, you don't want to set the caller ID to the number you called, I guess. |
| 13:19.11 | v0lZy | _**... |
| 13:19.13 | v0lZy | i mean |
| 13:19.16 | v0lZy | i need the user to press |
| 13:19.44 | pithagorians | <WIMPy> that's right. i need that this will be the DDI where from the call goes |
| 13:19.56 | v0lZy | <asterisk><asterisk><asterisk/digit><digit>[digit] |
| 13:20.01 | pithagorians | i think this is correct way |
| 13:20.02 | v0lZy | whats the pattern for that? |
| 13:20.09 | pithagorians | or i'm wrong ? |
| 13:20.15 | v0lZy | _**..X ? |
| 13:20.15 | WIMPy | pithagorians: Then put that back again. And what's the 2nd header about? |
| 13:20.24 | v0lZy | but the last digit is only optional... |
| 13:20.36 | [TK]D-Fender | v0lZy, "." = 1 or more characters of any kind |
| 13:21.14 | pithagorians | that's the one from the sipgate's documentation: For outgoing calls please set the desired caller id's in the E164-Format., meaning in international format without leading zeros or + sign, as new new header P-Preferred-Identity: |
| 13:21.14 | pithagorians | SipAddHeader(P-Preferred-Identity:<sip:492111234567@sipconnect.sipgate.de>) |
| 13:21.15 | pithagorians | To suppress the caller id you can do it like this |
| 13:21.15 | pithagorians | SipAddHeader(P-Preferred-Identity: <sip:492111234567@sipconnect.sipgate.de>) SipAddHeader(Privacy: id) |
| 13:21.40 | v0lZy | yeah |
| 13:21.42 | v0lZy | so |
| 13:21.50 | v0lZy | **.. and another one.. hold on |
| 13:22.07 | [TK]D-Fender | v0lZy, No ".." only a single dot |
| 13:22.13 | WIMPy | pithagorians: Yes, but you don;t want to suppress you number, do you? So why did you add that header? |
| 13:22.39 | v0lZy | hm... |
| 13:22.46 | [TK]D-Fender | v0lZy, "." 1 or LONGER. there isn't something to say "optionally one more".. You'd have to make 2 patterns |
| 13:22.55 | [TK]D-Fender | v0lZy, and not use "." |
| 13:22.58 | v0lZy | argh... |
| 13:23.04 | v0lZy | so |
| 13:23.15 | v0lZy | i mean what i would liek is like |
| 13:23.21 | v0lZy | **[*]XX actually |
| 13:23.32 | [TK]D-Fender | v0lZy, there is no "optional" |
| 13:23.32 | v0lZy | where the [*] is optional. |
| 13:23.39 | v0lZy | :( |
| 13:23.43 | [TK]D-Fender | 2 patterns |
| 13:24.08 | WIMPy | As [TK]D-Fender already said, there is nothig as "optional" in patterns. |
| 13:24.09 | v0lZy | either 2 patters |
| 13:24.10 | v0lZy | or |
| 13:24.12 | v0lZy | arbitrary right? |
| 13:24.15 | v0lZy | **. |
| 13:24.27 | pithagorians | <WIMPy> took it out |
| 13:24.53 | pithagorians | <WIMPy> left only exten => _XXXXXXXXX.,1,SipAddHeader(P-Preferred-Identity:<sip:492111234567@sipconnect.sipgate.de>) |
| 13:25.01 | v0lZy | i guess i can do *** to activate voiemail... |
| 13:25.07 | pithagorians | <WIMPy> and call comes from +492111234567 |
| 13:25.35 | [TK]D-Fender | v0lZy, You can make it whatever you want... |
| 13:25.51 | WIMPy | pithagoraians: See, it wasn't that hard :-) |
| 13:26.21 | v0lZy | [TK]D-Fender: how would i check the string |
| 13:26.26 | v0lZy | say the user inputs |
| 13:26.29 | v0lZy | **123 |
| 13:26.47 | v0lZy | i want to check if its longer than 2 characters after ** |
| 13:26.52 | v0lZy | and if it actually exists |
| 13:27.10 | [TK]D-Fender | v0lZy, "core show function DIALPLAN_EXISTS" |
| 13:27.13 | v0lZy | (dont want to enable forwarding to external number since that could cost us a lot) |
| 13:27.22 | [TK]D-Fender | v0lZy, "core show application chanisavail" |
| 13:27.48 | pithagorians | <WIMPy> yes, but i have bunch of DDIs and i want that the calls comes with each of this DDIs depending on how the DDI is mapped with the internal number |
| 13:27.52 | [TK]D-Fender | and you should look at your actualy forwarding code to see where you dump it into in the first palce. |
| 13:27.59 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 13:28.06 | [TK]D-Fender | place* |
| 13:29.12 | pithagorians | <WIMPy> this is how i do the mapping http://pastebin.com/nY3j16UC |
| 13:29.30 | WIMPy | pithagorians: Is it only about DDIs (iE converting abbreviated numbers to full ones) or something more flexible? |
| 13:29.33 | v0lZy | hm.. what id want ideally is to dump it either to the users own voicemail |
| 13:29.48 | v0lZy | or any internal number |
| 13:30.25 | [TK]D-Fender | v0lZy, If the number is expected to match a SIP devicename or something you can map then you could check that numebr against another function that gets info about them as well to validate... |
| 13:30.28 | *** join/#asterisk zerohalo (~zerohalo@67.103.230.22) |
| 13:30.54 | [TK]D-Fender | v0lZy, Or only allow forwarding if that # also matches a VM box #. |
| 13:31.01 | [TK]D-Fender | v0lZy, Tons of options to validate |
| 13:31.16 | pithagorians | <WIMPy> so far have to fix the simple issue after that maybe implement more flexible things. now need that the outgoing number will be the one corresponding to the DDI |
| 13:31.23 | v0lZy | well i suppose i dont want to forward to other users voicemail |
| 13:31.28 | v0lZy | so i can live with |
| 13:31.38 | v0lZy | *** to forward to the callers voicemail |
| 13:31.43 | v0lZy | erm |
| 13:31.58 | v0lZy | and **XX to an extension.. 2 patterens |
| 13:32.07 | v0lZy | and use the chanisavail for the XX |
| 13:36.15 | WIMPy | pithagorians: There is no relation. Just set the callerid in sip.conf as you want it to be. |
| 13:38.22 | pithagorians | <WIMPy> in sip.conf i have like callerid="John Doe 1" <7001> where 7001 is the internal number |
| 13:38.41 | v0lZy | hm |
| 13:39.37 | WIMPy | pithagorians: And what will the caller ID look like? Some prefix 7001 or something different? |
| 13:40.59 | pithagorians | when i call numbers internally than it comes John Doe or John Doe 1 or John Doe 2. depending on sip settings |
| 13:44.02 | v0lZy | [TK]D-Fender: whats the corresponding format of exten => h,1,DBdel(CFI/${CALLERID(num)}) to exten => _[0-9a-zA-Z*#]!,n,Goto(dizzy) |
| 13:44.16 | [TK]D-Fender | huh? |
| 13:44.27 | [TK]D-Fender | OH.. |
| 13:44.42 | v0lZy | my extensions.conf file has everything like that |
| 13:44.45 | [TK]D-Fender | exten => _[0-9a-zA-Z*#]! <- DANGEROUS for exactly that reason |
| 13:44.47 | *** join/#asterisk serafie (~erin@nat/digium/x-zxgontfosgtwivue) |
| 13:45.13 | v0lZy | i mean.. i thas everything written in both forms |
| 13:45.21 | v0lZy | i dont know the significance of this but probably because of the web interface? |
| 13:45.26 | [TK]D-Fender | anlost anything can hit that... it is something I would highly advise against doign except in the most controlled circumstances... |
| 13:45.33 | [TK]D-Fender | almost* |
| 13:45.50 | [TK]D-Fender | v0lZy, I also highly recommend against that interface as well.. |
| 13:45.56 | [TK]D-Fender | for separate reasons |
| 13:47.15 | v0lZy | my interface doesnt allow me to input |
| 13:47.18 | v0lZy | h,1,... |
| 13:47.22 | v0lZy | reports an error |
| 13:50.16 | [TK]D-Fender | I have no idea what you're inputting that into... this looks like raw dialplan to me that you would be doing directly in your own dialplan... |
| 13:50.22 | [TK]D-Fender | not vai a GUI |
| 13:50.25 | [TK]D-Fender | via* |
| 13:50.33 | v0lZy | yeah if i do the exten => thing |
| 13:50.40 | v0lZy | then thats the directly into the file |
| 13:50.49 | v0lZy | but if i make an application via the web interface |
| 13:51.01 | v0lZy | it automatically makes 2 versions in the same file |
| 13:51.13 | v0lZy | first the version with _[0-9a-zA-Z*#] |
| 13:51.19 | v0lZy | and after that logic, the one i discussed |
| 13:51.22 | v0lZy | i can show u an example |
| 13:51.43 | [TK]D-Fender | you should be doing this direct yourself then |
| 13:52.28 | v0lZy | if i use the web interface |
| 13:52.33 | v0lZy | this is the result i get in the extensions.conf file |
| 13:52.35 | v0lZy | http://pastebin.com/ny51uvKx |
| 13:52.54 | v0lZy | and i dont type the exten => |
| 13:52.59 | v0lZy | just everythign after exten => |
| 13:53.03 | v0lZy | so |
| 13:53.05 | v0lZy | i type |
| 13:53.24 | v0lZy | s,1,Answer(10) |
| 13:53.49 | v0lZy | s,n,voicemail({$EXTEN:1}) |
| 13:53.54 | v0lZy | and that u see in the pastebin is generated |
| 13:54.04 | [TK]D-Fender | v0lZy, That is messy crap.... you need to rethink how you define your pattern if you're going to use the GUI to start the coding for it. and as it requires another context... I highly advise against it... |
| 13:54.16 | [TK]D-Fender | v0lZy, don't use the GUi for this |
| 13:54.23 | v0lZy | but if i dont use the gui |
| 13:54.27 | v0lZy | then it doesnt show up in the guil |
| 13:54.30 | v0lZy | gui* |
| 13:56.28 | [TK]D-Fender | v0lZy, Well the other way adds crap. if "not working right" is an option.. then go for it |
| 13:57.00 | v0lZy | i think this is intentional |
| 13:57.24 | v0lZy | obviously the 2 patterns are different |
| 13:57.31 | v0lZy | 1 defines an extension |
| 13:57.34 | v0lZy | and the other one just the sequence |
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| 13:59.53 | v0lZy | then lower down in the file |
| 13:59.56 | v0lZy | it has the patteren defined like |
| 14:00.11 | v0lZy | exten => _*XX,n,Goto(DIALPLAN-APPLICATION-12196600124f56031ba4775,${EXTEN},1) |
| 14:00.13 | v0lZy | for example |
| 14:01.04 | v0lZy | under [internal] |
| 14:02.59 | *** join/#asterisk Gaiax (~Gaiax@unaffiliated/gaiax) |
| 14:03.24 | v0lZy | oh well |
| 14:03.26 | v0lZy | thanks for the help |
| 14:03.31 | v0lZy | illguess ill leave it until next time |
| 14:03.33 | v0lZy | have to catch a bus |
| 14:03.35 | v0lZy | bye! |
| 14:03.47 | v0lZy | and thanks again for helping me out |
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| 14:30.20 | mandla | tzafrir_laptop, hi, i am having a problem with loading firmwares for the astribank |
| 14:31.32 | tzafrir_laptop | mandla, Did you see my reply? Is that file missing? |
| 14:32.42 | mandla | tzafrir_laptop, yes thats the missing file. |
| 14:34.23 | tzafrir_laptop | Copy it, and run /usr/share/dahdi/xpp_fxloader load |
| 14:35.02 | mandla | tzafrir_laptop, thanx, it worked. But why was it deleted in the 1st place? |
| 14:36.39 | tzafrir_laptop | There are some stupid distribution limitations on it, which is why it's not included |
| 14:37.41 | mandla | ok, thanks. |
| 14:37.52 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 14:37.52 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 14:38.06 | tzafrir_laptop | The file in question: http://updates.xorcom.com/astribank/hwec/ |
| 14:39.18 | mandla | I mean like, my asterisk server has been working fine until a recent power cut, then when i boot it up i get that error. |
| 14:39.27 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 14:40.21 | *** join/#asterisk eduzimrs (~eduzimrs@201.22.86.124.static.gvt.net.br) |
| 14:40.51 | eduzimrs | Hi, how do u create an extension that matches an MAC ADDR ? |
| 14:42.57 | [TK]D-Fender | eduzimrs, _[a-z,A-Z,0-9][:][a-z,A-Z,0-9][:][a-z,A-Z,0-9][:][a-z,A-Z,0-9][:][a-z,A-Z,0-9]....... etc |
| 14:43.06 | [TK]D-Fender | well spaced for pairs... |
| 14:43.20 | [TK]D-Fender | it's a pattern like anything else |
| 14:43.40 | [TK]D-Fender | Or make a dangerously global match and validate it in the exten |
| 14:43.57 | eduzimrs | [TK]D-Fender: can i use like grep patterns ? |
| 14:44.20 | [TK]D-Fender | eduzimrs, no. *'s patterns are well documented. |
| 14:45.04 | eduzimrs | [TK]D-Fender: sure it was a doubt, tks for that |
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| 14:51.15 | eduzimrs | [TK]D-Fender; if i use "-" instead ":" should i scape "\-" ? |
| 14:51.25 | [TK]D-Fender | [-] |
| 14:51.29 | eduzimrs | ok |
| 14:52.03 | WIMPy | - is only special inside []. So I don't see any need for [-]. |
| 14:53.52 | joel | ^ |
| 14:58.37 | eduzimrs | "," are needed? |
| 14:59.03 | eduzimrs | its not matching |
| 14:59.25 | WIMPy | No. Only if you want to match commas as well. |
| 14:59.26 | leifmadsen | what is the pattern match? |
| 14:59.32 | WIMPy | Did you double the digits? |
| 14:59.57 | *** join/#asterisk qakhan (~qakhan@180.178.136.103) |
| 15:00.12 | eduzimrs | yeap, i did - exten => _[a-zA-Z0-9]-[a-zA-Z0-9]-[a-zA-Z0-9]-[a-zA-Z0-9]-[a-zA-Z0-9]-[a-zA-Z0-9],1,Dial(SIP/${EXTEN},35) |
| 15:00.21 | leifmadsen | why hyphen? |
| 15:00.25 | eduzimrs | MAC match |
| 15:00.26 | leifmadsen | wouldn't it be a colon? |
| 15:00.39 | leifmadsen | and if it's mac, you just need to match a-fA-F |
| 15:00.40 | WIMPy | You did not double the digits. |
| 15:00.42 | leifmadsen | not a-zA-Z |
| 15:00.48 | leifmadsen | and what WIMPy said |
| 15:01.03 | joel | does [] not need a quantifier? |
| 15:01.14 | leifmadsen | exten => _[a-fA-F0-9][a-fA-F0-9]-[a-fA-F0-9][a-fA-F0-9]-[a-fA-F0-9][a-fA-F0-9]... etc |
| 15:01.21 | leifmadsen | [...] matches a single character |
| 15:01.31 | leifmadsen | anything contained within those braces |
| 15:01.34 | leifmadsen | for a single position |
| 15:02.27 | WIMPy | joel: This is not regex, even if it looks similar. |
| 15:02.41 | joel | WIMPy: I'm aware. |
| 15:02.43 | leifmadsen | it is just literal matching of characters within the braces |
| 15:02.57 | leifmadsen | you can't use a quantifier with the method described above |
| 15:03.05 | joel | lame <3 pcre. |
| 15:03.14 | leifmadsen | this is dialplan |
| 15:05.18 | WIMPy | Yes, a little more sophisticated patterns would be nice. |
| 15:05.37 | WIMPy | But there are tons of other things that would be nice. |
| 15:05.53 | leifmadsen | if you need something more complex, there is always AGI() :) |
| 15:06.15 | WIMPy | But you can't do extension matching with that. |
| 15:06.39 | eduzimrs | @leifmadsen: exten => _[a-fA-F0-9]-[a-fA-F0-9]-[a-fA-F0-9][a-fA-F0-9]-[a-fA-F0-9][a-fA-F0-9],1,Dial(SIP/${EXTEN},35) doesnt match |
| 15:06.40 | leifmadsen | match everything and pass it to the AGI |
| 15:06.49 | joel | eduzimrs: earth to you |
| 15:06.50 | leifmadsen | eduzimrs: doesn't match what |
| 15:06.55 | joel | eduzimrs: how many sets of brackets do you have? |
| 15:07.04 | joel | eduzimrs: how many characters are there in a mac? |
| 15:07.10 | leifmadsen | eduzimrs: you're still doing it wrong |
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| 15:07.21 | eduzimrs | owwww |
| 15:07.23 | leifmadsen | you need 6 sets of double brackets |
| 15:07.27 | eduzimrs | yeap |
| 15:07.39 | eduzimrs | i landed now |
| 15:07.40 | eduzimrs | sry |
| 15:08.09 | eduzimrs | something like exten => _[a-fA-F0-9][a-fA-F0-9]- ..... |
| 15:08.22 | qakhan | i m getting this message Got SIP Response "Decline" from 192.168.5.3 |
| 15:08.25 | qakhan | please help me |
| 15:08.35 | leifmadsen | qakhan: other side declined your call |
| 15:08.47 | WIMPy | eduzimrs: Don;t worry, we all get issues with trees and forests in larger amounts :-) |
| 15:09.04 | qakhan | @leifmadsen i have queue with 4 agents |
| 15:09.23 | qakhan | when calls goes to agents then i receive this message |
| 15:09.31 | [TK]D-Fender | leifmadsen, He's posted that one single line for days with no background, no debug or anything... |
| 15:09.31 | leifmadsen | agent is rejecting the call then |
| 15:09.33 | *** join/#asterisk b3nt_pin (~b3ntpin@142.162.121.80) |
| 15:10.46 | qakhan | @leifmadsen every agent receive first call then i receive this message and after 1 min that agents receive calls |
| 15:11.02 | leifmadsen | not enough information to go on, so I have no idea |
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| 15:11.24 | *** mode/#asterisk [+o pabelanger] by ChanServ |
| 15:13.15 | eduzimrs | WIMPy: yeap could this pattern be placed into FILTER func ? exten => _X.1,Dial(SIP/${FILTER([a-fA-F0-9][a-fA-F0-9]-...,${EXTEN})}) would work ? |
| 15:14.03 | WIMPy | eduzimrs: Why do you want to use FILTER instead of an explicit extension? |
| 15:14.24 | eduzimrs | just i doubt, i wont use |
| 15:14.53 | qakhan | anyone else can help me |
| 15:15.09 | eduzimrs | WIMPy, Do u use polycom`s ? |
| 15:15.09 | qakhan | i have queue with 4 agents |
| 15:15.19 | qakhan | every agent receive first call then i receive this message and after 1 min that agents receive calls |
| 15:16.01 | [TK]D-Fender | qakhan, Your phone is rejecting the call. This is not an asterisk problem. Look at your phone. There is nothing for us to fix. All this aside from the fact you weren't even showing the full call. |
| 15:16.56 | *** join/#asterisk mac|gyve1 (~sanderrij@2001:960:66e:0:21c:b3ff:feb1:ba10) |
| 15:17.07 | eduzimrs | Anyone has problem when setting SRTP media to a polycom Sound Point IP series? |
| 15:17.44 | WIMPy | eduzimrs: Nope. They're very rare over here. |
| 15:17.46 | mac|gyve1 | I changed the INVITE string that goes to my trunk previously, I need to do it again. What's the best place to do this? |
| 15:19.44 | *** join/#asterisk din3sh (din3sh@41.136.80.67) |
| 15:20.46 | [TK]D-Fender | mac|gyve1, wherever you did it last |
| 15:20.50 | *** join/#asterisk Naikrovek (~uname@unaffiliated/naikrovek) |
| 15:21.15 | mac|gyve1 | thanks man |
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| 15:52.28 | rgsteele | I'm examining a newer installation (trying to make a custom app work as it did with an older Asterisk install), and I noticed that a 'core show channel <channelname>' is missing a few CDR variables. However, cdr_custom.conf is defined the same way for both hosts. |
| 15:52.35 | rgsteele | In particular, both have these three defined: "${CDR(start)}",${CDR(answer)}","${CDR(end)}", but on the newer install only 'start' is listed in the 'core channel show <channelname> |
| 15:53.37 | rgsteele | Is there something else that needs to be configured to expose that data/those variables? |
| 15:56.50 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v023-129.mobile.uci.edu) |
| 15:58.11 | rgsteele | Oh, nevermind - it looks like "disposition" is set to "No Answer" for some reason... |
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| 16:32.42 | Ice_Strike | What the minimum server spec do I need for 60 concurrent calls with recording (wav)? |
| 16:32.52 | Ice_Strike | call via VOIP |
| 16:33.24 | WIMPy | Your average wrist watch. |
| 16:34.21 | Ice_Strike | WIMPy talking to me? |
| 16:34.36 | WIMPy | yes |
| 16:34.43 | Ice_Strike | lol |
| 16:35.12 | WIMPy | Seriousely: There are many many factors, but just the number of simultanepus calls is none. |
| 16:35.42 | Ice_Strike | Hmmm |
| 16:35.57 | *** join/#asterisk mazpe (~mazpe@c-174-61-76-85.hsd1.fl.comcast.net) |
| 16:36.04 | Ice_Strike | I wanna but dedicated server, I just want to make sure it don't overkill with the spec |
| 16:36.10 | Ice_Strike | buy* |
| 16:37.40 | WIMPy | Depending on what you plan to do, Asterisk might not have anythign to do with the calls between setup and teardown. |
| 16:38.07 | WIMPy | In which case some old plastic router or AP would indeed be good enough. |
| 16:38.19 | kaldemar | *recording* |
| 16:38.56 | Ice_Strike | WIMPy VOIP Provider located someone in the UK and the server as well. I will be about 60 concurrent calls during 24/7 for 1 month. |
| 16:39.09 | Ice_Strike | Using Ulaw and recording into wav |
| 16:39.24 | WIMPy | What will you be doing with those calls? |
| 16:39.46 | WIMPy | And as said, the number of concurrent calls on itself is completely meaningless. |
| 16:39.58 | WIMPy | The number of call attempts would have more impact. |
| 16:41.01 | Ice_Strike | I see |
| 16:41.47 | Ice_Strike | Im just thinking should I get VPS or dedicated server |
| 16:42.19 | WIMPy | Do you want recording? local applications? DTMF features? Transcoding? |
| 16:43.01 | Ice_Strike | WIMPy DTMF + Recording WAV |
| 16:43.05 | Ice_Strike | what is local applications? |
| 16:43.16 | Ice_Strike | Transcoding maybe converting to MP3 at the end yea |
| 16:43.46 | WIMPy | applications like voicemail or IVR stuff, e.g. |
| 16:43.53 | Ice_Strike | nah nothing like that |
| 16:44.17 | Ice_Strike | will be using AMI and AGP |
| 16:44.28 | Ice_Strike | with Apache, PHP, mySQL. |
| 16:44.29 | WIMPy | Ok, recording requires som I/O and mp3 encoding require some CPU. |
| 16:45.22 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 16:45.47 | Ice_Strike | Hmm yes |
| 16:46.10 | Ice_Strike | Intel Core2Quad 2.4Ghz, 4GB PC2-5400 DDR2, 160GB SATA HDD |
| 16:46.14 | Ice_Strike | Overkill? |
| 16:47.44 | Ice_Strike | or even cheaper with VPS: 2048MB, 75GB (Raid 10) |
| 16:50.26 | *** join/#asterisk talntid (~talntid@173-160-189-58-Washington.hfc.comcastbusiness.net) |
| 16:50.32 | [TK]D-Fender | It isn't just a question of CPU, it's a question over process dedication. many VPS don't prioritize so you'll get choppy recordings, etc |
| 16:51.11 | talntid | Does anyone know where I can find a difinitive answer on the difference between Restricted, Unavailable, and Private callerid's, when they are incoming? Not really * related, but I have been searching and cannot find.. figured maybe someone here had ran into the same question... |
| 16:51.13 | *** join/#asterisk mazpe (~mazpe@c-174-61-76-85.hsd1.fl.comcast.net) |
| 16:53.17 | WIMPy | talntid: restricted/private means the calling user wanted to call anonymousely. Unavailable means it's unavailable due to interworking. |
| 16:55.10 | talntid | gotchs |
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| 17:52.03 | cj | stretches |
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| 18:22.50 | fubada | hi, will "messages => notice,warning,error |
| 18:22.58 | fubada | in logger.conf log dropped calls? |
| 18:23.11 | Qwell | It will log notice, warning, and error messages. |
| 18:23.24 | fubada | does that include a dropped call |
| 18:23.46 | Qwell | It depends on why it was "dropped". |
| 18:24.10 | fubada | any unexpected non-0 exit is waht im trying to find |
| 18:24.33 | fubada | we had a lot of dropped call issues with asterisk 1.6, so i upgraded to 10.4 |
| 18:24.41 | fubada | and there was a random dropped call on 10.4 |
| 18:24.46 | fubada | which im trying to track down |
| 18:25.02 | Qwell | You really aught to be looking at everything. No need to limit yourself to those log levels. |
| 18:25.34 | fubada | its difficult because all i have to go on is "just had a dropped call on outgoing at 1:31pm" |
| 18:26.08 | singler | fubada: tcpdump ftw :) |
| 18:27.36 | talntid | fubada, just FYI.... |
| 18:28.32 | talntid | I made it so that when people have dropped calls, bad call quality, etc... they pick up and dial extension "BAD" and it it takes the call recording, asterisk logs, the correct time & date, and a tracert to my service provider, and emails it to me |
| 18:29.08 | talntid | this is for a call center, where there is a lot of room for complaints... but it may be something you might want to look into.. |
| 18:29.28 | fubada | interesting |
| 18:29.45 | fubada | but thats after the drop right? |
| 18:29.49 | talntid | yes |
| 18:29.58 | talntid | Makes it a lot easier to diagnose it, as I can hear what happened, and have the data |
| 18:30.28 | fubada | thank you |
| 18:33.45 | *** join/#asterisk urvg4 (~ducdmann@host81-149-39-60.in-addr.btopenworld.com) |
| 18:35.33 | blizzow | I was looking for an alternative to gotomeeting. I was hoping to integrate something that allows people to dial a meetme conference and then have the person visit a website to share desktop. Are there any asterisk plugins that might do something of that sort, or does anyone here know something? The googs isn't so hot with "gotomeeting alternatives." |
| 18:36.34 | talntid | blizzow, if you figure that out, can you email me talntidtsi@gmail.com ? :) |
| 18:36.41 | talntid | I have tried join.me |
| 18:36.52 | blizzow | talntid, I'm looking at bigbluebutton.org right now. |
| 18:36.52 | talntid | but it doesn't work for Linux, which is what my thin clients are.. |
| 18:36.59 | Qwell | blizzow: BigBlueButton is supposedly kinda like that |
| 18:37.00 | talntid | but, that might work for you, if you are using Windows.. |
| 18:38.08 | blizzow | From their site, It looks like bbb integrates with freeswitch in some way. |
| 18:38.29 | talntid | and Linux |
| 18:38.40 | fubada | Whats a good alternative to Symphony app? |
| 18:38.45 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 18:39.54 | pabelanger | +1 for BBB |
| 18:40.13 | Qwell | We never could get it working though... |
| 18:40.48 | pabelanger | me and russellb had it working on his desktop, but when he used is laptop it would not work. |
| 18:40.51 | pabelanger | I blame java |
| 18:56.20 | *** topic/#asterisk by mjordan -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.4.2 (2012/05/30), 1.8.12.2 (2012/05/30), DAHDI-linux 2.6.1 (2012/04/20), DAHDI-tools 2.6.1 (2012/04/20), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
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| 19:05.24 | *** join/#asterisk schultza (~allen@rc1.rcherbals.com) |
| 19:05.35 | schultza | what is the kernel module for the clock? |
| 19:05.43 | schultza | ztdummy.. or something newer with dahdi |
| 19:05.49 | Qwell | Just dahdi |
| 19:06.51 | schultza | dahdi will do the clock that ztdummy used to do? |
| 19:17.16 | *** join/#asterisk micols (~t@rlogin.dk) |
| 19:20.45 | Digweed | Didn't dahdi use the USB port for timing or something? When there's no hardware card installed. |
| 19:23.14 | [TK]D-Fender | that was the OLD Zaptel. Way old... and only for 2.4 Kernels |
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| 19:34.28 | Digweed | Ok :) |
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| 20:06.35 | *** join/#asterisk l0st-soul (~renaud@109.61-78-194.adsl-static.isp.belgacom.be) |
| 20:06.59 | l0st-soul | hello |
| 20:07.25 | l0st-soul | anyone knows a way to check a sip peer existence with a dialplan command ? |
| 20:07.55 | l0st-soul | (kind of like voicemailexists, but for sip peers .. sippeerexist?) |
| 20:08.31 | l0st-soul | or is the only way to just dial it and check return code |
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| 20:18.14 | puzzled | l0st-soul: there is a solution for that. just can't remember what it was. if I remember I'll let you know |
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| 20:23.28 | [TK]D-Fender | l0st-soul, "core show function SIPPEER" |
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| 20:24.12 | puzzled | l0st-soul: SIPPEER function |
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| 20:24.30 | puzzled | heh TK beat me to it. not a surprise :) |
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| 20:44.37 | l0st-soul | thanks :) |
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| 21:10.03 | iunruh | is it possible through AGI or AMI to programmically transfer a channel to a meetme conference? |
| 21:10.30 | WIMPy | yes |
| 21:11.46 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:11.49 | iunruh | I basically want to write something that, whenever someone calls in, it parks the call instantly. then later, when it knows which conference to add the person to, transfer that channel to the conference |
| 21:12.34 | iunruh | which API would I use to do this? AGI or AMI? I always get confused on what each one's purpose is |
| 21:12.46 | WIMPy | Whatever "park" may mean in that context. Just wait, I suppose. |
| 21:12.59 | iunruh | Right |
| 21:13.32 | WIMPy | As you have a call and ar in the dialplan, you can use either. |
| 21:14.20 | [TK]D-Fender | iunruh: "when it knows"? Knows how? |
| 21:14.49 | iunruh | it asks a web service |
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| 21:15.06 | iunruh | I'm not using asterisk as a PBX as much as a VoIP provider |
| 21:15.13 | timholum | anyone know how to make an agi script to run but not wait until it complete's to continue the call? |
| 21:15.36 | [TK]D-Fender | iunruh: timNot possible |
| 21:15.40 | [TK]D-Fender | timNot possible |
| 21:15.44 | [TK]D-Fender | timholum: ^ |
| 21:15.51 | timholum | hmm, ok |
| 21:15.53 | [TK]D-Fender | timholum: All dialplan is linear |
| 21:16.09 | [TK]D-Fender | timholum: You can shell out to a backgrounded script, but not AGI |
| 21:16.15 | [TK]D-Fender | AGI = interactive |
| 21:17.03 | [TK]D-Fender | iunruh: You can shove your inbound calls into a loop awaiting being redirected or whatever else you'd like to do to them after |
| 21:17.51 | timholum | how do I exicute a background script? I dont need the interactivity, I basicly need to hit a url when a call comes in |
| 21:18.12 | iunruh | [TK]D-Fender: would that just be a simple dialplan and then I use something else to handle the "redirecting"? |
| 21:19.56 | [TK]D-Fender | iunruh: AMI |
| 21:20.00 | iunruh | ah |
| 21:20.04 | [TK]D-Fender | for the redirect |
| 21:20.35 | iunruh | and then can I use AMI to redirect the call later? |
| 21:20.46 | iunruh | or is it a one-shot redirect? |
| 21:21.20 | WIMPy | AGI is like external dialplan. AMI can do anything at any time. |
| 21:21.40 | iunruh | okay, that really helps |
| 21:23.25 | iunruh | so where do I "hold" the call until I use AMI to redirect it? |
| 21:23.33 | iunruh | in a queue? do I use call parking? |
| 21:23.43 | [TK]D-Fender | iunruh: Yes, redirect at any time |
| 21:23.56 | WIMPy | What do you want to happen? |
| 21:24.06 | WIMPy | You can just do nothing if you like. |
| 21:24.47 | iunruh | okay, that makes sense |
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