| 00:01.44 | igcewieling | slav3_kitten: it is never time to use slackware |
| 00:04.57 | slav3_kitten | igcewieling, why not? compile all thethings! |
| 00:05.12 | igcewieling | you are thinking of Gentoo |
| 00:05.30 | slav3_kitten | no i'm thinking slack back in the early 90's |
| 00:05.32 | igcewieling | Gentoo: FreeBSD but with a Linux kernel. |
| 00:05.33 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.118) |
| 00:05.58 | slav3_kitten | then again every linux distro was like that back then |
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| 00:11.03 | slav3_kitten | i miss the early 90's |
| 00:11.26 | slav3_kitten | in hindsight technology sucked, but it was the best thing since sliced bread |
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| 02:07.27 | talntid2 | http://pastebin.com/htZ9NPYu |
| 02:07.38 | talntid2 | more likely asterisk or postgres mad at me? |
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| 02:08.38 | ChannelZ | jesus, maybe. |
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| 02:09.37 | talntid2 | hm, restarting * resolved it |
| 02:09.42 | talntid2 | jesus told me to. |
| 02:09.45 | ChannelZ | Anyway SQL is mad, because Asterisk is telling it weird things. |
| 02:09.49 | ChannelZ | So, both. |
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| 02:10.13 | Quest | if iam following http://wiki.freepbx.org/pages/viewpage.action?pageId=1409028 but have already installed asterisk by apt-get repos. what do i need to skip in that link? |
| 02:10.20 | Quest | <PROTECTED> |
| 02:10.47 | ChannelZ | depends |
| 02:11.02 | ChannelZ | You don't need DAHDI or LIBPRI unless you're dealing with telephony hardware |
| 02:11.03 | Quest | ChannelZ, on? |
| 02:11.20 | Quest | ChannelZ, iam .. using ip phones for voip |
| 02:11.37 | ChannelZ | no, I mean interface hardware |
| 02:11.43 | Quest | hm ok |
| 02:11.54 | ChannelZ | like TDM cards |
| 02:12.02 | ChannelZ | for talking POTS or PRI |
| 02:12.11 | Quest | k |
| 02:12.22 | Quest | anything else i need to change/skip |
| 02:12.24 | Quest | ? |
| 02:12.28 | ChannelZ | anyway yes you can just compile and see what happens. There might be some dev dependencies you're missing |
| 02:13.24 | Quest | repo install is better always? |
| 02:13.45 | ChannelZ | not necessarily no |
| 02:14.16 | Quest | i have installed asterisk. so do i need to use .tar |
| 02:14.18 | Quest | again. |
| 02:14.21 | ChannelZ | The repos are usually outdated, which may or may not matter to some |
| 02:14.59 | ChannelZ | What do you mean you have installed asterisk |
| 02:15.30 | ChannelZ | you have it on and running already and you're just trying to put FreePBX on top of it? (this isn't a FreePBX channel by the way) |
| 02:15.49 | Quest | ChannelZ, i already did a sudo apt-get install asterisk |
| 02:16.04 | ChannelZ | so then presumably you'd start at "Install and Configure FreePBX" |
| 02:16.06 | Quest | ChannelZ, correct |
| 02:16.15 | Quest | ya |
| 02:16.36 | ChannelZ | and then go talk to #freepbx when it all goes sideways |
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| 02:44.08 | Quest | Enter your USERNAME to connect to the 'asterisk' database: i can give any user name? |
| 02:49.04 | igcewieling | ChannelZ: it is best not to feed the troll. |
| 02:49.15 | igcewieling | Quest: Ask on #freepbx |
| 02:49.39 | Quest | isnt that an asterisk question? |
| 02:49.54 | igcewieling | Quest: you are rapidly losing any possible credibility here. Ask the questions on the appropriate channel. |
| 02:50.03 | Quest | igcewieling, to whom you are addressing as troll? |
| 02:50.13 | igcewieling | Quest: there are virtually no Asterisk questions when using FreePBX. |
| 02:50.22 | igcewieling | Asterisk DOESN"T EVEN USE A DATABASE BY DEFAULT |
| 02:50.35 | Quest | Mr. igcewieling , in what capacity are you judging me and commenting me. and why using caps? are you shouting? |
| 02:50.53 | Quest | igcewieling, you could have said it politely |
| 02:50.59 | igcewieling | Quest: you repeatedly ask FreePBX questions on this channel. |
| 02:51.00 | Quest | I can read english |
| 02:51.06 | Quest | igcewieling, you could have said it politely |
| 02:51.21 | Quest | iam a new person. unaware. and not perfect |
| 02:51.26 | igcewieling | Quest: you have been referred to #Freepbx multiple times in the past week. Apparently politeness doesn't work. |
| 02:51.30 | Quest | thanks. take care nextime |
| 02:51.32 | *** part/#asterisk Quest (~kubuntu@139.190.72.29) |
| 02:51.49 | igcewieling | He will be back tomorrow asking the same stuff. |
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| 03:47.35 | terrayi | Hello. somebody available for help on asterisk + tdm2400p ? |
| 03:47.54 | terrayi | I am quite lost when asterisk box suddenly not working - not receiving any calls. it just hangs up right away if I call from my mobile phone. |
| 03:50.29 | terrayi | I get "ProSLIC on module 10 failed to powerup within 504 ms (0 mV only)" and "FAILED FXS (FCC)" for module 0 & 10 but no errors on FXO modules |
| 03:51.00 | terrayi | no configuration changed but it suddenly not working .. :( |
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| 04:39.16 | linocisco | hi all |
| 04:43.33 | terrayi | no one answers |
| 04:43.53 | terrayi | anyone awaken yet? |
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| 05:26.39 | ChannelZ | hardware failure? did it get zapped? dunno |
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| 07:02.44 | mirela666 | Good morning :) |
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| 07:49.17 | E-bola | I have a problem with a recently deploeyd asterisk installation (1.8.21). Call swork fine for both inc and outgoing. However when a user uses *8 to pickup a ringing extension, the call is connected and works fine for ~5seconds, but is the disconnected. |
| 07:49.24 | E-bola | Anybody ever seen this behavior? |
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| 08:20.17 | terrayi | hello? |
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| 09:20.27 | srp_ | Hi, I'm trying to install asterisk on a virtual machine in my private opestack cloud.. When I compile asterisk 11.2 from source (./configure, make, make install) as specified in the asterisk book and run it, it gives me the error: "Illegal instruction (core dumped)" |
| 09:20.35 | *** join/#asterisk HmdP_Mobile (~HmdP_Mobi@D9799130.cm-3-2c.dynamic.ziggo.nl) |
| 09:20.39 | srp_ | any ideas whats going wrong? |
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| 10:20.58 | tparcina | How to reload voicemail.conf configuration? |
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| 10:21.29 | mirela666 | tparcina: voicemail reload from CLI> |
| 10:21.38 | mirela666 | tparcina: CLI>voicemail reload |
| 10:21.38 | tparcina | mirela666: Thank you. |
| 10:23.43 | mirela666 | srp_: <try to see in /var/log/asterisk/full what's the last few things that were done before dump |
| 10:27.34 | srp_ | mirela666: one sec.. on it.. |
| 10:27.58 | mirela666 | srp_: I think it has something to do with prpocessor type 32-64 bit and version of asterisk you try to compile |
| 10:28.06 | mirela666 | srp_: http://forums.digium.com/viewtopic.php?f=1&t=83743 |
| 10:29.30 | srp_ | mirela666: i basically installed it on a ubuntu 64bit server image.. the instance specs are quite low.. 512MB RAM, 1 VCPU, 7 GB root Disk.. hope that is not creating any problems.. checking the forum.. |
| 10:31.24 | srp_ | mirela666: there is no file names full in /var/log/asterisk |
| 10:31.37 | srp_ | mirela666: there are only cdr, cel related folders there |
| 10:36.16 | mirela666 | srp_: https://issues.asterisk.org/jira/browse/ASTERISK-20128 |
| 10:36.17 | LieutPants | [ASTERISK-20128] [Status: Open] [Assigned: kmoore] Virtualized asterisk.org 1.8.14.0 no longer runs in a KVM virtualized environment. Compiles without error, but fails with Illegal instruction on launch Regression since 1.8.13.0 Last good 1.8.12.2 - https://issues.asterisk.org/jira/browse/ASTERISK-20128 |
| 10:36.32 | mirela666 | LieutPants: :) |
| 10:37.00 | srp_ | mirela666: LieutPants: but i'm running asterisk 11.2.. does that problem still exists? |
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| 11:12.09 | mirela666 | srp_: i'm not sure, i'm still on 1.8.12 :) |
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| 11:22.33 | srp_ | mirela666: I think i'll run one of the freepbx or asteriskNOW iso on my cloud and see.. let me see if that works.. :) |
| 11:22.50 | srp_ | mirela666: btw.. 1.8.12 works fine on the cloud instance? |
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| 11:25.23 | mirela666 | I had no problems |
| 11:26.21 | mirela666 | srp_: try to install from repository and see if that will work, you might get older version but should be stable |
| 11:27.53 | mirela666 | srp_: with apt-cache show asterisk i get Version: 1:1.8.13.1~dfsg-1ubuntu2 which is ok for me :) |
| 11:28.10 | mirela666 | srp_: try to find repo for 11 |
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| 11:39.05 | kaldemar | .. |
| 11:40.17 | WIMPy | : |
| 11:40.36 | E-bola | ; |
| 11:40.43 | srp_ | mirela666: okay |
| 11:41.17 | srp_ | mirela666: if i install from apt, can i configure like i usually do in menuselect? |
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| 12:11.11 | ayrus | Hi, how to get the customer input of IVR, I want to tell the agents; that how the customer reached to them from the IVR. Kindly help |
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| 12:19.26 | [TK]D-Fender | ayrus, "core show application dial" <- |
| 12:22.19 | ayrus | [TK]D-Fender: thank you. |
| 12:25.17 | ayrus | [TK]D-Fender: how to grep only the "key" pressed by customers? |
| 12:26.00 | [TK]D-Fender | ayrus, It's your IVR... you clearly already created the entens that matched their input... you already HAVE it... |
| 12:28.29 | ayrus | [TK]D-Fender: yes, I have created, but the agents are limited, so at last all the calls land to the same agents, but they don't know whats the problem of the customer. Like I have created, "If you got broken Item Press 3" and when the customer press 3 the call land to same agents. So he don't know, what the reason behind the call. Again agent need to ask to the customer, why he called us. |
| 12:29.20 | [TK]D-Fender | ayrus, And in Dial() you can specify something to PLAYBACK to the person who is called which could be a recording telling them that |
| 12:30.01 | ayrus | ok. |
| 12:33.28 | ayrus | [TK]D-Fender: right. Actually I'm using distro FreePBX & the recording is played of total hold time of the customer. I think, the same way I can configure the Key pressed by the customer. Am I correct? |
| 12:34.44 | [TK]D-Fender | ayrus, This is the wrong place to be asking. #freepbx <- |
| 12:34.59 | [TK]D-Fender | ayrus, GUI's are supported in their own channels |
| 12:35.20 | ayrus | [TK]D-Fender: ok, now I will not ask the question regarding the FreePBX |
| 12:35.44 | ayrus | [TK]D-Fender: Thank you. :) |
| 12:42.52 | *** join/#asterisk luissimoes (5f887f7f@gateway/web/freenode/ip.95.136.127.127) |
| 12:43.30 | luissimoes | hello |
| 12:44.25 | luissimoes | I'm doing a thesis on Open Source Project Management. Can u help me to find a contact to make a small interview about management best practices? |
| 12:49.29 | [TK]D-Fender | luissimoes, You might want to pick a project without commercial ties then.... |
| 12:49.55 | [TK]D-Fender | luissimoes, Because you're starting with baggage asking about Asterisk. |
| 12:50.47 | luissimoes | yeah, but it can also have commercial ties |
| 12:52.03 | *** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com) |
| 12:52.05 | luissimoes | it's specially to understand project management approach |
| 12:52.06 | [TK]D-Fender | luissimoes, www.digium.com <- contact them about getting in touch with the community liason. |
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| 12:52.17 | luissimoes | thanks ;) |
| 12:52.38 | E-bola | I have a problem with a recently deploeyd asterisk installation (1.8.21). Call swork fine for both inc and outgoing. However when a user uses *8 to pickup a ringing extension, the call is connected and works fine for ~5seconds, but is the disconnected. |
| 12:52.55 | E-bola | Anybody ever seen this behavior? |
| 12:53.29 | *** join/#asterisk ghost75 (~trechber@dslb-178-010-046-241.pools.arcor-ip.net) |
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| 13:05.58 | *** mode/#asterisk [+o malcolmd] by ChanServ |
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| 13:30.01 | igcewieling | E-bola: no, I've never heard of that problem. There is nothing in the Asterisk logs or CLI? |
| 13:34.05 | Katty | morning |
| 13:34.14 | Mango45 | Hi. |
| 13:34.22 | Katty | what's the scoop. |
| 13:34.31 | Mango45 | Does anyone know if ISP Telecom supports T.38? |
| 13:34.37 | Katty | and is it full of ice cream? |
| 13:34.52 | Katty | Mango45: why not call them and ask! |
| 13:35.10 | Mango45 | I am too busy eating the ice cream. |
| 13:35.21 | [TK]D-Fender | Katty, RAISINS |
| 13:35.27 | [TK]D-Fender | Katty, x2 |
| 13:35.36 | Katty | mmm raisins. |
| 13:35.42 | Katty | mmm kvas |
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| 13:49.23 | igcewieling | does anyone recall the correct location of license files like the Digium Commercial Fax license? |
| 13:50.04 | E-bola | igcewieling: There's a warning/error about sip retransmit, which is why I'm thinking its NAT related. However it would have to be pickup specific, since everything else works..... |
| 13:50.24 | E-bola | But I can not seem to find any issues relating to *8 pickup and nat |
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| 13:53.13 | igcewieling | E-bola: set directmedia=no |
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| 13:57.15 | Mango45 | Out of curiosity, does anyone here (besides myself) use OBi110 devices as FXO ports for an Asterisk server? |
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| 14:23.33 | E-bola | igcewieling: It's already set |
| 14:25.08 | igcewieling | E-bola: what version of Asterisk? |
| 14:25.57 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
| 14:28.23 | E-bola | 1.8.21 |
| 14:28.48 | E-bola | However it may not have been restarted (only reloaded) after adding directmedia=no |
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| 14:29.02 | E-bola | restarting it always messes up hints/blf's on the snom phones |
| 14:29.22 | igcewieling | you don't have to restart to apply sip.conf settings |
| 14:33.28 | igcewieling | *sigh* Our office network admin is more like an office disaster 8-| |
| 14:35.21 | igcewieling | last week he installed something which had a virus on the Big Boss's computer, today e-mail and DHCP are down and phones can't talk to the pbx. yay monday! |
| 14:36.08 | [TK]D-Fender | kill the port :) |
| 14:37.06 | igcewieling | [TK]D-Fender: Not My Problem. I do not in any way manage our office network. |
| 14:37.24 | *** join/#asterisk kresp0 (~screspo@186.248.78.188.dynamic.jazztel.es) |
| 14:37.38 | [TK]D-Fender | igcewieling, Pass the buck... |
| 14:37.54 | igcewieling | [TK]D-Fender: I don't even have access to many of our office switches. |
| 14:38.08 | igcewieling | and I don't have access to the PoS windows box which does DHCP on our network. |
| 14:38.39 | igcewieling | thank dog we run linux on all the boxes customers need. |
| 14:39.46 | [TK]D-Fender | "Hand out lease to infected machine? I think NOT!" |
| 14:39.53 | [TK]D-Fender | NO SOUP FOR YOU! |
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| 14:41.07 | igcewieling | the virus thing was early last week so it may not be related. |
| 14:42.26 | [TK]D-Fender | igcewieling, I'm learning about the toosl my new router has on it to help monitor this kind of stuff... |
| 14:42.35 | [TK]D-Fender | Of course.. I can't get the BASICS working jsut yet... |
| 14:43.34 | igcewieling | [TK]D-Fender: our office network is a total cluster fsck. Multiple gateways is just one example. Thankfully it is being redesigned this week or next week when we consolidate our multiple internet connections into one internet connection |
| 14:43.54 | igcewieling | We had one lower speed line with QoS for VoIP and one higher speed one without QoS |
| 14:44.43 | kresp0 | Hi all, |
| 14:44.43 | kresp0 | I'm having trouble recording a call using Record(). On the CLI everything seems ok, but no wav file is created after the call. CLI log and relevant extensions.conf config: http://pastebin.com/qG3afXFM |
| 14:45.44 | kaldemar | kresp0: "core show application Record" <-- option "k" |
| 14:46.24 | kresp0 | thanks kaldemar! |
| 14:46.41 | kresp0 | I'll try with k option |
| 14:47.22 | *** join/#asterisk jkroon (~jkroon@dsl-244-30-252.telkomadsl.co.za) |
| 14:47.55 | igcewieling | When using an Asterisk application it is often handy to check the documentation for the application. |
| 14:47.59 | jkroon | hi guys, chan_sip maps 604 (Does not exist anywhere) to AST_CAUSE_UNALLOCATED (1), would SUBSCRIBER_ABSENT (20) not be a better mapping? |
| 14:48.46 | jkroon | specifically, 604 says the responding service has authoritative information, but has no way of contacting the client, so a mapping of UNALLOCATED doesn't seem appropriate. |
| 14:48.48 | igcewieling | jkroon: if it doesn't exist, then it doesn't exist. If it was lagged or unreachable, then absent might be the correct response. |
| 14:49.13 | igcewieling | jkroon: does "sip show 604" show the peer? |
| 14:49.41 | [TK]D-Fender | igcewieling, SIP response.. not peer name |
| 14:49.42 | jkroon | no, i'm passing a call to an upstream provider, and if they can't reach the registered client they return a 604 code. |
| 14:49.44 | igcewieling | jkroon: also which version of Asterisk? There was a version where the cause codes were screwed up. |
| 14:49.47 | jkroon | 11.3.0 |
| 14:49.55 | igcewieling | that is not the version. 8-| |
| 14:50.02 | igcewieling | (not the version with the problem) |
| 14:50.33 | jkroon | igcewieling, yes, having made the bug report (one of probably a few) and having cooked a patch I'm overly familiar with that particular version somewhere in 1.8 (<13 IIRC) |
| 14:50.51 | file | it wouldn't surprise me if that mapping came from the RFC which defines it |
| 14:51.02 | kaldemar | jkroon: RFC 3389 specifies 604 to be mapped to unallocated number. |
| 14:51.17 | jkroon | hmm, ok, let me jump on the provider. |
| 14:51.56 | kaldemar | jkroon: sorry, 3398, not 3389. |
| 14:52.02 | jkroon | crap, but they're using the code correctly according to the specification (if I read wikipedia right) |
| 14:52.11 | file | could always remap it yourself in dialplan |
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| 14:52.16 | jkroon | how? |
| 14:52.26 | igcewieling | file: Do you think there any chance of HANGUPCAUSE and DIALSTATUS ever being writable from the dialolan? |
| 14:52.30 | file | https://wiki.asterisk.org/wiki/display/AST/Hangup+Cause |
| 14:52.44 | file | and Hangup allows you to specify a cause |
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| 14:56.32 | jkroon | file, yea, looks very handy. |
| 14:56.51 | jkroon | now, how do I get the actual SIP response code in order to remap it otherwise? |
| 14:57.20 | file | the hangup cause stuff will give you the technology specific cause |
| 14:57.52 | *** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone) |
| 14:58.57 | kresp0 | igcewieling, now it works, thanks ;) |
| 14:59.07 | jkroon | nice! that'll pick up if I need to rewrite, but won't give me the ability to change the HANGUPCAUSE will it? |
| 14:59.23 | file | Hangup allows you to specify the cause code |
| 14:59.46 | jkroon | or do I need to Dial(Local/20@hangmeup) with [hangmeup] containing something like _X!,1,Hangup(${EXTEN}) ? |
| 15:00.08 | jkroon | or can I just issue the Hangup() directly from the handler? |
| 15:00.28 | jkroon | hmm, now you've given me some cool stuff to test again ... argh! |
| 15:00.31 | file | I haven't done this, I just know the tools exist to accomplish it |
| 15:00.37 | file | (as is often the case) |
| 15:00.51 | *** join/#asterisk vfabi (~fabi@host-static-89-41-121-42.moldtelecom.md) |
| 15:01.25 | jkroon | ok, i'll mess with that |
| 15:01.55 | jkroon | is a 604 response appropriate if you "own" the dialed number, but that phone isn't reachable at the time? or is another code (eg, 480) perhaps more appropriate? |
| 15:06.08 | igcewieling | jkroon: check the RFC |
| 15:06.43 | *** join/#asterisk diskace (~kvirc@74.116.184.104) |
| 15:06.46 | igcewieling | then you'll have to check the SIP/ISUP RFC to see what the HANGUPCAUSE -> SIP reply should be |
| 15:07.29 | jkroon | igcewieling, that mapping is dodgy ... |
| 15:08.15 | jkroon | the RFC doesn't define what 604 is to be used for, merely the mapping. |
| 15:08.39 | igcewieling | which RFC number? |
| 15:08.47 | jkroon | 3398 |
| 15:10.12 | igcewieling | jkroon: correct. Now check http://www.ietf.org/rfc/rfc2543.txt to see what the codes are for |
| 15:12.48 | jkroon | igcewieling, that actually is much more in line with what I need :D |
| 15:12.58 | jkroon | 604 is NOT appropriate in the case they're sending it to me. |
| 15:13.03 | jkroon | goes off to pick a fight |
| 15:16.25 | igcewieling | "(11:06:08 AM) igcewieling: jkroon: check the RFC" <-- I meant the SIP RFC. |
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| 15:17.07 | igcewieling | jkroon: there are two things you can assume when dealing with telcos 1) they will randomly fail to deliver your call and 2) they will randomly send back the wrong response. |
| 15:18.31 | jkroon | igcewieling, oh yea, but I do have some clout with this particular provider. And since I picked a fight with them regarding me phoning numbers on their network from my cell getting messages like "the number you have dialed does not exist" they might actually take this one serious. It is something they can fix. |
| 15:18.49 | jkroon | and it's not random in this case, whenever the subscriber is absent, I get 604 instead of 480. |
| 15:19.09 | jkroon | which would explain the "phoning from my cell" scenario too (which had them baffled for some time). |
| 15:21.40 | *** join/#asterisk boch (~boch@186.182.116.157) |
| 15:22.00 | boch | hi all |
| 15:24.02 | diskace | hi guys, i am trying to find a solution preferably 'off the shelf' to display the length of a call queue to agents in my call center. I would like to display the queue on a LED screen in the call center room so agents can see for how many minutes clients are on hold. |
| 15:25.16 | diskace | we use switchboard as the backend. there is function in switchboard to display that in your browser but where really looking at parsing the info on an external screen. |
| 15:25.31 | boch | quick question, has asterisk queues a native/built-in way to callback customer when agent is ready to attend? i mean, without loosing its position |
| 15:25.43 | diskace | any idea ? :) Thx a bunch ! |
| 15:27.23 | [TK]D-Fender | diskace, "option in switchboard"? What "switchboard"? |
| 15:27.48 | [TK]D-Fender | boch, No, you'll have to code this yourself |
| 15:28.01 | diskace | [TK]D-Fender: digium switchboard or switchvox product |
| 15:28.19 | [TK]D-Fender | diskace, then you'll have to contact Digium directly. This is not a support channel for Switchvox |
| 15:29.00 | diskace | [TK]D-Fender: i did and they refer back to asterisk unfortunately i was looking at some feedback from the community. |
| 15:29.31 | [TK]D-Fender | diskace, Then there is nothing off the shelf and you have the usualy 2 tools to get this information yourself : AMI and Asterisk CLI |
| 15:29.54 | boch | [TK]D-Fender, alright, thank you |
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| 15:51.47 | newtonr | diskace: no one here works with Switchvox.. You are talking about working with the API Switchvox provides. I'd recommend asking on the Switchvox forums http://forums.asterisk.org/viewforum.php?f=29&sid=a95961998da13801382403ff7cfc1031 or contacting an integrator that regularly develops Switchvox applications |
| 15:53.16 | diskace | newtonr: thanks, i am currently reading about the switchvox api and it seems you are right. it's really a different platform and digium pointed me in the wrong direction lol |
| 15:54.29 | leifmadsen | ya, irc is for vanilla asterisk |
| 15:54.38 | leifmadsen | switchvox is an entirely separate product |
| 15:54.57 | newtonr | diskace: Sorry whoever you spoke with pointed you in the wrong direction... they must have been new.. |
| 15:55.03 | *** join/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk) |
| 15:55.15 | igcewieling | Release the hounds! |
| 15:55.17 | diskace | newtonr: the live chat... level1 agent :P |
| 15:56.49 | [TK]D-Fender | diskace, Switchvox is using * underneath though |
| 15:57.00 | [TK]D-Fender | diskace, and the 2 tools I mentioned ARE the place to get that info. |
| 15:57.03 | *** join/#asterisk jblack (~jblack@173-160-189-58-Washington.hfc.comcastbusiness.net) |
| 15:57.06 | autofsckk | hello i have this notices handle_request_invite: Sending fake auth rejection for device 100 for what i read is that my box is being attacked? but what can i do to avoid it? |
| 15:57.11 | diskace | [TK]D-Fender: i know but the cli is locked down |
| 15:57.20 | [TK]D-Fender | diskace, Which leaves AMI <- |
| 15:57.33 | diskace | [TK]D-Fender: well there is an xml extend api |
| 15:57.41 | diskace | [TK]D-Fender: http://developers.digium.com/switchvox/wiki/index.php/WebService_overview |
| 15:58.21 | [TK]D-Fender | diskace, that is something completely foreign... |
| 15:59.23 | [TK]D-Fender | diskace, Basic queue information is in AMI. This also seems to cover it in some capacity : http://developers.digium.com/switchvox/wiki/index.php/Switchvox.extensions.callQueues.getInfo |
| 15:59.49 | diskace | [TK]D-Fender: exactly |
| 16:00.41 | diskace | [TK]D-Fender: im just really newb when it comes to xml so ill read about it |
| 16:05.12 | *** join/#asterisk djjunde (~djjunde@dsl-espbrasgw1-54f9cf-178.dhcp.inet.fi) |
| 16:05.25 | *** join/#asterisk slicknick5181 (~slicknick@204.195.131.94) |
| 16:05.43 | newtonr | Yeah AGI/AMI are not exposed. The site you are on is the way to find what you want. |
| 16:06.04 | slicknick5181 | Does anybody here have a working Polycom soundpoint 301 or Mitel 5220 with *1.8 |
| 16:06.16 | [TK]D-Fender | diskace, Actually.. THIS : http://developers.digium.com/switchvox/wiki/index.php/Switchvox.callQueues.getCurrentStatus |
| 16:06.34 | [TK]D-Fender | diskace, That earlier one is got "setup" info, not "status" |
| 16:06.56 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 16:06.58 | [TK]D-Fender | slicknick5181, Tons of us |
| 16:07.22 | [TK]D-Fender | slicknick5181, The phone was never picky about any particular version of *..... |
| 16:08.55 | slicknick5181 | [TK]D-Fender, I can't get transmission sound from either the polycoms or the mitel but I do have a sip gateway with working sound both ways |
| 16:09.22 | diskace | [TK]D-Fender: yeah even better |
| 16:09.46 | slicknick5181 | [TK]D-Fender, But on the Mitel 5220 the speaker phone works both ways but the handset doesn't |
| 16:09.59 | [TK]D-Fender | slicknick5181, then you have a networking setup error |
| 16:10.04 | igcewieling | slicknick5181: we have several hundred Polycoms talking to Asteisk 1.8 |
| 16:10.16 | [TK]D-Fender | slicknick5181, Typically NAT / firewall |
| 16:10.32 | *** join/#asterisk dtcrshr (~datacrush@unaffiliated/datacrusher) |
| 16:12.58 | slicknick5181 | [TK]D-Fender, My gateway works though so I figured I had set up NAT correctly |
| 16:18.23 | [TK]D-Fender | slicknick5181, So far that's Figured=1, debug=0 |
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| 16:20.02 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 16:35.13 | *** join/#asterisk JustinAiken (~Adium@74.118.180.13) |
| 16:36.03 | JustinAiken | For confbridge, you can use the M option and opitonally supply the music_on_hold class; would that syntax look like this: ConfBridge("101", M(some_music_class)) ? |
| 16:38.04 | [TK]D-Fender | JustinAiken, Yes, adn the conf wirthout quotes though |
| 16:38.29 | JustinAiken | ConfBridge(101, M(some_music_class)) ? |
| 16:38.43 | [TK]D-Fender | And no extra spaces... |
| 16:39.48 | JustinAiken | sweet, thanks :) |
| 16:39.49 | igcewieling | never ever add extra spaces |
| 16:42.43 | *** join/#asterisk danfromuk (~IceChat77@unaffiliated/danfromuk) |
| 16:44.18 | *** join/#asterisk deo (~deo@112.198.90.145) |
| 16:45.24 | danfromuk | Hi, The mysql realtime definition Ive got for queues.conf doesnt seem to have a field for 'announce-position'. Does anyone know what the field is supposed to be called? |
| 16:46.02 | igcewieling | danfromuk: where did you get the realtime definition from? |
| 16:46.31 | danfromuk | voip-info. Cant seem to find it on the wiki |
| 16:46.56 | cusco | danfromuk: called just that: announce-position |
| 16:46.58 | igcewieling | try getting it from /path/to/src/asterisk/contrib/realtime/mysql |
| 16:47.28 | *** join/#asterisk OverOnTheRock (~irc@162.r1.ray.transact.bm) |
| 16:47.33 | igcewieling | voip-info should be the last place you look for Asterisk information, not the first place |
| 16:47.59 | *** part/#asterisk JustinAiken (~Adium@74.118.180.13) |
| 16:48.47 | danfromuk | igcewieling: I know. Bad habit ive got in to. |
| 16:49.02 | igcewieling | danfromuk: bad habits like that waste a lot of your time. |
| 16:49.10 | danfromuk | But queues.conf doesnt have a sample sql file in that folder. |
| 16:50.53 | igcewieling | report it as a bug 8-) |
| 16:55.17 | navaismo | I have mine as: announce_position varchar(5) |
| 16:55.27 | kaldemar | danfromuk: take a peek in the other folders |
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| 17:11.57 | danfromuk | navaismo: thanks. that was it. |
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| 17:18.25 | antsygeek | is there a way i can initiate a SIP call to a non-numerical address, like me@sip.mydomain.com? |
| 17:18.50 | antsygeek | sip:100@sip.mydomain.com works |
| 17:20.02 | igcewieling | yes. |
| 17:20.45 | igcewieling | in sip there is no real difference between number and non-number before the @ |
| 17:21.22 | *** join/#asterisk ghost75 (~trechber@dslb-178-010-046-241.pools.arcor-ip.net) |
| 17:21.40 | antsygeek | igcewiigok so i defined a [me] in sip.conf, but asterisk seems not to find it |
| 17:22.01 | [TK]D-Fender | antsygeek, because you match in the DIALPLAN |
| 17:22.29 | [TK]D-Fender | antsygeek, extensions.conf is what processes calls |
| 17:22.59 | [TK]D-Fender | antsygeek, So if you dial "me" .... you need a match in your dialplan for it just like anything else. |
| 17:23.19 | antsygeek | oh |
| 17:23.33 | antsygeek | i can do exten => me,1,Whatever()? |
| 17:25.02 | [TK]D-Fender | antsygeek, Yes |
| 17:25.15 | *** join/#asterisk vlad_sta_ (~vlad_star@91.206.59.137) |
| 17:25.24 | antsygeek | nice |
| 17:25.25 | antsygeek | thanks |
| 17:27.02 | [TK]D-Fender | antsygeek, You're welcome |
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| 17:27.51 | antsygeek | well, i just tried it but does not work. may i show you my config? |
| 17:28.27 | [TK]D-Fender | antsygeek, pastebin the call attempt at verbose 10, SIP DEBUG enabled as well |
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| 17:30.00 | antsygeek | http://pastie.org/7698437 |
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| 17:30.36 | [TK]D-Fender | waits on the call.... |
| 17:31.10 | antsygeek | and this was in the console: http://pastie.org/7698442 |
| 17:31.42 | cusco | the error seems self-explanatory |
| 17:31.53 | [TK]D-Fender | Indeed |
| 17:31.57 | cusco | (in the console) |
| 17:32.01 | [TK]D-Fender | and that wasn't even with SIP DEBUG enabled as requested |
| 17:32.03 | antsygeek | so i need to add it to the default context |
| 17:32.07 | antsygeek | ok, sorry |
| 17:32.07 | [TK]D-Fender | "sip set debug on" <- |
| 17:32.17 | [TK]D-Fender | antsygeek, Not yet... |
| 17:32.35 | [TK]D-Fender | antsygeek, We don't see WHY it's looking in [default] |
| 17:32.52 | [TK]D-Fender | antsygeek, Got more info on that while we await new call debug? |
| 17:32.53 | jeffspeff | ok, so one of our offices is using some version of Cisco unified communication server, i believe they're using SCCP from phones to the Cisco PBX. Apparently, the Cisco UC didn't have voicemail features, so they setup a seperate asterisk box for VM. Now, I've been tasked with getting the phones (which register to the Cisco UC) to receive MWI from asterisk. Ideas? |
| 17:33.39 | [TK]D-Fender | jeffspeff, Setup fixed IP's for the phones and add them as peers. |
| 17:33.56 | cusco | what did 120 call? |
| 17:33.58 | [TK]D-Fender | jeffspeff, And pray they don't reject non-reg-server comms |
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| 17:34.11 | [TK]D-Fender | cusco, "me" obviously! |
| 17:34.40 | _Corey_ | jeffspeff: Cisco has some *xx type of integration for external voicemail systems to turn on/off lamps. You could do something with the externnotify app to make that work. |
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| 17:35.18 | jeffspeff | [TK]D-Fender, ok, so the phones would be static peers and send MWI even though the phone isn't registered? |
| 17:35.23 | _Corey_ | jeffspeff: (i.e. you dial some code+exten to turn on the lamp, then someothercode+exten to turn off) |
| 17:35.40 | cusco | antsygeek: what is the context of peer 120 ? |
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| 17:35.46 | [TK]D-Fender | jeffspeff, Possible as far as * goes |
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| 17:36.00 | antsygeek | oh. ok it works now. i did a reload and it works now |
| 17:36.15 | cusco | ok |
| 17:36.18 | antsygeek | i did the reload after i saved the new configuration, but apparently it didnt work |
| 17:36.20 | jeffspeff | _Corey_, so you're suggesting to have * dial a context on the CUCM that would turn the lamp on or off? |
| 17:36.21 | antsygeek | strang |
| 17:36.27 | antsygeek | thanks guys!! |
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| 17:43.25 | _Corey_ | jeffspeff: I've got a customer who did a tie-in with an Avaya VM in that way using CUCM |
| 17:44.08 | jeffspeff | the problem is i'm not the least bit familiar with CUCM or its dialplaning/features |
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| 17:47.26 | _Corey_ | jeffspeff: Looks like there's a "Message Waiting Inidcator" extension that can be enabled on the Cisco side. I'd look for some documentation on that or call a Cisco engineer. |
| 17:48.55 | jeffspeff | _Corey_, thanks for the help. this is just some crap that tossed on me by my boss. they should have just installed asterisk in the beginning (like i told them to) instead of using CUCM (which nobody knew how to use). for some reason by boss really likes spending crap tons of money to impress people. |
| 17:49.06 | jeffspeff | thanks though. i'll look into that and TK's suggestions |
| 17:49.34 | _Corey_ | Perfect occasion to migrate to Asterisk... :) |
| 17:50.46 | jeffspeff | they consulted me on which PBX to go with prior to setting up that facility. I've got asterisk running throughout my entire division, they've poured too much money into that solution now to turn back |
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| 18:02.22 | igcewieling | The nice thing about Asterisk is that, unlike CCM, you don't need scientific notation to quote the price. |
| 18:02.39 | tzanger | haha |
| 18:05.27 | [TK]D-Fender | Hence the true origin of Cisco's claims to be the most "powerful" networking company... |
| 18:06.44 | drmessano | Cisco is the most powerful networking company.. Their shit uses more power than anything else in the rack |
| 18:09.16 | cusco | lol |
| 18:09.18 | cusco | thus powerfull |
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| 18:09.44 | [TK]D-Fender | and costs many powers.... |
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| 18:15.07 | drmessano | I remember how incredibly painful it was to get time conditions set up in CME |
| 18:15.13 | drmessano | Two calls to Cisco |
| 18:15.18 | drmessano | Why oh why |
| 18:15.22 | drmessano | Nothing documented |
| 18:15.28 | SeRi | yawns. |
| 18:15.36 | SeRi | waz up all |
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| 19:27.49 | Kobaz | so |
| 19:28.01 | Kobaz | quick random anecdotal survey |
| 19:28.04 | jeffspeff | _Corey_, hey, just in case you ever need this... http://www.hurdman.net/mirror/voip-info/wiki/view/Asterisk+Cisco+CallManager+Voicemail+Integration.html |
| 19:28.10 | Kobaz | what's everyone's failure rates for polycom soundpoint phones? |
| 19:28.22 | [TK]D-Fender | Kobaz, I've had one. Ever. |
| 19:28.36 | Kobaz | The one customer, with about 150 phones, one goes bad in some fashion about once a month |
| 19:28.51 | Kobaz | usually it's the headset jack |
| 19:29.01 | [TK]D-Fender | Kobaz, I have had phones where the CSR literally rubbed the frosted icons right off the speakerphone key :) |
| 19:29.04 | Kobaz | but occasionally things like random buttons stop working, or all of the buttons stop working |
| 19:29.10 | jeffspeff | Kobaz, i've got about 200 of them in production right now, and i love them, haven't had any go bad in the year we've had them and we baught them used |
| 19:29.11 | Kobaz | yeah, i've seen that |
| 19:29.33 | Kobaz | i have about 1000 in production and they have been all okay except this one customer |
| 19:29.42 | Kobaz | i think it's the carpet dust or something, i dont know |
| 19:29.47 | jeffspeff | lmao |
| 19:30.01 | Kobaz | they are a carpet warehouse with call centers |
| 19:30.25 | Kobaz | but i've had like 10-15 phones have their headset jack go bad |
| 19:30.33 | Kobaz | 331's |
| 19:31.16 | Kobaz | the latest casualty is that this 550's buttons all stopped working |
| 19:31.38 | Kobaz | the phone can receive calls, it's registered in *, and you can pick up the handset and get the locally generated dialtone |
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| 19:37.04 | malcolmd | http://www.viziflex.com/telephone/ environmental covers for phones. they make 'em for digium phones, too. |
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| 20:07.22 | igcewieling | Kobaz: 331s? Those must be some pretty old phones |
| 20:10.20 | drmessano | 331's are old? |
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| 20:16.00 | igcewieling | drmessano: I'd have to check, but 4 yrs or so I think. |
| 20:18.49 | drmessano | I thought when I bought a bunch of them in 2011 that they were a relatively new replacement for the 330 |
| 20:19.01 | igcewieling | interesting, it doesn't look like they are actually EOLd |
| 20:19.11 | igcewieling | drmessano: I thought Polycom EOLd all their non-HD phones |
| 20:19.26 | igcewieling | Apparently not. |
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| 20:41.45 | przerull | so i'm using the originate dialplan application to dial out to a local channel, how can I specify the ORIGINATE_STATUS that get's returned by my local extension? |
| 20:42.05 | przerull | ps i'm on asterisk 1.8.10 |
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| 20:54.16 | FLeiXiuS | How can I play multiple digits with Playback? |
| 20:54.36 | FLeiXiuS | ei, Playback(thirty,thousand) |
| 20:57.46 | igcewieling | FLeiXiuS: See "core show application saydigits" and "core show application playback" core show application and core show function are your friends, hang out with them, buy them a beer, get to know them. |
| 20:58.25 | igcewieling | There is also a SayNumber application as well. |
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| 21:04.56 | przerull | any ideas on hte ORIGINATE_STATUS issue? |
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| 23:59.40 | navaismo | igcewieling, which is the problematic ethernet controller? |