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| 04:02.53 | snadge | exten => _+611[38].,1,Set(CALLERID(dnid)=${EXTEN:3}) |
| 04:02.54 | snadge | exten => _+611[38].,n,goto(${EXTEN:3},1) |
| 04:02.59 | snadge | does that do what i think it should do? |
| 04:03.26 | snadge | ie.. if the number starts with +6113 or +6118 then.. strip the first 3 digits.. eg.. dial 13. or 18. |
| 04:03.55 | snadge | i guess i could test it.. live on a production system and see ;) |
| 04:28.05 | snadge | that worked.. thanks guys.. i love you so much ;) |
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| 07:03.56 | Lope | [TK]D-Fender: you recommended I not use extensions.ael. So I'll switch to extensions.conf now |
| 07:05.21 | Lope | Can someone please show an example of how to greet a caller like "Hi, you've reached blah" Dial(,30) (after 30s timeout) "Please leave a message?" |
| 07:09.33 | robmal | Playback, Dial, Playback |
| 07:09.48 | robmal | I can draw a diagram if you want. |
| 07:09.52 | Lope | Can you show a real world example please? |
| 07:11.37 | [TK]D-Fender | exten => 12345,1,Playback(hello) |
| 07:11.51 | [TK]D-Fender | exten => 12345,n,Dial(SIP/myphone,30) |
| 07:11.52 | robmal | https://drive.google.com/file/d/0BxBTJ6nGbNdSZ1lRUlh4d1hjS3M/view?usp=sharing |
| 07:12.10 | [TK]D-Fender | exten => 12345,n,Playback(guessimnothere) |
| 07:12.32 | robmal | The last playback should be tt-monkeys :-( |
| 07:12.43 | [TK]D-Fender | Monkeys don't ask about messages |
| 07:13.10 | robmal | Some people could assume you answered. |
| 07:14.07 | Lope | What does exten mean in the above example? lets say my DID number is 7777777 would it be 7777777 => 12345,1,Playback(hello) ? |
| 07:14.14 | Lope | What is the 12345 part? |
| 07:14.26 | Lope | (I know the 1 and n are priorities) |
| 07:14.29 | robmal | Its the number of satan. |
| 07:15.04 | Lope | oh, cool, I've been meaning to have a chat with him. |
| 07:15.09 | [TK]D-Fender | no |
| 07:15.21 | [TK]D-Fender | exten => 7777777,1,Playback(hello) |
| 07:15.24 | [TK]D-Fender | Read the book |
| 07:15.41 | [TK]D-Fender | exten => EXTENSION,PRIORITY,APPLICATION(DATA) |
| 07:15.53 | Lope | Okay cool, thanks. |
| 07:15.57 | robmal | Why no caps? |
| 07:16.14 | Lope | !title |
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| 07:17.59 | Lope | where is the book? |
| 07:18.30 | [TK]D-Fender | ~book |
| 07:18.30 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 07:18.33 | [TK]D-Fender | ^ |
| 07:19.44 | Lope | Whats a simple way to setup a mailbox such that caller's messages are stored as files with their phone number, the extension they dialled and the date? |
| 07:21.23 | [TK]D-Fender | We like to call that VOICEMAIL |
| 07:21.27 | [TK]D-Fender | And it's already a dialplan app.... |
| 07:21.31 | Lope | I'm looking inside voicemail.conf. I see all this stuff about IMAP, I'd prefer to just store them as WAV and then execute my script afterwards to compress it. |
| 07:21.35 | [TK]D-Fender | "core show application voicemail" |
| 07:21.43 | [TK]D-Fender | And go read the sample config for it |
| 07:21.55 | Lope | I saw the Voicemail() thing being called in the sample extensions.conf file |
| 07:21.59 | [TK]D-Fender | storing as files is the DEFAULT |
| 07:22.02 | [TK]D-Fender | IMAP, etc are OPTIONS |
| 07:22.19 | Lope | but the params were so simple I figured there must be more config |
| 07:22.31 | [TK]D-Fender | no, it is that simple |
| 07:23.07 | [TK]D-Fender | pile of lines for some general stuff, then1 line per mailbox to define the box withing a VM context |
| 07:23.24 | [TK]D-Fender | Then you have to actuall USE the dialplan apps |
| 07:23.43 | Lope | Where can I see a list of variable definitions like ${user} ${mbx} etc? |
| 07:24.11 | [TK]D-Fender | Depends if they are standard or not |
| 07:24.26 | [TK]D-Fender | Otherwise your own variables... have no definition except that which you set them to |
| 07:24.34 | [TK]D-Fender | "Asterisk Standard Variables" |
| 07:24.38 | [TK]D-Fender | VEWRY Googlable |
| 07:24.57 | Lope | okay, thanks. sorry will google more. |
| 07:25.08 | [TK]D-Fender | And go play around |
| 07:25.52 | [TK]D-Fender | off to bed, later.... |
| 07:26.40 | Lope | thanks, later |
| 07:28.36 | Lope | by default my ubuntu asterisk install has no sounds in /var/lib/asterisk I see there's a apt package called `asterisk-core-sounds-en` and also `asterisk-core-sounds-en-(gsm|wav|g722)` |
| 07:28.46 | Lope | which package do you recommend? |
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| 07:33.23 | Lope | weird. I already have the asterisk-core-sounds-en asterisk-core-sounds-en-gsm packages installed without having done anything. |
| 07:33.53 | robmal | Playback(tt-monkeys) ! |
| 07:39.49 | Lope | so pbx_ael is for the extensions.ael file. What module is used to read extensions.conf? |
| 07:43.43 | Lope | a friend help me setup asterisk to use extensions.ael. But I want to use extensions.conf now. How can I switch? |
| 07:44.02 | Lope | with debug/verbose 10 and core reload, I see it's not loading extensions.conf |
| 07:44.54 | Lope | extconfig seems to be relating to useing a DB instead of flat files. |
| 07:45.46 | robmal | ~book |
| 07:45.47 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 07:47.52 | ChannelZ | pbx_config |
| 07:48.09 | Lope | thanks mate |
| 07:52.50 | Lope | WARNING[8485][C-00000003]: pbx.c:4878 pbx_extension_helper: No application 'Playback' |
| 07:53.05 | Lope | I've loaded these modules: app_controlplayback.so app_playback.so app_playtones.so |
| 07:53.16 | Lope | they are all that exists for *play* |
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| 08:07.24 | Lope | haha lol I heard the monkeys when I loaded all modules. now the question is which module is it... |
| 08:08.40 | robmal | It should be just app_playback, without it some things can go... bananas. |
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| 08:10.49 | Lope | i've got app_playback loaded. |
| 08:11.15 | Lope | oh no, I don't. |
| 08:13.22 | Lope | haha. I just listed the new modules... but did not have load= infront :D |
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| 08:38.14 | Lope | lol it's quite funny chaining an Alison (your call will be answered by the next available representative' with tt-monkeys WAAAH WAHHH waaaa! |
| 08:43.09 | robmal | That's what tt-monkeys are for. |
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| 08:49.56 | snadge | oh crap.. what does WARNING[28801]: acl.c:328 ast_append_ha: Invalid IP address in |
| 08:49.58 | snadge | lots of those mean |
| 08:54.12 | Lope | i've got /usr/share/asterisk/sounds/en_US_f_Allison/tt-monkeys.gsm and other files that can be played using Playback(tt-monkeys). now I've put my own WAV inside /usr/share/asterisk/sounds/custom/foo.wav and trying Playback(foo). But it says "File foo does not exist in any format" |
| 08:56.17 | Lope | The permissions are the same as the Alison files. |
| 08:57.13 | robmal | tt-monkeys.gsm |
| 08:57.17 | robmal | foo.wav |
| 08:57.25 | robmal | Guess what? |
| 09:00.01 | Lope | okay now I specified the absolute path (google said I can) and now it says File /usr/share/asterisk/sounds/custom/foo.wav does not exist in any format.... but `ls` says otherwise? |
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| 09:03.04 | Lope | show file formats> no such command |
| 09:06.38 | snadge | okay.. so we had a network outage.. and now it would appear that one of our core servers has a corrupt realtime db |
| 09:07.00 | snadge | i dont even.. how does..sig |
| 09:07.49 | snadge | how is that even possible |
| 09:08.06 | snadge | how to i refresh it? |
| 09:09.37 | Lope | none of these commands work in my new ubuntu asterisk install |
| 09:09.54 | Lope | show file formats, show translation ? |
| 09:12.27 | robmal | Meh. |
| 09:12.35 | robmal | Lope: voip-info is outdated. |
| 09:12.46 | robmal | core show file whatever |
| 09:13.06 | robmal | snadge: And the db is? MySQL? |
| 09:13.29 | Lope | I've got plenty of wav formats showing |
| 09:13.37 | coppice | voip-historical-info |
| 09:13.39 | Lope | Any idea why it doesn't find my file that exists? |
| 09:13.55 | robmal | Because it is in the wrong format. |
| 09:13.59 | snadge | yes.. the db is mysql |
| 09:14.11 | Lope | I'm going to check that it's in the correct format, but I would have thought it would be a different error in that case. |
| 09:14.27 | snadge | i can access it from mysql workbench.. it seems to look no different to the other databases, from casual inspection |
| 09:14.40 | snadge | all the peers are showing unreachable.. i cant register to it |
| 09:15.26 | robmal | snadge: Copy all mysql files to a safe place, then mysqlcheck or whatever. |
| 09:18.21 | ChannelZ | chances are your permissions are completely screwed up |
| 09:18.42 | ChannelZ | (Lope.. for the sounds) |
| 09:22.12 | Lope | Strange. I'm using audacity. I resampled my 22050hz Wav as 8000hz. I also changed the format from 32bit to 16 bit. I exported it. But the exported file is still 22050 32 bit :/\ |
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| 09:32.26 | ChannelZ | Audacity is free but retarded. You need to change the project rate |
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| 09:33.56 | ChannelZ | The 32-bit is a lie if you're checking by reloading it in to Audacity and seeing it say that in the track |
| 09:34.50 | Bordr | Hoping someone can help... Having a problem with chan_sip.c lagging out extensions. They are local to the system (i.e. - same subnet). No firewall, single POE switch to the server. |
| 09:36.06 | ChannelZ | By 'lagging out' do you mean you have Qualify turned on and it's saying they've become unreachable? |
| 09:36.48 | Bordr | Yes. I get the following lines: |
| 09:36.48 | Bordr | [2016-01-06 02:28:59] NOTICE[2692] chan_sip.c: Peer '117' is now Lagged. (2035ms / 2000ms) |
| 09:36.48 | Bordr | [2016-01-06 02:32:09] NOTICE[2692] chan_sip.c: Peer '111' is now Lagged. (2038ms / 2000ms) |
| 09:37.01 | robmal | Oh noes! |
| 09:37.16 | Bordr | The calls will drop, go silent and recover, or recover with 1 way audio. |
| 09:37.21 | ChannelZ | hmm.. well that's 2 seconds |
| 09:37.24 | robmal | Oh noes! |
| 09:37.29 | ChannelZ | It sounds like you have some network issues |
| 09:37.50 | robmal | Si. |
| 09:38.03 | ChannelZ | Packets whizzing off into space, being dropped, who knows |
| 09:38.08 | Bordr | That's what I thought as well. Checked everything I could think of and found nothing except in the /var/log/messages... Had a statement about over full arp cache |
| 09:38.21 | ChannelZ | Or the phones are particularly crappy and losing their minds |
| 09:38.28 | Bordr | increased the threshold to double and the error went away. |
| 09:38.28 | Lope | Okay I managed to convert the 32bit wav to 16 bit. I did this by changing the default sample format to 16 bits, then starting a new project and importing the WAV again. |
| 09:38.58 | ChannelZ | Does ifconfig show tons of errors or anything? |
| 09:39.03 | Bordr | iperf shows near 98% throughput between links |
| 09:40.38 | Bordr | ifconfig just shows the network interfaces.... look normal to me |
| 09:40.39 | Bordr | <PROTECTED> |
| 09:40.39 | Bordr | <PROTECTED> |
| 09:41.05 | robmal | Meh. |
| 09:41.13 | Lope | okay, now my wav file is 8000hz 16 bit Mono. And it still says file /usr/share/asterisk/sounds/custom/foo.wav does not exist in any format? |
| 09:41.29 | ChannelZ | increasing the threshold might make the problem 'go away' but you've really just hidden it rather than fixed it. If the devices are truly taking over 2 seconds to respond, that doesn't seem right to me, unless they are doing that on purpose just for the notify packets |
| 09:41.51 | ChannelZ | What kind of phones are they? |
| 09:42.31 | ChannelZ | Lope, RE: check permissions. Whatever user your askterisk runs as, if not root, might not have access to the entire path/file. |
| 09:43.08 | Bordr | sorry should have mentioned that Aastra 6867i and a SPA112 ATA. |
| 09:43.11 | Lope | I checked the permissions, the file is owned by root:root with rw r r (same as all my other .gsm files that Playback() can play (eg: tt-monkeys) |
| 09:43.30 | robmal | module reload whatever |
| 09:43.33 | ChannelZ | but that's presumably not in your 'custom' folder |
| 09:43.46 | ChannelZ | does 'custom' have +rx? |
| 09:44.08 | Lope | I am getting one strange error where it says Unable to open /path/to/custom/foo.wav (format (ulaw)): No such file or directory |
| 09:44.19 | Lope | ulaw, is that the same as WAV? |
| 09:44.24 | ChannelZ | no |
| 09:44.50 | ChannelZ | but it's misleading anyway. And you shouldn't include the extension in your Playback() |
| 09:44.53 | Lope | custom is actually a symlink |
| 09:45.14 | Lope | (created by the apt asterisk package) i'll move it somewhere normal |
| 09:45.45 | ChannelZ | regardless, is the real directory accessable? |
| 09:46.21 | Bordr | increasing the threshold just stopped the arp cache errors... i agree, not sure what could be causing them. went onsite because I thought for sure they made a switching loop. sounds like it the way the phones drop calls. |
| 09:46.48 | Lope | I've moved the file to the root dir now, and referenging it as Playback(/foo.wav) |
| 09:47.43 | Bordr | also, it seams at night the interval is much better... only getting them every few minutes. during the day it's about every 2-30 seconds - assuming due to traffic. on a side note, behind ext 117 i can ping the server constantly without any abnormal ping times... |
| 09:48.59 | ChannelZ | still wrong.. Playback(/foo) - you don't put the extension on, as I said |
| 09:49.50 | ChannelZ | Asterisk will look for your audio files with various extensions on its own, preferring to find a version which is in the codec of the channel it's being played on |
| 09:50.35 | Lope | https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Playback doesn't offer a place to choose a playback format |
| 09:50.36 | ChannelZ | IE you can have foo.wav, foo.ulaw, foo.gsm, and it will determine which one to best play. |
| 09:50.51 | Lope | I'm using Playback(/foo.wav) the .wav should suggest the format, right? |
| 09:50.59 | ChannelZ | You're not listening to me |
| 09:51.07 | ChannelZ | DON'T PUT .wav ON THE END |
| 09:51.20 | Lope | oh, okay. |
| 09:51.26 | ChannelZ | (of your Playback filename.. leave it on the actual file) |
| 09:52.28 | Lope | okay, so basically the console says it looks for a ulaw, but doesn't find it. |
| 09:52.44 | Lope | however it's quite happy to play tt-monkeys as tt-monkeys.gsm |
| 09:52.48 | ChannelZ | it's just worded misleadingly |
| 09:54.27 | ChannelZ | so long as you have format_wav loaded it should find it |
| 09:54.37 | Lope | oh wow. okay got somewhere |
| 09:54.39 | ChannelZ | Also it needs to be 8kHz, 16-bit, MONO |
| 09:54.53 | Lope | When I give the absolute path, excluding the .wav it says |
| 09:55.14 | Lope | playing /path/to/foo.slin language en |
| 09:55.27 | ChannelZ | as I said |
| 09:55.38 | Lope | My wav is 8000hz 16 bit mono |
| 09:55.58 | ChannelZ | then all is well |
| 09:56.14 | Lope | what is foo.slin (i don't have a file with that name) |
| 09:56.34 | ChannelZ | signed linear. It's what it gets converted to internally |
| 09:56.40 | Lope | oh wow, okay it does actually play. it's just very short :) |
| 09:56.49 | Lope | I see |
| 09:56.56 | Lope | i need to add a delay or something |
| 09:57.35 | ChannelZ | format_wav is just a format handler that knows how to read out of a wav container and then turns it into slin |
| 09:58.27 | Lope | ChannelZ: thanks so much :) |
| 09:59.43 | ChannelZ | sure |
| 10:00.05 | Lope | Can I ask the caller if they'd like to leave a voice message or continue to hold? |
| 10:01.04 | Lope | I've got various DID's and 1 softphone. I only want to allow 1 conversation to go through to the softphone simultaneously. |
| 10:01.38 | Bordr | Lope - You can in Queues. We create an IVR and use the IVR breakout menu. |
| 10:01.55 | Bordr | Lope - You can do that also in Queues by enabling skip busy agents. |
| 10:03.37 | Bordr | By skipping busy agents you can have them wait on hold and announce every say minute the option to leave a message. If they stay in the queue too long you can then drop them to VM or elevate to another Queue. |
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| 10:29.38 | Bordr | Lope, did that work for you? |
| 10:30.30 | Bordr | ChannelZ, can you recommend anywhere else to look? |
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| 12:26.20 | napnap | Hi all. Since many weeks/months I have trouble with my VoIP, I have cut during conversation (few seconds of blank). After the advice I got here I captured packets on my PBX. |
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| 12:28.25 | napnap | And I see many "PDUType: Resupply offer" from my PBX to my phone and "PDUType: Unknown" from my phone to PBX during the time when it cuts (I think) |
| 12:29.11 | napnap | I have no idea what this is and if it can have a link? |
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| 12:44.21 | napnap | Ok... according to https://ask.wireshark.org/questions/3609/dis-pdu-filtering is not DIS protocol but RTP sended on port 3000 |
| 12:44.27 | napnap | (I have aastra phones) |
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| 13:11.08 | weissbier | hi |
| 13:12.04 | *** join/#asterisk skrusty (~skrusty@168.63.14.171) |
| 13:13.39 | weissbier | if i use the "b" option within the "Dial" application, how do i correctly specify the arguments? at the moment it looks like this: Ttb(internal-gosub^context^1) |
| 13:18.40 | weissbier | disregard that, i should do some more googling ;) https://wiki.asterisk.org/wiki/display/AST/Pre-dial+handlers+Specification |
| 13:20.59 | *** join/#asterisk Oatmeal (~Suzeanne@75-103-145-152.ccrtc.com) |
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| 13:45.45 | AsteriskNoob | Hi Fellas , a quick question , to keep my NAT pinholes open using Qaulify=yes is not doing keeping my nat connections open , and at moment only few extension have this issue , but could easily spread to others so Should the Value for Qualify=XXX be less than deafukt of 2000ms or more than this, Please Help, Thanks |
| 13:47.45 | *** join/#asterisk [TK]D-Fender (~chatzilla@216-191-106-163.dedicated.allstream.net) |
| 13:52.22 | WIMPy | 1. You should use keepalive on your phones if you have trouble with expiring NAT, not qualify. |
| 13:53.25 | *** join/#asterisk Draecos (~Draecos@203-121-194-11.e-wire.net.au) |
| 13:53.27 | WIMPy | And if 2s works for qualify depends on your network situation, but if there is no answer within 2s, do you really want to send a VoIP call that way? |
| 13:55.48 | *** join/#asterisk jpastore (~jpastore@50-200-51-210-static.hfc.comcastbusiness.net) |
| 13:55.57 | AsteriskNoob | I have 20s Keepalive on phones but no effect, and one of the affected ones is support line agent |
| 13:56.05 | AsteriskNoob | so need to resolve this |
| 13:56.39 | WIMPy | Looks like you need to take care of that router then. |
| 13:56.39 | AsteriskNoob | any ideas anyone on Grandstream GXP 1200 phones |
| 13:59.53 | *** join/#asterisk happy-dude (uid62780@gateway/web/irccloud.com/x-glpcqraqyqmrpxca) |
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| 14:08.22 | AsteriskNoob | WIMPy my router xlate are set to timeout every one 1 hour |
| 14:29.10 | *** part/#asterisk eschmidbauer (~eschmidba@unaffiliated/eschmidbauer) |
| 14:31.01 | *** part/#asterisk weissbier (~weissbier@2001-4ba0-ffa4-23d-0-0-0-1002.ipv6dyn.netcologne.de) |
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| 14:50.21 | WIMPy | Maybe it has run out of RAM? |
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| 15:28.28 | *** join/#asterisk dsmith_9031 (~chatzilla@70-90-28-109-BusName-pa.hfc.comcastbusiness.net) |
| 15:29.54 | dsmith_9031 | Good Morning all, anyone home? |
| 15:30.18 | [TK]D-Fender | Some of us are at work... |
| 15:30.42 | dsmith_9031 | As am I, curiously I didn't think of that |
| 15:33.51 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 15:33.57 | dsmith_9031 | But nevertheless, here's a question I'm search for the answer to. I have an existing PBX system through our phone company, but I would like to add Asterisk as an addendum to this system to automatically answer inbound calls on the first ring. Like an answering machine, and present a simple auto attendant menu to direct calls to extensions. Is this something that can be accomplished using... |
| 15:33.58 | dsmith_9031 | ...the Asterisk platform? |
| 15:41.46 | *** join/#asterisk KNERD (~KNERD@netservisity.com) |
| 15:42.53 | dsmith_9031 | I've deduced as much that I will need a analog card for the server, but past that I come up short. Obviously this differs from the standard Asterisk setup because there is already an existing PBX I want it to interact with. |
| 15:46.53 | [TK]D-Fender | No such thing as "standard Asterisk" |
| 15:46.59 | [TK]D-Fender | It is whatever you configure it to be |
| 15:47.08 | [TK]D-Fender | For me it's a jukebox and coffee maker |
| 15:48.37 | *** join/#asterisk elguero (cf9a22a3@gateway/web/freenode/ip.207.154.34.163) |
| 15:48.58 | dsmith_9031 | That's awesome, I suppose my terminology was off. I guess the gist of what I'm asking, is there a manner in which I can configure an installation of the program to perform this task that you are aware of. |
| 15:49.24 | [TK]D-Fender | Want to take calls in from analog lines? Get and interface for the lines. |
| 15:49.35 | [TK]D-Fender | What to answer those calls and give a menu? Go for it. |
| 15:49.53 | [TK]D-Fender | What to call extension on your existing PBX based on that .... does it let you do that directtly? |
| 15:50.00 | [TK]D-Fender | This we don't know |
| 15:50.45 | [TK]D-Fender | Perhaps if you use station ports on it instead of line ports. Then again you'd lose things like CallerID, etc |
| 15:50.58 | [TK]D-Fender | And the entire idea is kludgy of course |
| 15:51.12 | [TK]D-Fender | What do you acttually have right now? |
| 15:55.28 | dsmith_9031 | kludgy, excellent word, to which I agree. As far as the setup we have right now, it's a Lucent Partner system with 10 phone extensions and four incoming lines. None of the phones even have CallerID presently |
| 15:57.10 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-dizthmayprfkqfgv) |
| 15:57.28 | *** join/#asterisk rmudgett (rmudgett@nat/digium/x-npoxcovthbgzimvd) |
| 15:57.38 | [TK]D-Fender | I would very serioulsy look at ripping & replacing the whole thing |
| 16:00.06 | dsmith_9031 | I would in an instant. But since I only work for the small business instead of running I don't hold the purse strings, but I see the wisdom in that. |
| 16:00.59 | dsmith_9031 | I'm mostly trying to come up with an inexpensive way to lighten the load on our office workers who currently answer and direct every call. |
| 16:01.09 | WIMPy | Remember to always look twice. |
| 16:01.55 | *** join/#asterisk mcargile (~mikec@rrcs-97-76-33-146.se.biz.rr.com) |
| 16:02.38 | [TK]D-Fender | well your current plan has you paying for an assumed 4 ports IN (required), and 4 OUT (this is the waste factor) |
| 16:03.22 | [TK]D-Fender | Far better to deal with 10 phones direct then to go through the old system at all. |
| 16:03.30 | mcargile | can someone help me with SIP REFERs getting 603 declines? I have canreinvite=yes and allowtransfer=yes |
| 16:03.41 | [TK]D-Fender | Either get a direct interface for the phones, or replace them direct and ditch all the old remains |
| 16:03.57 | [TK]D-Fender | mcargile: those settings show nothing of use |
| 16:04.15 | [TK]D-Fender | mcargile: Show us the actual failed call in full. |
| 16:05.04 | dsmith_9031 | Fender: Thank you. I'll take that and run with it. |
| 16:05.39 | mcargile | http://pastebin.com/McgUDf0k |
| 16:05.40 | [TK]D-Fender | dsmith_9031: can you show us your exact phones? |
| 16:05.46 | *** join/#asterisk Temper (~Temper@173.216.248.237) |
| 16:06.09 | dsmith_9031 | These are the ones: http://www.amazon.com/AT-MLS-12-Partner-MLS-12/dp/B000GQ1KIO |
| 16:06.15 | [TK]D-Fender | dsmith_9031: Curious if they are worth even thinking about vs replacing with used SIP phones. Older models still work great and cost next to nothing... |
| 16:06.30 | [TK]D-Fender | Wow.... 1980 is calling... |
| 16:06.39 | dsmith_9031 | You're telling me haha |
| 16:06.44 | [TK]D-Fender | yup, no supporting those things... |
| 16:06.50 | mcargile | In the process of ditching a Asterisk 1.4 system and going to new polycom phones. When I do a blind transfer it fails with a 603. |
| 16:06.55 | Temper | what is a cheap easy to integrate with asterisk sip service? |
| 16:07.08 | mcargile | setting up an asterisk 11 system. |
| 16:07.42 | [TK]D-Fender | Looks a lot like a vantage 24 system I made a CDR / logging / blacklist interface for over 20 years ago |
| 16:08.26 | [TK]D-Fender | mcargile: [Jan 6 10:38:46] SIP transfer to extension 208@defaultlog by 204@192.168.198.1 |
| 16:08.40 | [TK]D-Fender | mcargile: looks like a DIALPLAN error. nowhere to go |
| 16:09.20 | *** join/#asterisk C1ph3r (~C1ph3r@187.58.26.212) |
| 16:09.27 | dsmith_9031 | Well let me think, the business opened in 1978 and as long as they work, no one thinks twice about the system |
| 16:09.57 | mcargile | here is the defaultlog context: |
| 16:09.57 | mcargile | http://pastebin.com/MXsaQa6Z |
| 16:10.17 | mcargile | the catch all at the bottom should have grabbed it |
| 16:12.03 | [TK]D-Fender | mcargile: "dialplan show defaultlog" |
| 16:12.26 | [TK]D-Fender | dsmith_9031: That phone pic corroborates it. |
| 16:12.43 | mcargile | http://pastebin.com/d8EWqedp |
| 16:12.54 | [TK]D-Fender | dsmith_9031: If you want IVR, might as well get voicemail, callerid, etc and spend a handful of dollars. |
| 16:16.34 | [TK]D-Fender | mcargile: That output looks very filtered |
| 16:16.44 | [TK]D-Fender | mcargile: I see a call from 204 and no ack as to where it's going |
| 16:19.21 | mcargile | brand new call from beginning to end: http://pastebin.com/JDnQGbkE |
| 16:24.24 | [TK]D-Fender | Still doesn't look right |
| 16:25.02 | [TK]D-Fender | 367 204 starts a new call which I suspect it's going to try to transfer the call to. |
| 16:25.25 | *** join/#asterisk zerohalo (~zerohalo@c-174-62-142-83.hsd1.ma.comcast.net) |
| 16:25.31 | [TK]D-Fender | Where we see the "trying" we SHOULD see where it is dumpiong it... but we DON'T |
| 16:26.00 | [TK]D-Fender | #424 the call gets "answered" (only reason for a SIP 200 OK) |
| 16:26.19 | [TK]D-Fender | How the hell can it be answered... without hitting DILAPLAN |
| 16:26.28 | [TK]D-Fender | still isn't believing what he's seeing |
| 16:27.44 | mcargile | I thought that was a bit oodd |
| 16:28.25 | mcargile | on the polycom. I am hitting the transfer button. Thats when I see an INVITE sent out even though I have not hit any other keys on the phone |
| 16:29.47 | [TK]D-Fender | [Jan 6 11:18:02] INVITE sip:asterisk@192.168.198.1:5060;transport=udp SIP/2.0 |
| 16:29.59 | [TK]D-Fender | That is a call to "asterisk" |
| 16:30.12 | [TK]D-Fender | Which you certainly don't have a match for |
| 16:30.16 | [TK]D-Fender | But I don't see it stating that it's looking |
| 16:30.50 | [TK]D-Fender | I'm betting this phone's provisioning has been suitably mangled |
| 16:31.32 | mcargile | quite possible |
| 16:36.06 | mcargile | someone had me disable URL dialing on the polycom (no clue why). Just reenabled it. That seems to have fixed it. |
| 16:36.40 | [TK]D-Fender | You shouldn't have URL dialing on |
| 16:37.22 | mcargile | nope nevermind |
| 16:37.58 | mcargile | regular transfers work, blind doesnt which is what I was testing. |
| 16:47.55 | *** part/#asterisk marceloamorim (~marcelo@189-90-192-72.isimples.com.br) |
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| 17:40.25 | *** part/#asterisk bcalhoun (~bcalhoun@71-14-6-250.static.gwnt.ga.charter.com) |
| 17:40.28 | gtTuna | quick question/poll i wanted to bounce off you guys regarding CDRs -- what method(s) do you guys use to determine if a call is considered local or long distance? |
| 17:41.09 | *** part/#asterisk mitchthecleaver (~mitch@47-32-123-58.dhcp.gwnt.ga.charter.com) |
| 17:42.25 | [TK]D-Fender | The very thing that defines it |
| 17:43.00 | [TK]D-Fender | When my plan says "US48" for the same price ... what does long-distance even MEAN? |
| 17:43.04 | [TK]D-Fender | Whose MAP? |
| 17:47.54 | gtTuna | well, that might be fine in your configuration |
| 17:49.02 | *** join/#asterisk azerus (~badass@unaffiliated/badass) |
| 17:49.34 | [TK]D-Fender | And you need to define what "long distance" actually means in your case |
| 17:49.41 | [TK]D-Fender | I don't know. |
| 17:49.49 | [TK]D-Fender | I doubt too many others do. |
| 17:50.23 | AsteriskNoob | should canreinvite=no when nat=froce_rport |
| 17:50.36 | gtTuna | calls from a NPA-NXX to another NPA-NXX that are not considered local |
| 17:50.49 | [TK]D-Fender | Considered local by who? |
| 17:51.43 | gtTuna | why is it that any time I come here and ask a question, you're always the patronizing asshole that responds? |
| 17:52.05 | [TK]D-Fender | First this isn't patronizing, and second I help a lot of people in here |
| 17:52.15 | [TK]D-Fender | You seem to be completely missing the point |
| 17:52.19 | [TK]D-Fender | 'This is YOUR TELCO |
| 17:52.28 | [TK]D-Fender | Mikne doesn't draw the borders thhat YOURS does. |
| 17:52.39 | [TK]D-Fender | Your map is YOURS. |
| 17:52.47 | [TK]D-Fender | Go get your telco's list |
| 17:53.18 | [TK]D-Fender | I clearly don't have it since my plan does draw a boder that isn't international or divided by an entire ocean |
| 17:53.36 | [TK]D-Fender | This is not patronizizing. This is a simple matter of fact |
| 17:53.56 | [TK]D-Fender | Some telcos in the US may say "your entire state is "local" |
| 17:54.10 | [TK]D-Fender | Some not. Immediate difference right from there |
| 17:54.46 | [TK]D-Fender | Places that would take me an hour to drive to would be free and one I could walk to in half that time might cross a zone line |
| 17:55.13 | *** join/#asterisk youtmon (~yout@c-73-46-212-142.hsd1.fl.comcast.net) |
| 18:02.37 | AsteriskNoob | should canreinvite=no when nat=froce_rport ? |
| 18:03.27 | [TK]D-Fender | You should never allow reinvites wherever ANY NAY it involved |
| 18:03.41 | [TK]D-Fender | and the setting is directmedia, not "canreeinvite". |
| 18:03.49 | [TK]D-Fender | This changed over half a decade ago |
| 18:24.41 | *** join/#asterisk aness (~aness@cm-84.215.107.62.getinternet.no) |
| 18:30.38 | mcargile | [TK]D-Fender: pretty sure I have narrowed down the problem. THat weird invite is the phone trying to put the call into Music on hold. The strange sip:asterisk@blahblahblah was because my coworker set the caller id name of the call to the phone but not the number. |
| 18:31.28 | [TK]D-Fender | * shouldn't be trying to contact a 3rd party MoH server |
| 18:31.36 | [TK]D-Fender | the standard hold should work fine |
| 18:31.43 | [TK]D-Fender | fix your provisioning |
| 18:32.34 | mcargile | It works perectly when I call from phone A to phone B and then try transfering to phone C. But when the AGI triggers the call to phone B and B tries to transfer to C it fails. |
| 18:33.17 | [TK]D-Fender | AGI has nothing to do with with that borked invite it is sending out |
| 18:33.37 | *** join/#asterisk DragonAzul (~DragonAzu@187.208.5.129) |
| 18:37.22 | mcargile | here is that invite that is created when I hit the transfer button on a phone to phone transfer: http://pastebin.com/L97xdAsP it ends with the call being placed into MoH. You are saying it is not suppose to do that? |
| 18:37.55 | mcargile | or is something malformed with the other Invite? |
| 18:39.15 | *** join/#asterisk ipn (d8bb0020@gateway/web/freenode/ip.216.187.0.32) |
| 18:39.40 | *** join/#asterisk tzafrir (~tzafrir@bzq-179-40-172.cust.bezeqint.net) |
| 18:40.10 | ipn | can someone help me out real quick. I have a server already running centos. I want to install the asterisk.iso on that server. How do I do that? Do I need to create a bootable USB or can I wget and do it that way? |
| 18:40.51 | [TK]D-Fender | Boot it however iut can boot |
| 18:41.01 | [TK]D-Fender | cd/dvd/usb/whatever |
| 18:41.12 | [TK]D-Fender | wahtever your machine supports |
| 18:41.23 | ipn | if i wget, how do i tell the machine to boot that ISO? |
| 18:41.33 | *** part/#asterisk DragonAzul (~DragonAzu@187.208.5.129) |
| 18:41.37 | [TK]D-Fender | BURN IT <- |
| 18:41.38 | WIMPy | Or network, but that will require RAM, the size of that image. |
| 18:41.51 | ipn | I dont have a cd/dvd drive on the machine |
| 18:41.55 | WIMPy | Use dd to copy it to some spare medium. |
| 18:42.09 | WIMPy | USB stick memory card, spard HDD. |
| 18:42.12 | [TK]D-Fender | ^ |
| 18:42.22 | [TK]D-Fender | [13:41][TK]D-Fendercd/dvd/usb/whatever |
| 18:43.37 | ipn | not sure i follow |
| 18:44.15 | [TK]D-Fender | make a bootable USB drive |
| 18:44.27 | [TK]D-Fender | or get a USB optical drive. |
| 18:44.29 | [TK]D-Fender | Or whatever |
| 18:44.38 | ipn | guess thats easiest - and I can change boot order for USB |
| 18:44.47 | ipn | but I was hoping if I did a wget how would I do that |
| 18:45.19 | mcargile | wget the iso. use dd to copy it to the usb. reboot off the usb. |
| 18:46.29 | *** join/#asterisk mahlon (~mahlon@martini.nu) |
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| 19:11.43 | sweettea | dd if=as.iso of=/dev/sd_usb |
| 19:13.49 | *** join/#asterisk lambda79 (~lambda79@unaffiliated/lambda79) |
| 19:17.41 | lambda79 | hi everyone, do you know any massively scalable and fault-tolerant system integrating Asterisk ? |
| 19:20.02 | lambda79 | i was looking Digium's Switchvox but i guess it is a only commercial offer |
| 19:21.17 | newtonr | You'll probably want to define "massively scalable" and "fault-tolerant" in order to get good recommendations |
| 19:21.36 | lambda79 | yes |
| 19:21.52 | lambda79 | massively scalable:Â from 5Â to 500Â users |
| 19:22.19 | lambda79 | fault tolerant:Â phone could survive the failure of its call-setup server and later transfer the same call |
| 19:22.45 | robmal | Google: VMWare |
| 19:24.08 | lambda79 | i'd say massively autoscalable and extentd to 5000Â users |
| 19:24.49 | robmal | Still, one HA cluster with asterisk can do this. |
| 19:24.54 | lambda79 | robmal: vmware is very time consuming and storage consuming |
| 19:25.13 | robmal | Explain. |
| 19:26.08 | lambda79 | it is not easy to cloud provision efficiently a UCommunication system |
| 19:26.13 | lambda79 | with VMware |
| 19:26.20 | lambda79 | (at least i am not good at) |
| 19:27.00 | robmal | Yes. |
| 19:27.47 | lambda79 | i was looking for something more 'readily' |
| 19:28.43 | lambda79 | and maybe more 'carrier grade oriented' |
| 19:29.24 | robmal | Something like, lets say, Switchvox? |
| 19:29.33 | lambda79 | yes but opensource |
| 19:29.47 | robmal | Oh. |
| 19:29.54 | robmal | There's freepbx. |
| 19:30.00 | lambda79 | ... |
| 19:30.06 | robmal | What? |
| 19:30.12 | lambda79 | not an ipbx alone |
| 19:30.16 | WIMPy | chews popcorn |
| 19:30.50 | robmal | But freepbx has all that you need. |
| 19:31.02 | lambda79 | hm freepbx != freeswitch you re true |
| 19:31.19 | lambda79 | ok gonna have a look at that |
| 19:31.49 | robmal | They even have a support channel here on freenode. Very helpful people. |
| 19:32.45 | *** join/#asterisk goddva (~goddva@77.40.154.242) |
| 19:32.48 | robmal | But be aware. Once you say you have a freepbx problem on #asterisk your cows will stop giving milk and your future children will be born with a hump. |
| 19:33.00 | lambda79 | :-) |
| 19:33.53 | lambda79 | robmal: |
| 19:33.56 | lambda79 | last question |
| 19:34.35 | lambda79 | what do you sincerely think of Kazoo in a term of cloud telephony system ? (if you ever know ) |
| 19:35.00 | robmal | Never heard of it. |
| 19:35.06 | lambda79 | ok |
| 19:35.33 | robmal | Looks nice. |
| 19:35.55 | lambda79 | no special advice about freeswitch ? |
| 19:36.02 | WIMPy | That was a quick sale! |
| 19:37.03 | robmal | Nobody uses freeswitch. |
| 19:37.22 | lambda79 | that is the main issue of Kazoo |
| 19:38.22 | robmal | Maybe you'll get a reward for being their first customer. |
| 19:38.46 | robmal | Like WinRAR. |
| 19:39.23 | lambda79 | i am not gonna drop asterisk, i get used to it |
| 19:39.42 | ipn | so I downloaded the ISO and I enter: dd if=source.iso /dev/sde1 |
| 19:39.52 | ipn | but when i try to boot from usb it says missing operating system |
| 19:40.17 | ipn | is it a bad iso or did i do something wrong |
| 19:40.44 | WIMPy | You should copy to sde, not sde1. |
| 19:41.04 | ipn | damn - ok |
| 19:41.07 | [TK]D-Fender | It also has to be an img that is meant to be nurned tto USB. |
| 19:43.39 | [TK]D-Fender | burned* |
| 19:54.35 | sweettea | what wimpy said :D |
| 19:54.57 | WIMPy | didn't say anyting |
| 19:55.00 | WIMPy | :-) |
| 20:04.32 | *** join/#asterisk UncleKiwi (~UncleKiwi@unaffiliated/unclekiwi) |
| 20:08.17 | lambda79 | do any of you know Xivo, is it a reliable Asterisk appliance ? |
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| 20:49.17 | Spengler | With POTS we use to have a scenario where in a bank of three numbers if one was busy an incoming call would roll to the next line |
| 20:49.35 | Spengler | how does this work with VOIP, on a service like Vitelity for example |
| 20:50.58 | [TK]D-Fender | Depends what service you pay them for |
| 20:51.35 | [TK]D-Fender | With POTS each line typically has a number associated to it. Virtually no other kind of circuit is limited in this way |
| 20:52.16 | [TK]D-Fender | take ISDN (PRI/BRI). These support upwards of 30 channels to qhich tthere is no fixed number of DID's you can associate to it. |
| 20:52.38 | Spengler | i see |
| 20:52.42 | [TK]D-Fender | I have a T1-PRI at the office (23 channels) and over 100 DID's on the account |
| 20:52.48 | WIMPy | And often no fixed number of calls, either. |
| 20:53.29 | Spengler | so i spoke with the rep and i believe she said 4 inbound channels for each did |
| 20:53.49 | Spengler | If I want more I guess I should just increase it |
| 20:54.05 | Spengler | but I think 4 should be enough as they only had 3 pots lines previously |
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| 20:55.47 | *** join/#asterisk acidfoo (45a5a03c@gateway/web/freenode/ip.69.165.160.60) |
| 20:55.48 | [TK]D-Fender | They have many different package options. |
| 20:56.24 | acidfoo | (Asterisk 11) In sip.conf and rtp.conf I've set icesupport=no, I'm still getting this error => WARNING[1858][C-00000000]: res_rtp_asterisk.c:800 ast_rtp_ice_set_role: Set role failed; no ice instance (0x7f4398024b48) |
| 20:56.31 | acidfoo | any Idea ?? thank you |
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| 21:16.51 | file | that warning message has been silenced in later versions |
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| 21:25.38 | acidfoo | ok |
| 21:30.28 | acidfoo | when trying to make a call, asterisk always repond with 403 forbidden, directmedia=no, icesupport=no, nat=yes |
| 21:31.02 | acidfoo | I don't know what should I check first, I presume I have something misconfigured in sip.conf :| |
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| 21:48.42 | NeuhNeuh | Hello everybody :) |
| 21:49.20 | NeuhNeuh | I have a question, I try to put a « Waiting music » for my phone, but I get .... Not sound when I call my phone number http://pastebin.com/2N1RGrPF |
| 21:49.26 | NeuhNeuh | How fix that ? :x |
| 21:49.33 | NeuhNeuh | I have a .ogg file into /var/lib/asterisk/moh |
| 21:50.00 | NeuhNeuh | http://pastebin.com/L6WDNQ8F And its my musiconhold.conf |
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| 22:38.37 | NeuhNeuh | Ok I progress lol |
| 22:38.56 | NeuhNeuh | I have waiting music but block on music on hold (SIP don't ring) |
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