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| 01:31.55 | f2knight | Pride day. ... Gay pride day? |
| 01:32.01 | f2knight | thought that was june 2nd. |
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| 02:25.03 | devcoder | anyone know if there is an asterisk patch for festival v2.1 |
| 02:25.24 | devcoder | 1.4.3 is too old and has compile issues because of it |
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| 02:30.39 | devcoder | eh 1.96 seems to work, just wish it wasn't "beta" |
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| 06:25.49 | lkthomas | hey guys |
| 06:25.56 | lkthomas | does anyone using asterisk with cisco phone set ? |
| 06:26.16 | lkthomas | is it possible to do auto provisioning for cisco phone set button ? |
| 06:27.06 | p3nguin | I don't know anything about a set button, but I use Cisco phones on Asterisk. |
| 06:27.36 | lkthomas | so do you need to edit XML one by one ? |
| 06:28.09 | p3nguin | You typically have a default and one for each phone. |
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| 06:31.37 | lkthomas | well |
| 06:31.45 | lkthomas | we are using a dir call zultys |
| 06:32.03 | lkthomas | inside that, it have macaddress.conf.xml |
| 06:32.11 | lkthomas | so each phone have their own xml file |
| 06:32.44 | lkthomas | and then each xml the authname and authpassword will be different, as well as the featureLabel |
| 06:35.01 | lkthomas | any way to automate it ? |
| 06:36.08 | singler | lkthomas: if phones can take config by http, you can automate it by using some sort of script |
| 06:36.44 | lkthomas | that's a good question, I am not sure on this |
| 06:36.46 | singler | or you can write a script/program which would rewrite all needed files after edit |
| 06:36.47 | lkthomas | p3nguin, any idea? |
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| 07:02.18 | schmidts | good morning |
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| 07:12.05 | devil_evoxxx | hi all |
| 07:13.57 | schmidts | does anyone of you use chan_ss7? i have read that they have load tested it with 120 concurrent channels but if i want to use it, it would be a little more than this. Has anyone experiance with the power of chan_ss7? |
| 07:21.29 | saxa | heh, strange, when I receive a call I can talk with the other party, when I place a call I cant hear anything. But strangely when I connect to my voicemail I hear the voice announcing the mailbox. |
| 07:21.35 | saxa | any ideas ? |
| 07:21.43 | saxa | its a SIP phone |
| 07:21.52 | saxa | grandstream GXP 2120 |
| 07:22.03 | devil_evoxxx | nat / rtp problem? |
| 07:22.14 | saxa | probably not, its on the same net |
| 07:22.26 | saxa | 192.168.0.100 phone |
| 07:22.33 | saxa | 192.168.0.5 * |
| 07:22.50 | saxa | i have probably misconfigurred something |
| 07:23.11 | saxa | need to try with sip debug on |
| 07:23.29 | devil_evoxxx | are you trying to make call internally? |
| 07:23.55 | kaldemar | saxa: place a call where? |
| 07:24.32 | kaldemar | saxa: definitely sounds like a NAT issue if you're dialing to outside network with SIP. |
| 07:24.45 | saxa | <--- Reliably Transmitting (NAT) to 192.168.0.100:5060 ---> |
| 07:24.45 | saxa | SIP/2.0 401 Unauthorized |
| 07:24.54 | saxa | does this means anything ? |
| 07:24.57 | kaldemar | no. |
| 07:25.18 | saxa | ok maybe I have enabled the nat |
| 07:25.26 | kaldemar | the phone is not the point of interest here, but how asterisk is configured to function behind the NAT: |
| 07:25.26 | saxa | let me try to play with that part |
| 07:26.03 | kaldemar | you seem to have nat=yes under [general], add localnet and externaddr. |
| 07:26.28 | saxa | kaldemar: I place a call from my sip phone to my mobile phone, it rings, I answer on the mobile, but I do not hear anything on no one side |
| 07:26.38 | saxa | kaldemar: yes I have those |
| 07:28.28 | kaldemar | pastebin a sip debug of a call and the relevant parts of your sip.conf. |
| 07:28.32 | saxa | just removed nat=yes and same thing |
| 07:31.50 | saxa | http://pastebin.com/UcHSsnY1 <- sip debug on |
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| 07:33.51 | saxa | http://pastebin.com/GDUpKf8Q <-- sip.conf |
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| 07:34.58 | kaldemar | saxa: that sip debug is completely useless. pastebin a whole call. |
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| 07:35.31 | kaldemar | externhost=217.223.95.245:5060 |
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| 07:36.43 | kaldemar | externhost is supposed to get a host as a value, not an address. if you have a static address, use externaddr. and set nat=yes. |
| 07:37.51 | saxa | oh let me try |
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| 07:38.08 | saxa | kaldemar: basically the call I places was that sip debug |
| 07:38.16 | saxa | I mean |
| 07:38.31 | saxa | I have enabled sip debug, and then placed a call |
| 07:38.39 | saxa | that was what I got. |
| 07:39.01 | lkthomas | guys |
| 07:39.03 | lkthomas | interesting |
| 07:39.11 | lkthomas | I could ssh into my cisco phone |
| 07:39.16 | lkthomas | but reboot command is blocked |
| 07:39.36 | kaldemar | saxa: let the call connect before hanging up from the sip phone side. |
| 07:40.19 | saxa | i can wait all the time, but all is mutted |
| 07:40.41 | saxa | the sip.conf should be ok in my opinion |
| 07:40.43 | *** join/#asterisk asterisk-Tester (~RAMYT@193.227.170.247) |
| 07:40.57 | saxa | i use that same sip.conf on another machine and it works perfectly |
| 07:41.08 | saxa | maybe i have some wrong settings in the phone |
| 07:41.14 | kaldemar | saxa: the sip.conf you pasted was not ok. |
| 07:41.20 | saxa | :) |
| 07:41.42 | kaldemar | will you pastebin a whole call? |
| 07:41.45 | saxa | i corrected that extrenaddr=IPNUM |
| 07:41.56 | saxa | so it does not change anything at all |
| 07:45.00 | saxa | http://pastebin.com/GgU6hfhh <-- here is a new sip debug |
| 07:45.21 | saxa | thats the one with externaddr= |
| 07:45.28 | saxa | and I disconnected the call on my cell phone |
| 07:46.59 | kaldemar | did you only enable debug for the phone peer? |
| 07:47.23 | kaldemar | the ITSP part is the informative one. |
| 07:48.32 | devil_evoxxx | kaldemar: are you using ast 1.8.5 or > ? |
| 07:48.58 | *** join/#asterisk bjhaid (~abejide@41-147.rv.ipnxtelecoms.com) |
| 07:49.46 | saxa | kaldemar: yes, I only enable sip set debug on |
| 07:50.18 | jacc0 | @lkthomas: you can also push an xml to tha cisco spa phone to reboot it |
| 07:51.20 | kaldemar | saxa: and how does the call go out of your asterisk box? |
| 07:51.34 | kaldemar | devil_evoxxx: how so? |
| 07:51.47 | jacc0 | <PROTECTED> |
| 07:51.48 | saxa | kaldemar: dahdi analog line |
| 07:52.13 | saxa | kaldemar: if i place a call from another phone, on an FXS |
| 07:52.24 | saxa | kaldemar: I get it call and I can hear everything |
| 07:52.55 | saxa | it seems something is still wrong in my sip part probably |
| 07:53.14 | devil_evoxxx | kaldemar: because i'm having a strange problem.. if i try to debug rtp stream with "rtp set debug ip ip-of-the-phone" i cant see nothing |
| 07:53.50 | devil_evoxxx | i'm using 1.8.7 |
| 07:54.37 | kaldemar | devil_evoxxx: do you see any packets with tcpdump for example? are you supposed to see any packets? is directmedia enabled? |
| 07:55.21 | devil_evoxxx | i suppose that packet is present, because i try to make a call. directmedia for phone peer is set to no |
| 07:55.38 | bjhaid | hi, I am using originate with asterisk, how do I get the status of a call? |
| 07:55.39 | kaldemar | saxa: now you tell that it's an analog line... well, check with rtp debug that you see packets going to both directions. |
| 07:55.48 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 07:55.57 | devil_evoxxx | i migrate my configuration from 1.4.x to 1.8.x, and i change canreinvite=no with directmedia=no |
| 07:56.07 | kaldemar | bjhaid: which originate? where do you want to see the status? |
| 07:56.44 | kaldemar | devil_evoxxx: do you hear sound on a call? |
| 07:56.50 | devil_evoxxx | my phone is all under nat (nat=yes) |
| 07:57.11 | devil_evoxxx | kaldemar: yes, the call is ok..i can hear in both case (receive and make call) |
| 07:57.12 | bjhaid | kaldemar: originate to dial an extension from a script, and I want to determine the status of the called party, if he is busy, not available or whatever |
| 07:58.56 | saxa | kaldemar: Sent RTP packet to 192.168.0.100:5004 (type 00, seq 009249, ts 234560, len 000160) |
| 07:59.09 | saxa | kaldemar: i just get a bunch of those |
| 07:59.21 | saxa | and now has not even placed a call |
| 08:03.07 | bjhaid | kaldemar: any ideas? |
| 08:03.23 | kaldemar | bjhaid: deliver DIALSTATUS variable contents to the script after the Dial app. |
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| 08:06.11 | bjhaid | okay |
| 08:06.17 | bjhaid | thanks |
| 08:06.20 | bjhaid | would look that up |
| 08:10.02 | devil_evoxxx | kaldemar: when you say if direcmedia is enabled you intend in general section of sip.conf |
| 08:10.05 | devil_evoxxx | or |
| 08:10.12 | devil_evoxxx | on single peer of the phone? |
| 08:11.52 | *** join/#asterisk DennisG (~DennisG@145.93.253.128) |
| 08:15.48 | kaldemar | devil_evoxxx: i mean enabled for the parties that are involved in the call. putting the setting under a peer definition doesn't make it any more enabled for a peer. as long as the possibly correct general setting is not overrun in the peer configuration. |
| 08:20.11 | devil_evoxxx | kaldemar: ok, then my configuration is correct |
| 08:20.17 | devil_evoxxx | but, i still can see rtp stream :( |
| 08:20.27 | devil_evoxxx | and i don't understand where i'm wrong |
| 08:22.16 | GreatSUN | good morning! |
| 08:22.25 | kaldemar | devil_evoxxx: what does sip debug say? |
| 08:22.32 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 08:22.58 | kaldemar | devil_evoxxx: what is the scenario like in general? what are the endpoints? what is the network setup like? |
| 08:24.41 | devil_evoxxx | my server have a public ip address. Is connected with my telco provider directly in sip. the call go trought the peer with my provider, and the configuration of this peer is --> http://pastebin.com/sGVE66R0 |
| 08:26.17 | devil_evoxxx | my phone is registered directly on this server (where is configured the peer with telco provider), and all of my phone are behind nat. for example the configuration of my phone is : http://pastebin.com/2JEfZ4h2 |
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| 08:27.20 | devil_evoxxx | the call is ok, but i can't debug rtp stream like i did with 1.4 |
| 08:29.31 | *** join/#asterisk singler (~singler@81-7-123-162.static.zebra.lt) |
| 08:31.54 | kaldemar | devil_evoxxx: and sip debug? |
| 08:32.47 | devil_evoxxx | sip debug work |
| 08:33.04 | *** join/#asterisk Fritz09 (~Adium@pop1-73.catv.wtnet.de) |
| 08:33.18 | devil_evoxxx | i can see the handshaking.. |
| 08:34.52 | devil_evoxxx | the strange thing is that if i set rtp set debug on i can see all rtp stream (also the stream of my phone). In particular i have seen that rtp stream was started on ip-phone:16674, and if i set rtp set debug ip ip-phone:16674 i can see 1-way rtp (and this is correct) |
| 08:35.18 | devil_evoxxx | but when i want to debug rtp in both way ( from and to), i've to set only the ip of the phone (and asterisk say ip-phone:0) |
| 08:35.22 | devil_evoxxx | but this , don't work |
| 08:36.36 | devil_evoxxx | dont'work in 1.8.7.. in 1.4 work |
| 08:38.43 | kaldemar | you lost me there. |
| 08:40.17 | GreatSUN | can someone of you specialists tell me where I can configure asterisk to listen on port 5060? |
| 08:40.42 | kaldemar | devil_evoxxx: "if i try to debug rtp stream with "rtp set debug ip ip-of-the-phone" i cant see nothing" "if i set rtp set debug ip ip-phone:16674 i can see 1-way rtp" |
| 08:41.02 | kaldemar | well, if actually "can't see nothing" means that you see something... |
| 08:41.21 | kaldemar | GreatSUN: be more precise. what do you want to listen to port 5060? |
| 08:41.50 | GreatSUN | kaldemar: i want it to listen on 5060 udp and accept new client phones |
| 08:42.04 | kaldemar | GreatSUN: what protocol? SIP? |
| 08:42.06 | GreatSUN | configured via extensions they should be able to login |
| 08:42.08 | GreatSUN | yeah |
| 08:42.10 | GreatSUN | for sure |
| 08:43.19 | kaldemar | GreatSUN: sip.conf. however, there is no such thing as login in SIP and phones are configured as devices in sip.conf, extensions are part of the dialplan and configured in extensions.conf. |
| 08:43.23 | kaldemar | ~book |
| 08:43.23 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/ or ~buybook |
| 08:43.30 | *** join/#asterisk Tribbers (~joey@host217-37-142-238.in-addr.btopenworld.com) |
| 08:44.06 | devil_evoxxx | sorry kaldemar ..i was wrong to write. So, if i set "rtp set debug ip ip-of-the-phone" no rtp stream shown in asterisk cli, but if i set rtp set debug ip ip-phone:16674 i can see 1-way rtp stream |
| 08:46.08 | GreatSUN | lol |
| 08:46.17 | GreatSUN | now I got my error |
| 08:46.24 | GreatSUN | was running asterisk with wrong user |
| 08:46.25 | GreatSUN | :D |
| 08:53.16 | devil_evoxxx | kaldemar: sorry i was wrong to write. So, if i set "rtp set debug ip ip-of-the-phone" no rtp stream shown in asterisk cli, but if i set rtp set debug ip ip-phone:16674 i can see 1-way rtp stream |
| 08:58.31 | jacc0 | I want to add a 3th party to an existing call; what would be the best approuch? |
| 09:00.06 | *** join/#asterisk benngard (~mabe@213.88.138.230) |
| 09:00.12 | jacc0 | *approach |
| 09:00.32 | jacc0 | do I need meedme or confbridge for that? |
| 09:02.45 | kaldemar | jacc0: if you're using sip phones, no. the phone probably implements it. |
| 09:03.44 | *** join/#asterisk [sr] (~Unknowned@pa8-84-91-197-8.netvisao.pt) |
| 09:05.22 | *** join/#asterisk Nasga (~Nasga@186.212.10.93.rev.sfr.net) |
| 09:05.36 | jacc0 | I want to do it server side |
| 09:07.02 | jacc0 | I want to call a 3th sip client(that has auto answer) en connect it muted; then bij pressing dtmf 3 I want to unmute the 3th client |
| 09:07.13 | jacc0 | end mute the caller |
| 09:07.35 | jacc0 | then if you pres dtmf 1 I want to unmute the caller and mute the 3th client |
| 09:10.10 | jacc0 | so I can toggle between the caller en the 3th client using dtmf 1&3 |
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| 09:17.50 | mgco3 | is there a way to use regex in the dial function? something like exten => 1000,Dial(SIP/_20X0,20) |
| 09:19.53 | kaldemar | jacc0: you probably need to make an extension for that and use confbridge/meetme, and activate it through features.conf. |
| 09:20.12 | kaldemar | mgco3: what are you tring to achieve? what do you want X to include? |
| 09:20.52 | SunTsu | I guess - call 2000, 2010, 2020 and so on |
| 09:21.14 | mgco3 | kaldemar, at the moment it looks like exten => 1000,Dial(SIP/2010&SIP/2020&SIP/2030,20) |
| 09:21.24 | mgco3 | SunTsu, ack :) |
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| 09:31.37 | kaldemar | mgco3: no, not really. a dirty workaround would be to #exec a script that sets a global variable and use the variable as an argument for app Dial. |
| 09:39.29 | *** join/#asterisk joat (~joat@ip70-160-216-251.hr.hr.cox.net) |
| 09:42.27 | bjhaid | I specify a file (.g723) which is located in the home folder for asterisk playback command, but it doesnt play it, any reason why this occurs |
| 09:43.57 | kaldemar | bjhaid: what does your Playback command look like? |
| 09:44.01 | lkthomas | anyone using cisco phone ? |
| 09:44.12 | lkthomas | I try to config cisco phone directory button to read an xml |
| 09:44.26 | lkthomas | it does read but I can't find anything related with directory service |
| 09:47.27 | bjhaid | 400,1,Playback(/home/vas/vm-toforward) |
| 09:47.44 | bjhaid | kaldemar: 400,1,Playback(/home/vas/vm-toforward) |
| 09:48.34 | *** join/#asterisk zealiod (~zealiod@94-30-97-220.xdsl.murphx.net) |
| 09:49.37 | zealiod | I am setting up ODBC for realtime mapping but have received this error. I have tested the ODBC connection at the command line⦠all works OK. What do you think? |
| 09:49.37 | zealiod | Realtime mapping for 'sippeers' found to engine 'odbc', but the engine is not available |
| 09:50.18 | bjhaid | kaldemar: any other location i can put the file where asterisk can locate easily? |
| 09:50.27 | bjhaid | without specifying the path |
| 09:50.44 | kaldemar | bjhaid: you should be able to put it anywhere, just as long as the file is readable by asterisk. |
| 09:51.49 | kaldemar | bjhaid: if no path is specified, the file is looked for in astvarlibdir/sounds/languageprefix/ |
| 09:52.00 | lkthomas | does anyone use cisco phone ? |
| 09:52.07 | kaldemar | bjhaid: which is for example /var/lib/asterisk/sounds/en |
| 09:52.13 | bjhaid | kaldemar: its readable by asterisk, from a script it worked. from an extension, it doesnt, wondering what exactly is wrong, though the file is in the same location with the script, so it works, but wondering why it would work here |
| 09:52.28 | bjhaid | okay let me look that location and dump the file there, thanks |
| 09:52.38 | kaldemar | bjhaid: do you have format_g723.so loaded? |
| 09:56.21 | bjhaid | well, not too sure, but it worked with the script, so i guess it should work straight away, since the script loads it via asterisk |
| 09:56.35 | bjhaid | let me dump the file in the path you specified and see if it works |
| 10:01.31 | lkthomas | very funny that I defined directoryURL and directory menu does not appear |
| 10:02.45 | bjhaid | kaldemar: thanks worked |
| 10:03.49 | lkthomas | anyone using cisco ip phone ?! |
| 10:03.52 | lkthomas | I really need some help here |
| 10:24.30 | *** join/#asterisk libryder (~dryder@host-74-211-9-139.beyondbb.com) |
| 10:25.21 | *** join/#asterisk singler (~singler@81-7-123-162.static.zebra.lt) |
| 10:35.13 | *** join/#asterisk tonsofpcs (~tonsofpcs@cpe-69-205-240-64.stny.res.rr.com) |
| 10:43.05 | *** join/#asterisk bn-7bc (bjarne@pdpc/supporter/active/bn-7bc) |
| 10:43.28 | benngard | hmm, i am playing a little with asterisk 10b2, is there some new feature to get number presentation to work? |
| 10:46.04 | *** join/#asterisk cyborg-one (1000@31.31.109.231) |
| 10:46.21 | *** join/#asterisk MarKsaitis (~MarKsaiti@94-30-69-205.xdsl.murphx.net) |
| 10:53.27 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 10:58.39 | *** join/#asterisk alfadir (~alfadir@79.141.248.181) |
| 11:03.49 | benngard | From: <sip:+46317998975@pbx.telenor.se;user=phone>; <- could the + sign cause any problems? |
| 11:05.48 | libryder | benngard: yes, the + symbol specifies country code |
| 11:06.22 | lkthomas | anyone using cisco ip phone ? |
| 11:07.12 | benngard | libryder: i know that |
| 11:07.25 | libryder | oh, sorry... |
| 11:07.32 | wdoekes2 | benngard: it shouldn't cause problems.. it can when bad sip parsers are present |
| 11:07.44 | benngard | libryder: i wonder if that could cause the number presentation to break in some way |
| 11:08.44 | benngard | did a very simple test, 1 softphone, 1 phone and a sip trunk to telenor, softphone to phone, number pres is working, telenor to phone no number presentation |
| 11:09.17 | benngard | and the only big diff i see is that telenor alwas prepend with for example +46 |
| 11:09.51 | wdoekes2 | perhaps your phone has issues with the +. my linksys doesn't mind the plus, except when using the 'lcr' button, when it suddenly strips the plus |
| 11:09.58 | SunTsu | lkthomas: is your goal to have everybody in this channel to respond with yes or no? |
| 11:10.08 | benngard | i did remove the plus in the dialplan |
| 11:10.56 | benngard | Set(CALLERID(num)=0${CALLERID(num):3}) |
| 11:11.09 | benngard | just removed +46 and added a 0 |
| 11:12.26 | lkthomas | SunTsu, Actually I have problem on directories button |
| 11:12.45 | lkthomas | the services button work when I press it, which direct to same XML as directories button |
| 11:12.52 | benngard | i see the call coming in, From: <sip:+46317998975@pbx.telenor.se; but later i see, From: "asterisk" <sip:asterisk@10.242.14.183>; |
| 11:13.10 | SunTsu | lkthomas: then go ahead and ask that. People in IRC normally don't know on questions as general as your original one |
| 11:13.25 | SunTsu | s/know/react/ |
| 11:13.44 | benngard | and my phone ofc displays asterisk :( |
| 11:14.06 | lkthomas | I followed this URL on "Company telephone Directory" but after I press directories button, it does not show directory menu:http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+79xx |
| 11:14.18 | lkthomas | XML already triple checked which is correct |
| 11:15.37 | wdoekes2 | benngard: ok.. so you had an asterisk in between.. and it's not parsing the CLI from telenor properly (OR is simply replacing it for whatever reason) ok.. that clarifies things |
| 11:16.47 | wdoekes2 | benngard: you probably want to pb some stuff from the console (incoming SIP INVITE, any parsing comments from asterisk) |
| 11:17.36 | benngard | sec |
| 11:19.11 | benngard | http://pastebin.com/h5tyz0dy |
| 11:19.42 | libryder | benngard: omfg |
| 11:19.55 | *** join/#asterisk corretico (~luis@201.201.44.82) |
| 11:21.09 | wdoekes2 | benngard: do you have fromuser=asterisk in your sip.conf? |
| 11:22.16 | benngard | no |
| 11:23.07 | wdoekes2 | pb the sip.conf, if you will (clean out the secrets and other stuff we don't need to know about) |
| 11:23.29 | libryder | wdoekes2 or benngard - you write dialplans directly in asterisk? |
| 11:23.50 | wdoekes2 | not in asterisk, but in vim |
| 11:24.33 | *** join/#asterisk zealiod (~zealiod@94-30-97-220.xdsl.murphx.net) |
| 11:24.59 | benngard | http://pastebin.com/PPtKszPg |
| 11:24.59 | zealiod | Can anyone please help me with this error: |
| 11:25.00 | zealiod | Realtime mapping for 'sippeers' found to engine 'odbc', but the engine is not available |
| 11:25.23 | wdoekes2 | module load res_odbc |
| 11:25.29 | wdoekes2 | module load res_config_odbc |
| 11:25.46 | benngard | libryder: yes |
| 11:26.06 | libryder | wdoekes2: i don't know if you are doing this professionally or personally, but it might be worth it to look into adhearsion: http://adhearsion.com/ |
| 11:26.35 | libryder | asterisk config on small deployments is a pita |
| 11:27.09 | jacc0 | okay I've have something strange; it might be a bug |
| 11:28.09 | zealiod | wdoekes2: thanks, first one said already loaded - second one loaded - how do I make sure this loads default? what do i need in the modules.conf? |
| 11:28.26 | jacc0 | I have client A calling in; then multiple sip clients(B) are called. The first one to pick up and press dtmf 2 will be bridged with client A |
| 11:29.18 | jacc0 | in the dialplan of client B I've added a feature for dtmf 3 ; when pressed it bridges with some other (already called client (C)) |
| 11:29.37 | jacc0 | the strange thing is that client c always gets connected with client A |
| 11:29.49 | jacc0 | I would expact it to be connected to client B |
| 11:29.50 | wdoekes2 | zealiod: see configs/modules.conf in source dir |
| 11:29.59 | kaldemar | libryder: how is it a pita? |
| 11:29.59 | benngard | hmm, very strange, did send a call to the same extension from our avaya over ooh323 trunk that call had number pres... |
| 11:30.33 | libryder | kaldemar: by not understanding the asterisk dialplan syntax |
| 11:30.51 | puzzled | zealiod: use something like load => res_odbc.so |
| 11:31.02 | kaldemar | libryder: when you understand it, any layers in between become a pita. |
| 11:31.20 | libryder | kaldemar: what do you mean by layers? |
| 11:31.53 | kaldemar | libryder: abstraction layers like adhearsion or any other library that in the end just generates configs. |
| 11:32.27 | puzzled | jacc0: pastebin your dialplan |
| 11:33.45 | wdoekes2 | benngard: I don't know why it goes wrong. what does ${CALLERPRES()} say? |
| 11:35.18 | benngard | wdoekes2: sec |
| 11:35.24 | wdoekes2 | (and try setting sendrpid to no) |
| 11:36.59 | benngard | wdoekes2: hmm prohib_not_screened |
| 11:37.07 | zealiod | is it possible to see what SQL statement it's chucking at the ODBC? It's not authoring a user⦠and I'd like to do some debug⦠i'm currently on -vx8 |
| 11:37.29 | wdoekes2 | ok.. well that does explain why it doesn't get sent |
| 11:37.40 | *** join/#asterisk sgimeno (~kvirc@163.117.206.10) |
| 11:37.46 | benngard | must be telenor that sends it to me :( |
| 11:38.28 | wdoekes2 | no.. it does Privacy: none, so it should be allowed |
| 11:38.39 | wdoekes2 | do you have trustrpid=pai in Telenor sip config? |
| 11:39.52 | benngard | i have trustrpid=yes should it be pai? |
| 11:40.07 | jacc0 | @puzzled: I could do that, but it is a bit of a complicated dialplan; I'll make a simplified one |
| 11:40.26 | *** join/#asterisk freeedrich| (friedrich@perplexa.be) |
| 11:41.20 | benngard | trustpid=pai did the trick thx all! |
| 11:42.25 | wdoekes2 | hm.. seems like a bug.. or at least an oddity: no rpid is supplied, so imho it shouldn't assume that privacy=on |
| 11:45.00 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
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| 11:50.21 | tuxx- | hai guise. Where can i find the database scheme for realtime for asterisk 1.8? :) |
| 11:51.42 | tuxx- | ah, in the source: contrib/realtime/mysql/ apparantly :-) |
| 11:52.00 | *** part/#asterisk libryder (~dryder@host-74-211-9-139.beyondbb.com) |
| 12:00.15 | *** join/#asterisk jacc0 (~jacc0@D522448D.static.ziggozakelijk.nl) |
| 12:06.58 | tuxx- | hm, there is no queue_members.sql. anyone got an idea where i can get that scheme for 1.8? |
| 12:09.50 | saxa | <PROTECTED> |
| 12:09.55 | saxa | what that means ? |
| 12:10.28 | saxa | let me paste the full line |
| 12:10.38 | saxa | [Oct 3 13:27:48] WARNING[6103]: chan_iax2.c:10666 socket_process: Call rejected by 187.127.184.112: No authority found |
| 12:12.16 | dym | looks like credential error |
| 12:12.28 | dym | is this a transition between 2 servers? |
| 12:12.28 | kaldemar | tuxx-: takt a look at contrib/realtime/ in the source package. |
| 12:12.52 | kaldemar | saxa: it means that the call doesn't match a peer. |
| 12:13.44 | *** join/#asterisk timahvo1 (~rogue@41.215.1.35) |
| 12:15.11 | *** join/#asterisk mirelab (~mirelab@212.200.146.253) |
| 12:15.57 | *** part/#asterisk mirelab (~mirelab@212.200.146.253) |
| 12:16.14 | *** join/#asterisk bchia (~Adium@nat/digium/x-xbwtygcsxdmnogbo) |
| 12:16.22 | tuxx- | kaldemar: i did, but there is no queue_member.sql or queues.sql file :P |
| 12:16.29 | tuxx- | root@voip02:~/asterisk-1.8.7.0/contrib/realtime# ls mysql/ |
| 12:16.29 | tuxx- | iaxfriends.sql meetme.sql musiconhold.sql queue_log.sql sipfriends.sql voicemail_data.sql voicemail_messages.sql voicemail.sql |
| 12:16.33 | tuxx- | ;-) |
| 12:16.40 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 12:16.45 | wdoekes2 | check app_queue.c ;) |
| 12:17.12 | wdoekes2 | you may have to read past some code here and there |
| 12:17.20 | tuxx- | mkay |
| 12:18.00 | tuxx- | oh, there is a scheme for postgresql. Could just port that to mysql.. :) |
| 12:21.55 | kaldemar | tuxx-: postgresql/realtime.sql |
| 12:22.01 | leifmadsen | that sql files are terribly out of date anyways |
| 12:22.15 | leifmadsen | you'll have to check code likely anyways to get the "real" table layouts |
| 12:22.18 | irroot | ola folks |
| 12:22.44 | kaldemar | those go as a base, then just fix according to the errors. |
| 12:22.49 | devil_evoxxx | ola irroot |
| 12:22.52 | devil_evoxxx | :) |
| 12:23.17 | irroot | been lurkin abducted by the busy |
| 12:23.59 | zealiod | After doing minimal config changes (just ODBC) - I now restart asterisk and it won't start - gives 4 errors, this line being an example of one: WARNING[906]: pbx.c:2981 ast_register_application: Already have an application 'VoiceMail' |
| 12:24.22 | zealiod | I have removed the ODBC load statements in modules.conf I put it - but I still have the same issue - what's happened? |
| 12:24.43 | zealiod | any help appreicated |
| 12:25.47 | jacc0 | this is my dialplan (simplified) : http://pastebin.com/PyWF8Ks3 |
| 12:26.10 | jacc0 | features.conf holds: switch => 3,callee,macro,switch |
| 12:26.29 | jacc0 | I have client A calling in; then multiple sip clients(B) are called. The first one to pick up and press dtmf 2 will be bridged with client A |
| 12:26.42 | jacc0 | in the dialplan of client B I've added a feature for dtmf 3 ; when pressed it bridges with some other (already called client (C)) |
| 12:26.56 | jacc0 | he strange thing is that client c always gets connected with client A . I would expact it to be connected to client B (the operator) |
| 12:27.04 | irroot | zealiod shut it down properly may have to kill it |
| 12:27.34 | zealiod | irroot: yeah, already did |
| 12:27.55 | jacc0 | setting option : callee,celler or peer in featers.conf makes no difference |
| 12:28.00 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 12:28.49 | jacc0 | how can I make sure the Bridge() that is executed by the feature is executed in the right call-leg? |
| 12:29.27 | tuxx- | thanks kaldemar / leifmadsen / wdoekes2 :) |
| 12:29.36 | jacc0 | or is this a bug (as I expect)? |
| 12:30.13 | [TK]D-Fender | jacc0, You need to set the APPLICATIONMAP channel variable to allow a call access to dynamic features you define. For debugging it would also help if you included your features.conf in a pastebin. |
| 12:32.03 | jacc0 | everything is working; I have the applicationmap in my features; the only strange thing is that it gets bridged with the wrong call-leg |
| 12:32.29 | *** part/#asterisk benngard (~mabe@213.88.138.230) |
| 12:32.34 | jacc0 | but if you realy want to see my features.conf: http://pastebin.com/SYfuv04v |
| 12:32.39 | *** join/#asterisk mirelab (~mirelab@212.200.146.253) |
| 12:32.46 | puzzled | jacc0: but do you set the applicationmap in your dialplan? see features.conf for an example of what [TK]D-Fender referred to: Set(DYNAMIC_FEATURES=myfeature1#myfeature2#myfeature3) |
| 12:32.50 | *** part/#asterisk mirelab (~mirelab@212.200.146.253) |
| 12:33.01 | *** join/#asterisk Quintana (~sylvain@aghnar.doowan.net) |
| 12:34.19 | jacc0 | it is there: 17.exten = _X.,n(connect),Set(DYNAMIC_FEATURES=switch) |
| 12:34.32 | jacc0 | see the pastebin |
| 12:34.53 | jacc0 | it is not that the feature isn't working; it's just bridging to the wrong call leg |
| 12:35.23 | jacc0 | and this is a simplified version of my dialplan; it could hold some typ0's; please don't point them out to me |
| 12:35.31 | [TK]D-Fender | jacc0, describe the call-flow. Where does this start? |
| 12:35.32 | jacc0 | because that is not the question |
| 12:35.40 | puzzled | like incomming instead of incoming :) |
| 12:35.59 | jacc0 | yes, typ0's like that |
| 12:36.15 | [TK]D-Fender | "typo" |
| 12:36.17 | jacc0 | I have client A calling in[incomming]; then multiple sip clients(B) are called. The first one to pick up and press dtmf 2 will be bridged with client A |
| 12:36.43 | jacc0 | client B enters in the [operator] context |
| 12:36.59 | jacc0 | so when he presses dtmf 2 he gets bridged with the caller (client A) |
| 12:37.49 | jacc0 | then when the operator presses dtmf 3 , the operator should be connected with client C (the listening post) |
| 12:38.03 | [TK]D-Fender | jacc0, How many people get called at a time with this? |
| 12:38.18 | jacc0 | like 3 operator |
| 12:38.30 | [TK]D-Fender | jacc0, then this was entirely unnecessary. |
| 12:38.42 | jacc0 | why? |
| 12:38.42 | [TK]D-Fender | jacc0, just Dial() them all at once witha single confirm macro |
| 12:38.49 | jacc0 | no |
| 12:39.10 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 12:39.10 | [TK]D-Fender | <jacc0> I have client A calling in[incomming]; then multiple sip clients(B) are called. The first one to pick up and press dtmf 2 will be bridged with client A <- dial +M does this already. |
| 12:39.21 | jacc0 | because when one picks up the line others stop ringing (so when a voicemail answers the line nobody is called) |
| 12:40.03 | [TK]D-Fender | jacc0, it is not "answered" until actually answered. Dial them as local channels if you have to. |
| 12:40.12 | [TK]D-Fender | jacc0, Still < 1/3 the complexity |
| 12:40.15 | jacc0 | I want alle the operators phones to keep ringing until one of the operators answers the line and presses dtmf 2 |
| 12:41.11 | jacc0 | but that is not my question; because this is a very simplified version of my dialplan |
| 12:41.38 | jacc0 | operators can also recieve an text message or some thing called an interactive text message |
| 12:41.51 | jacc0 | but this all has nothin to do with my question |
| 12:42.31 | jacc0 | why is client A bridged with client C if client B(the operator) presses (the feature) dtmf 3? |
| 12:42.31 | [TK]D-Fender | jacc0, Dial multiple local channels. Have each of those do the actual dial with M. This should work leaving the others ringing until the macro actually accepts |
| 12:43.08 | jacc0 | okay, that might be true in this simple dial plan |
| 12:43.32 | jacc0 | but not in my more complex version; it als sends out text , message en some thing called interactive messages |
| 12:43.42 | jacc0 | s/als/also |
| 12:44.47 | jacc0 | so some of the operators get a text message and call in end then get connected with the incomming client A |
| 12:45.09 | [TK]D-Fender | jacc0, exten = _X.,n,macro(startCallingOperators,${CallID}) <- don't see this macro. |
| 12:45.52 | jacc0 | true, it originates several calles, sends out text messages, interactive message en does some more database querys |
| 12:46.29 | jacc0 | it would make the dialplan less understandable and would result in even more answers to questinos i've never asked |
| 12:48.13 | jacc0 | my question is : why is client A connected to client C when client B excutes the features that only holds : bridge(${channelidOfClientC}) |
| 12:48.31 | jacc0 | I would expect client B to be connected to client C |
| 12:49.50 | jacc0 | and how can I change this? |
| 12:51.03 | [TK]D-Fender | jacc0, we don't get to see where ${CallID} really comes from, I don't see how you pass any information on to those operators you're calling, or seeing how you're doing it at all... can;t really help when you have not shown the key segments of this |
| 12:51.34 | jacc0 | forget the hole dialplan |
| 12:51.41 | jacc0 | <PROTECTED> |
| 12:52.36 | jacc0 | CallID is yous a simple autoincrement value from the database) |
| 12:53.11 | jacc0 | that is returned bij macro(createNewCall) in the first row of the dialplan |
| 12:53.39 | [TK]D-Fender | Which is code we don't see |
| 12:54.56 | jacc0 | which code you do not need |
| 12:55.32 | jacc0 | but I will give it to you if you want; exten = _X.,1,Set(CallID=1) |
| 12:55.34 | jacc0 | there it is |
| 12:55.37 | [TK]D-Fender | By default I don't trust that CallID you are setting and that also happens to be something critical to the mapping and the one thing you are refusing to show us. |
| 12:55.51 | *** join/#asterisk nighty^ (~nighty@TOROON12-1279662182.sdsl.bell.ca) |
| 12:56.30 | [TK]D-Fender | jacc0, that is not "auto-incrementing" or your actual code. You are wasting my time with this sad diversion |
| 12:56.46 | jacc0 | ty |
| 12:56.48 | [TK]D-Fender | moves on to more productive matters |
| 12:57.04 | jacc0 | <PROTECTED> |
| 12:57.30 | jacc0 | and how can I change this? |
| 12:58.23 | jacc0 | I would expect the bridge to be executed on the call leg of client B and would bridge client B with client C |
| 12:58.32 | jacc0 | ut it isn't |
| 12:58.35 | jacc0 | *but |
| 12:58.45 | jacc0 | would that be a bug? |
| 13:00.03 | kaldemar | jacc0: you're the only one capable of judging that since no one else sees what your dialplan does. |
| 13:00.31 | jacc0 | it is realy no use to post the hole thing |
| 13:01.08 | jacc0 | you would also need the entire database, alle the external application including source code |
| 13:01.45 | jacc0 | it would take you days or weeks to get it even running on your local machine ; and would take you months to see the hole picture |
| 13:02.21 | jacc0 | so forget about the dialplan |
| 13:03.13 | jacc0 | can someone please answer this: why is client A, that is talking to client B , connected to client C when client B excutes the features that only holds : bridge(${channelidOfClientC})? anyone? |
| 13:03.54 | jacc0 | I would expect the bridge to me executed on the call leg of client b |
| 13:04.08 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 13:04.46 | jacc0 | and so I would expect client B to be connected to client C |
| 13:05.21 | *** join/#asterisk serafie (~erin@nat/digium/x-kzerfvvjexdhhyem) |
| 13:05.45 | jacc0 | and the strange thing is if I let client A execute the feature, client A still gets connected to client C |
| 13:11.46 | jacc0 | s/me/be |
| 13:18.26 | jacc0 | so in other words: client A is talking to client B. client B executes a feature that only holds bridge(${channelIDOfClientC}). to what client should client C be bridged? and why? |
| 13:20.41 | Katty | drags in |
| 13:21.12 | [TK]D-Fender | Katty, Mew. |
| 13:21.17 | bchia | jacc0: I see you dilemma - it seems to me that your call should absolutely be bridging B & C and not channel A |
| 13:21.35 | bchia | My only answer is to say that "Macro does not work as expected with the application map" |
| 13:21.41 | bchia | see the sample.features.conf |
| 13:21.50 | bchia | ; IMPORTANT NOTE: The applicationmap is not intended to be used for all Asterisk |
| 13:21.50 | bchia | ; applications. When applications are used in extensions.conf, they are executed |
| 13:21.50 | bchia | ; by the PBX core. In this case, these applications are executed outside of the |
| 13:21.50 | bchia | ; PBX core, so it does *not* make sense to use any application which has any |
| 13:21.50 | bchia | ; concept of dialplan flow. Examples of this would be things like Macro, Goto, |
| 13:21.51 | bchia | ; Background, WaitExten, and many more. |
| 13:21.55 | Katty | hi fender bender. |
| 13:22.00 | bchia | sorry :( |
| 13:22.30 | [TK]D-Fender | bchia, Yes, pastebin is your friend. |
| 13:22.44 | jacc0 | ty, bchia |
| 13:22.59 | jacc0 | I'll try to avoid using macro |
| 13:23.09 | jacc0 | and let you know the result |
| 13:25.20 | zealiod | OK, all sorted - got the DSN working⦠is it normal in -v output to see 100s of NOTICE' like these on the screen> |
| 13:25.21 | zealiod | NOTICE[1614]: chan_sip.c:12759 handle_response_peerpoke: Peer '205' is now Reachable. |
| 13:25.33 | zealiod | like 20 of these a second? |
| 13:27.44 | [TK]D-Fender | zealiod, sounds like your peers regularly timeout on you. This would be due to network conditions or heavy system load |
| 13:28.16 | zealiod | [TK]D-Fender: I just have one soft phone installed on one extension for testing |
| 13:28.29 | zealiod | [TK]D-Fender: It's hammering mySQL server also with queries |
| 13:28.38 | zealiod | Got to be something wrong... |
| 13:29.26 | [TK]D-Fender | zealiod, Actually I suppose that if mySQL is holding your peers then if * stalls on a DB loopup to match a qualify packet then that added delay could compound in a "failure" |
| 13:29.43 | [TK]D-Fender | zealiod, This could be purey on mySQL's side |
| 13:29.48 | [TK]D-Fender | purely* |
| 13:31.20 | zealiod | [TK]D-Fender: I don't follow - can this be fixed. I have a very simple setup. No load on the mySQL box, no load on the asterisk box⦠just a simple proof of concept setup |
| 13:31.47 | zealiod | this sounds similar - back in July 08 http://lists.digium.com/pipermail/asterisk-bugs/2008-July/020335.html |
| 13:31.50 | [TK]D-Fender | the MySQL is not on the same server? |
| 13:31.59 | zealiod | [TK]D-Fender: nope |
| 13:32.17 | leifmadsen | zealiod: try changing your qualify= value to somethign larger |
| 13:32.21 | [TK]D-Fender | zealiod, you could pad out your qualify timeout to give a largr grace period. |
| 13:32.28 | [TK]D-Fender | qualify=5000 |
| 13:32.32 | leifmadsen | if not, check the SIP traffic and see if the registration is timing out or something else failing to be responded to |
| 13:32.35 | zealiod | where is that value kept? |
| 13:32.36 | [TK]D-Fender | (yes = 2000) |
| 13:32.42 | [TK]D-Fender | zealiod, in the SIP peer |
| 13:32.46 | zealiod | ok |
| 13:32.46 | leifmadsen | zealiod: in the peer configuration in sip.conf |
| 13:33.41 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:33.41 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:38.59 | zealiod | currently it is set to yes - what does 5000 mean? |
| 13:40.32 | zealiod | I still get the same issue |
| 13:40.36 | zealiod | here's a small paste |
| 13:40.38 | zealiod | [Oct 3 13:40:06] NOTICE[1686]: chan_sip.c:12759 handle_response_peerpoke: Peer '205' is now Reachable. (59ms / 500ms) |
| 13:40.38 | zealiod | [Oct 3 13:40:06] NOTICE[1686]: chan_sip.c:12759 handle_response_peerpoke: Peer '205' is now Reachable. (52ms / 500ms) |
| 13:40.51 | dym | could have nopasted :) |
| 13:40.59 | zealiod | sorry :/ |
| 13:41.44 | [TK]D-Fender | <PROTECTED> |
| 13:41.50 | [TK]D-Fender | Looks like you did 500, not 5000 |
| 13:45.02 | zealiod | [TK]D-Fender: I have tried with both 500 and 5000 - still getting flooded |
| 13:45.20 | zealiod | 20 NOTICES per second |
| 13:45.34 | *** join/#asterisk Azrael808 (~peter@212.161.9.162) |
| 13:45.55 | [TK]D-Fender | Aer you indeed using a DB for your SIP peers? |
| 13:46.09 | zealiod | yup |
| 13:47.36 | [TK]D-Fender | Check that you have write permissions back as IIRC * may write some of that info back |
| 13:47.51 | zealiod | can you please explain |
| 13:47.58 | [TK]D-Fender | if it staleld a write I would posibly fail to reset a counter on that. |
| 13:48.15 | [TK]D-Fender | check that *'s user has read/write on those tabels |
| 13:48.19 | [TK]D-Fender | tables* |
| 13:48.25 | zealiod | ok |
| 13:48.34 | zealiod | ok the mySQL tables right? |
| 13:48.40 | [TK]D-Fender | yes |
| 13:48.43 | zealiod | ok |
| 13:49.15 | zealiod | yes - it has SELECT, INSERT, UPDATE, DELETE |
| 13:49.26 | zealiod | on the database |
| 13:50.04 | *** join/#asterisk lcat (~lcat@187.45.254.55) |
| 13:50.07 | [TK]D-Fender | Ok, I'm a little at a loss from here. Do erstart * though to ensure changes have taken |
| 13:50.19 | zealiod | ok |
| 13:51.04 | zealiod | nope - the same |
| 13:53.55 | zealiod | This seems to be the same issue |
| 13:53.57 | zealiod | https://issues.asterisk.org/view.php?id=12508 |
| 13:54.14 | zealiod | I don't understand the terms used- does it look like the issue can be resolved? |
| 13:56.01 | *** join/#asterisk coppice (~chatzilla@116.92.17.112) |
| 13:57.51 | [TK]D-Fender | zealiod, that is a very old report which was resolved.... |
| 13:58.00 | zealiod | OK |
| 13:58.08 | zealiod | [TK]D-Fender: It just read very similar |
| 13:58.27 | zealiod | [TK]D-Fender: What are my options now? |
| 13:58.35 | [TK]D-Fender | zealiod, Yes part of the symptoms do match... |
| 13:59.10 | [TK]D-Fender | zealiod, Suppose you could try to open up another issue ont he tracker for your case, provide all the machine specs, versions, etc and see who bites... |
| 13:59.21 | [TK]D-Fender | zealiod, probably your best odds |
| 14:04.24 | zealiod | I was just looking at the support plans too... |
| 14:05.04 | [TK]D-Fender | zealiod, as this is a "test", what does your full-scale deployment look like? |
| 14:05.20 | zealiod | Just a simple VOIP setup |
| 14:05.26 | zealiod | one trunk in, one out |
| 14:05.32 | zealiod | and a bunch of extensions |
| 14:05.47 | zealiod | just for 20-30 users |
| 14:06.00 | zealiod | Seems silly to pay $595 to get it working |
| 14:06.27 | zealiod | I'd hoped the open source route - with a bit of persistence would work⦠its seeming not |
| 14:06.52 | *** join/#asterisk master_of_master (~master_of@p57B549F8.dip.t-dialin.net) |
| 14:08.38 | [TK]D-Fender | zealiod, Nah, overkill |
| 14:08.52 | *** join/#asterisk d1j0n (~Adium@75-150-66-252-NewEngland.hfc.comcastbusiness.net) |
| 14:09.25 | [TK]D-Fender | zealiod, First, I'd remove the DB factor from this. Start with the normal flat configs. You need to remove points of failure, and DB doesn't sounds at all necessary here. |
| 14:09.49 | zealiod | [TK]D-Fender: It works with out the DB |
| 14:09.57 | [TK]D-Fender | zealiod, And you have a very singular road-bump here. Don't get discouraged over this. |
| 14:10.15 | [TK]D-Fender | zealiod, Could be a limited bug or freak scenario with your setup. |
| 14:10.41 | zealiod | But - I'd like to manage users through mySQL - one of the seasons I got excited enough to try this out. |
| 14:10.43 | *** join/#asterisk file (~file@asterisk/developer-and-muffin-lover/file) |
| 14:10.43 | *** mode/#asterisk [+o file] by ChanServ |
| 14:11.43 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 14:11.43 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 14:11.46 | [TK]D-Fender | zealiod, What ver aer you on? |
| 14:14.29 | zealiod | OK, I got it! |
| 14:14.39 | [TK]D-Fender | zealiod, Ok, what did you find? |
| 14:14.54 | zealiod | [TK]D-Fender: "Asterisk 1.4.21.2 (on Debian) does not work if qualify is set and rtcachefriends=yes (which is default). |
| 14:14.54 | zealiod | qualify must be NULL or should not be in the table at all. Otherwise this error message will flood your log file/console:" |
| 14:15.14 | [TK]D-Fender | zealiod, that is disgustingly old and should not be run :) |
| 14:15.20 | zealiod | As it happens - I don't have a field rtcachefriends in my peer table - but if I set qualify to NULL - all good. |
| 14:15.40 | zealiod | [TK]D-Fender: heh |
| 14:15.56 | zealiod | [TK]D-Fender: OK, but it shows something interesting... |
| 14:16.20 | zealiod | [TK]D-Fender: Is qualify all that important> |
| 14:16.24 | [TK]D-Fender | zealiod, large number of security fixes since then and 4 branches in full release since then with another on the way |
| 14:16.55 | [TK]D-Fender | zealiod, it is generally necessary as a NAT keep-alive. You can usually live without it though. |
| 14:17.05 | [TK]D-Fender | (in all other cases) |
| 14:17.13 | zealiod | [TK]D-Fender: Well it might help to have an up to date schema⦠any ideas where I can grab one |
| 14:17.16 | [TK]D-Fender | zealiod, it is also a sanity check./ |
| 14:17.36 | [TK]D-Fender | zealiod, You're asking for "up to date" ... about 1.4? :p |
| 14:18.10 | [TK]D-Fender | zealiod, BTW, they;re up to 1.4.42. You are only 1/2 way through that brach at all :) |
| 14:18.10 | zealiod | Asterisk 1.4.21.2~dfsg-3+lenny5 |
| 14:18.31 | zealiod | [TK]D-Fender: I just got the pre-compiles |
| 14:18.43 | *** join/#asterisk endemic (~endemic@2001:49f0:400e::5) |
| 14:19.20 | [TK]D-Fender | zealiod, I highly recommend you not use those and use another repo, or compile your own... |
| 14:26.28 | *** join/#asterisk SwK (~SwK@freeswitch/developer/swk) |
| 14:26.58 | jacc0 | okay, I figured it out : switch => 3,self,macro,switch |
| 14:27.15 | jacc0 | now it's bridging with the right channel |
| 14:32.19 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
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| 15:04.51 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 15:10.04 | zealiod | Can any one help me understand this error - I think I have this configured in my dial plans⦠so I am unsure... |
| 15:10.12 | zealiod | Call from 'XXX' to extension 'XXX' rejected because extension not found. |
| 15:10.15 | zealiod | Both XXX are the same |
| 15:11.24 | *** join/#asterisk SwK (~SwK@freeswitch/developer/swk) |
| 15:12.18 | *** join/#asterisk singler (~singler@78-60-139-121.static.zebra.lt) |
| 15:12.42 | navaismo | the extension XXX doesnt exist in the context of XXX |
| 15:13.43 | p3nguin | Seems clear to me. |
| 15:15.38 | *** join/#asterisk gabeh (~gabeh@67.133.93.182) |
| 15:18.59 | gabeh | Hello, total phone system newbie here. I have tried two phone systems, (Asterisk 1.8.6, FreePBX 2.9, Grandstream GX4104 Gateway) and also (3CX v10 SP3, Sangoma A200). For both systems, only when someone on a Verizon cellphone calls, there is this strange ringback tone, it seriously sounds like darth vader in between rings. So its like RING, darth vader noise, RING, darth vader noiseâ¦etc. Is anyone aware of what I'm doing wrong to make verizon phones sou |
| 15:19.00 | gabeh | weird? |
| 15:21.13 | oldhack | Luke, I'm your father! |
| 15:22.04 | gabeh | yeah exactly! |
| 15:22.11 | _Corey_ | gabeh: Maybe you need to embrace the dark side... |
| 15:22.11 | Faustov | gabeh: perhaps you have a mistake in indications.conf? |
| 15:22.25 | Faustov | have you specified your location? |
| 15:22.58 | gabeh | sorry, going slow now, what would I be looking for in order to check that? |
| 15:24.18 | [TK]D-Fender | zealiod, Your peer XXX dialed XXX and there was no match in whatever context was being searched based on the peer/general context (as applied) |
| 15:25.10 | zealiod | [TK]D-Fender: but I have set up default contexts in the extensions table and have this line in the .conf file... |
| 15:25.10 | zealiod | [default] |
| 15:25.10 | zealiod | switch => Realtime/default@extentions |
| 15:25.36 | zealiod | That should be good to go? I've migrated it from the .conf file... |
| 15:26.12 | Faustov | gabeh: all clues are in that config file |
| 15:26.16 | gabeh | Faustov: Do you have any idea why this would only happen on Verizon Wireless phones? |
| 15:26.30 | [TK]D-Fender | zealiod, verify what context your devices are even pointed to, pastebin your conf, and dump the MySQL |
| 15:26.38 | Faustov | gabeh: they might use different indication methods, but not sure exactly why |
| 15:29.57 | gabeh | Faustov: I didn't touch that file. And the strange thing is it happened when I was running 3CX as well |
| 15:30.36 | zealiod | [TK]D-Fender: is there an easy way to dump all the config - I have the SQLDump ready.... |
| 15:30.48 | [TK]D-Fender | copy & paste stuff... |
| 15:30.53 | [TK]D-Fender | Not hard |
| 15:31.37 | p3nguin | Just don't paste it here. |
| 15:33.39 | zealiod | [TK]D-Fender: OK, if you need anything else - just say, .conf files under SQL dump |
| 15:33.40 | zealiod | http://pastebin.com/jsVFCGAG |
| 15:34.31 | [TK]D-Fender | zealiod, and pastebin the failed call at * CLI with sip debug enabled |
| 15:38.09 | [TK]D-Fender | zealiod, do not mask anything from that |
| 15:38.16 | [TK]D-Fender | Passwords only. |
| 15:38.44 | tuxx- | eughhh, sucks i cant attend astricon this year ;-( |
| 15:39.36 | zealiod | [TK]D-Fender: OK |
| 15:40.41 | zealiod | [TK]D-Fender: Is that what you needed? |
| 15:40.49 | [TK]D-Fender | zealiod, pastebin "dialplan show" from * CLI. I'm suspecting it isn't hitting your DB for this... |
| 15:42.04 | zealiod | Thanks |
| 15:43.01 | zealiod | Context 'default' created by 'pbx_config' ] - look spretty empty |
| 15:43.02 | [TK]D-Fender | zealiod, INSERT INTO `extensions` (`id`, `context`, `exten`, `priority`, `app`, `appdata`) VALUES |
| 15:43.02 | zealiod | Hmm |
| 15:43.18 | [TK]D-Fender | zealiod, Alt. Switch => 'Realtime/default@extentions' [pbx_config] |
| 15:43.25 | [TK]D-Fender | zealiod, extensions != extentions |
| 15:43.35 | zealiod | ahh f* |
| 15:45.09 | zealiod | [TK]D-Fender: strangely - all now working :) |
| 15:45.12 | zealiod | sorry :( |
| 15:45.23 | [TK]D-Fender | zealiod, yes, very "strange" ;) |
| 15:56.16 | zealiod | [TK]D-Fender: Is there a defined up-to-date schema for MySQL? |
| 15:56.25 | zealiod | I just want to make sure I have all the right fields, etc. |
| 15:57.14 | *** join/#asterisk SuperNull (~FreeManof@24-148-101-238.ip.mhcable.com) |
| 15:57.31 | SuperNull | Hey All. |
| 16:00.21 | *** join/#asterisk engrxyz (~fgdfgfdg@212.23.51.7) |
| 16:01.32 | SuperNull | I am having a problem getting regserver to populate on asterisk 1.8 using the mysql driver for asterisk realtime. I have a pastebin of the table syntax as is currently. http://pastebin.com/dj9m9Uju |
| 16:01.44 | SuperNull | Looking to see if anyone has insight why it is not updating regserver. |
| 16:02.16 | [TK]D-Fender | ? book |
| 16:02.21 | [TK]D-Fender | ~book |
| 16:02.21 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/ or ~buybook |
| 16:02.36 | [TK]D-Fender | and there is the old one which was more dated for you : |
| 16:03.06 | SuperNull | [TK]D-Fender is for me? |
| 16:03.11 | [TK]D-Fender | SuperNull, No. |
| 16:05.11 | *** join/#asterisk ruied_ (~ruied@19.198.108.93.rev.vodafone.pt) |
| 16:06.45 | ruied_ | Hi! Can I send BLF status manualy to a phone? |
| 16:08.09 | *** join/#asterisk aberrios (~aberrios@195.171.4.82) |
| 16:09.10 | *** join/#asterisk Dovid (Dovid@office.mypbxmanager.net) |
| 16:09.37 | *** join/#asterisk singler (~singler@78-60-139-121.static.zebra.lt) |
| 16:15.48 | Katty | ohai |
| 16:16.55 | n3hxs | HiO Katty |
| 16:20.02 | carrar | hi KattyRooooo |
| 16:21.12 | Dovid | hi all. i think i have a "possible" bug |
| 16:22.12 | Dovid | when using set(SIP_CODEC=g722) the peer I calling does not support G722. in such a case why doesn't asterisk trancode the call ? |
| 16:22.36 | *** join/#asterisk hobodave (~hobodave@pdpc/supporter/professional/hobodave) |
| 16:23.27 | p3nguin | Because you are trying to force asterisk to use a codec that the peer does not support. |
| 16:23.48 | p3nguin | If no codec can be agreed on, the call will die. |
| 16:24.15 | Dovid | p3nguin: It ends up using G729. |
| 16:24.30 | Dovid | through and through. my question why wont asterisk transcode the call? |
| 16:24.36 | Dovid | because I allow G729 as well ? |
| 16:24.51 | [TK]D-Fender | ruied_, describe your full goal |
| 16:25.13 | p3nguin | I'd imagine asterisk will transcode between two peers with dissimilar codecs. |
| 16:25.15 | [TK]D-Fender | Dovid, What are you running? |
| 16:25.44 | Dovid | 1.8.6.0 |
| 16:26.21 | [TK]D-Fender | Dovid, I don't believe that necessarly changes the codec of the caller.... |
| 16:26.33 | Dovid | p3nguin: My issue actually is that I want to enable G722 but don't want to trandscode. I want G722 to work only if it will not transcode, it works like i want it to but want to make sure i wont end up transcoding all calls |
| 16:27.17 | Dovid | [TK]D-Fender: What does not change it ? set(SIP_CODEC=G72) ? iknow with G729 and Alaw it does |
| 16:27.23 | Dovid | G722* |
| 16:27.38 | [TK]D-Fender | Callee, not caller |
| 16:27.42 | *** join/#asterisk yoyolala (~yoyolala@unaffiliated/yoyolala) |
| 16:27.44 | p3nguin | If you have codec translators available, transcoding will happen for calls between peers with dissimilar codecs. |
| 16:29.13 | Dovid | [TK]D-Fender: Correct. It's for the calleee however when i send the call out to the carrier who does not support G722 Asterisk shows "[Oct 3 12:19:10] NOTICE[23978]: chan_sip.c:6304 try_suggested_sip_codec: Ignoring ${SIP_CODEC} variable because it is not shared by both ends." |
| 16:30.33 | Dovid | p3nguin: SO Asterisk will only transcode when the codec set up is the same ? or is the reason that it uses G729 is because both support it and both do not support G722. before I enable it I just want to make sure I wont be doing a lot of transcoding |
| 16:30.35 | [TK]D-Fender | Well, you don't want to transcode, and it isn't supported by both ends.... |
| 16:30.40 | [TK]D-Fender | Sounds like exactly what you asked for |
| 16:31.32 | Dovid | [TK]D-Fender: 100% what I want. I just want to make sure that it wont trancode unless it's G722 on both ends. the only way I see it transcoding is if the phone only support G722 and the other end only supports any other codec. is that correct ? |
| 16:32.06 | leifmadsen | why would asterisk transcode if both ends support the codec? |
| 16:32.11 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 16:32.15 | leifmadsen | I think you're confusing what "transcoding" means |
| 16:32.19 | [TK]D-Fender | Dovid, that message seems to imply this is forced behavior. It won't accept unless matched |
| 16:32.35 | leifmadsen | disregard |
| 16:32.37 | leifmadsen | I misread |
| 16:32.42 | leifmadsen | swears at mondays |
| 16:33.03 | *** join/#asterisk dirkD (~dirk@84-245-20-6.dsl.cambrium.nl) |
| 16:33.45 | Dovid | leifmadsen: OK so it would seem that is the case because i am forcing. but just enabled it might act differently |
| 16:33.50 | p3nguin | I'll say it again: transcoding will happen for calls between peers with dissimilar codecs. |
| 16:34.00 | leifmadsen | ^^^ that |
| 16:34.23 | Dovid | p3nguin: Ok. So Asterisk will only transcode where phoneA and PhoneB have NO similar codecs |
| 16:34.31 | leifmadsen | you can see in the setup when you enable sip debug the codecs that are offered between both sides and what is similar |
| 16:34.36 | p3nguin | No. That's not what I said. |
| 16:34.55 | p3nguin | Asterisk will transcode calls between peers which are USING codecs that are different. |
| 16:35.09 | leifmadsen | an end point can offer multiple cdecs |
| 16:35.11 | leifmadsen | codecs* |
| 16:35.21 | p3nguin | If you call from phone1 using g729 to phone2 which is using g722, asterisk has to transcode it. |
| 16:35.37 | Dovid | p3nguin: So if PhoneA has G722, G729, G711A and Phone2 is G729, G711A, G711U it will trancode ? |
| 16:35.46 | p3nguin | They can support whatever they want, but if the codecs USED are not the same, transcoding must happen. |
| 16:35.54 | leifmadsen | Dovid: no |
| 16:36.06 | leifmadsen | it will use G711A |
| 16:36.09 | Dovid | p3nguin: Thats a given. but when the invite has multiple codecs like what I said aboce |
| 16:36.29 | Dovid | leifmadsen: Thanks. let me get the exact example from the invite |
| 16:36.37 | p3nguin | There are similar codecs available there, but if one has used a codec that the other has not used, there will be transcoding. |
| 16:36.41 | leifmadsen | Dovid: I don't really care about the exact example :) |
| 16:36.52 | leifmadsen | just look at the sip debug |
| 16:36.55 | leifmadsen | it'll tell you exactly what is happening |
| 16:37.09 | leifmadsen | just test for the scenarios you need to handle and determine if it is doing what you need |
| 16:37.15 | p3nguin | If your phone negotiates g722, and then asterisk negotiates g729 to the other phone, there will be transcoding. |
| 16:37.15 | leifmadsen | and now I go back to work |
| 16:37.16 | leifmadsen | c/lose |
| 16:37.18 | *** part/#asterisk leifmadsen (~leifmadse@asterisk/documenteur-extraordinaire/blitzrage) |
| 16:37.26 | Faustov | fial! |
| 16:38.01 | p3nguin | If your phone negotiates g711u, and then asterisk negotiates g711u with the other phone, there will be no transcoding. |
| 16:38.47 | Dovid | p3nguin: If it ONLY negotiates G722. Correct ? |
| 16:38.56 | p3nguin | No. |
| 16:39.32 | p3nguin | If your phone negotiates g729, and then asterisk negotiates g711u to the other phone, there will be transcoding. |
| 16:39.53 | p3nguin | Any time there is a dissimilar codec on either leg of a call, there will be transcoding. |
| 16:40.38 | p3nguin | There is only NO transcoding when both legs of the call are using the same codec. |
| 16:41.17 | Katty | hugs n3hxs and carrar |
| 16:41.24 | carrar | woohoo! |
| 16:42.40 | Dovid | p3nguin: My question is when there are multiple codecs in the invite. and when you say " Any time there is a dissimilar codec" You mean when they both dont have... ? |
| 16:42.46 | Dovid | sorry i am a bit slow today |
| 16:44.39 | SuperNull | OMG im gonna start pulling hair out (not my own likely) if i cannot figure out why regserver isn't filling. it recognized it |
| 16:45.19 | ruied_ | [TK]D-Fender, I would like to make a blf with the 999 extension to activate/deactivate Night mode in the dialplan, when night mode is activated, the blf key will turn red |
| 16:46.01 | irroot | exten => 666,1,System("shutdown -h now") |
| 16:46.51 | ruied_ | nightmode/daymode switching in the dialplan is already working, when I press it goes to night mode, if I press again it will back to day mode... the goal is to turn red key on when in night mode... |
| 16:47.46 | p3nguin | dovid: Just because both sides support a codec does not mean both sides will be using the same codec. |
| 16:48.10 | p3nguin | It does not matter if they both HAVE the codec... they must USE the same codec to avoid transcoding. |
| 16:48.51 | [TK]D-Fender | ruied_, "core show function DEVICE_STATE" |
| 16:49.18 | *** join/#asterisk Ineedabettername (~Ineedabet@pool-98-119-41-128.lsanca.fios.verizon.net) |
| 16:49.35 | ruied_ | [TK]D-Fender, I've managed this with the sendtext(), but in grandstream phone it goes to instant messages box, not displayng directly at the phone, so was triing to know if I can do this with the BLF... |
| 16:50.01 | ruied_ | [TK]D-Fender, going to check |
| 16:50.28 | *** join/#asterisk r0m|u (~are@darkstar.rice.edu) |
| 16:50.59 | r0m|u | p3nguin, you avail? |
| 16:51.06 | p3nguin | Maybe. |
| 16:51.16 | r0m|u | Its me SeRi |
| 16:51.22 | r0m|u | did you get the sim? |
| 16:51.25 | p3nguin | What's up? |
| 16:51.27 | p3nguin | Not yet. |
| 16:51.47 | p3nguin | I haven't checked today, though. |
| 16:51.55 | zealiod | Can any one please refer me to some simple extension rules using exten => XXXXXX,n,Background(enter-ext-of-person) - where an extension is listened for via DTMF and then connected, or if dud Goto top... |
| 16:52.19 | r0m|u | ok got it. |
| 16:52.43 | r0m|u | strange though. It should have been there by now :/ |
| 16:53.00 | carrar | zealiod |
| 16:53.02 | carrar | ~book |
| 16:53.02 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/ or ~buybook |
| 16:54.47 | p3nguin | zealiod: Create the extensions in the same context or in an included context. If you don't have enough time during the playing of the sound file in BackGround(), use WaitExten() afterward. |
| 16:58.02 | zealiod | p3nguin: Sure I got that far, just wanted to then use those digits to dial the extension... |
| 16:58.07 | zealiod | Just downloading the book now.... |
| 16:58.48 | p3nguin | You don't "use those digits" to dial... those digits dial. Do you understand the difference? |
| 16:59.17 | p3nguin | During the BackGround() and/or WaitExten(), if you enter in, for example, 123 on your keypad, extension 123 is executed. |
| 16:59.48 | p3nguin | Then extension 123 can do whatever you want, including Dial() one of your phones. |
| 17:00.06 | p3nguin | exten => 123,1,Dial(SIP/jane,30) |
| 17:00.23 | *** join/#asterisk Rroet (~Rroet@5354C380.cm-6-5d.dynamic.ziggo.nl) |
| 17:00.33 | Rroet | gooed evening all. |
| 17:00.41 | Rroet | <at least, in my time zone> |
| 17:01.20 | puzzled | evening Rroet |
| 17:01.53 | Rroet | my I ask a few Asterisk beginner questions? |
| 17:02.02 | *** join/#asterisk Sylnai (~andi@andimiller.net) |
| 17:02.22 | [TK]D-Fender | Rroet, Ask away |
| 17:03.30 | Rroet | I've gotten a default ubuntu (11.04) asterisk setup and added the asterisk gui to it. I've then setup 2 users and a voipbuster sip trunk. I can call between the 2 users (sip) perfectly. But I can't seem to route any calls to voipbuster and I'm stuck at this point. |
| 17:03.56 | Rroet | <besides the asterisk-gui making a huge mess of the .conf files> |
| 17:04.22 | p3nguin | We can't help you here with configuration via GUI. |
| 17:04.27 | p3nguin | But... |
| 17:04.39 | Rroet | I know my way around the cli as well ;) |
| 17:04.52 | p3nguin | You probably didn't configure any extension to dial out over the voipbuster peer. |
| 17:06.05 | Rroet | probably what they call "outgoing/incoming calling rules" in the gui? |
| 17:06.25 | p3nguin | (1204.22) <p3nguin> We can't help you here with configuration via GUI. |
| 17:07.10 | Rroet | I know, just relating back to what I started off with configuring. If the gui s*cks b*lls, I intend to not use it. |
| 17:07.28 | p3nguin | I'm not going to say it sucks, but we can't support it here. |
| 17:07.41 | Rroet | is there # that does support it? |
| 17:07.50 | p3nguin | #asterisk-gui maybe. |
| 17:08.01 | Rroet | too obvious d'oh! |
| 17:08.13 | Ineedabettername | any of you well practiced with iax trunking? |
| 17:08.21 | p3nguin | ~ask |
| 17:08.21 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 17:10.02 | Ineedabettername | provider wont allow traffic on default 4569...must use port >65000...have both sides set up to use 65001 in bindport and in my register line...in debug on both sides I see the REGREQ go out on 65001 but POKE's go out on 4569 |
| 17:10.36 | Ineedabettername | is there somewhere else that needs to get set? |
| 17:11.31 | *** join/#asterisk paulc (~paulc@unaffiliated/paulc) |
| 17:11.45 | irroot | Ineedabettername bassed on your discription im guessing the provider does not use IAX ?? as iax uses one port that cant be guessed ... SIP+RTP has this behavior where port can be random in any range |
| 17:12.38 | [TK]D-Fender | Rroet, the GUI is very incomplete and "barely maintained" |
| 17:13.05 | Rroet | so in other words, it's advised to be tossed out the window and make the cli your best friend? ;) |
| 17:13.23 | p3nguin | Yes. |
| 17:13.31 | Rroet | I like #asterisk already. |
| 17:14.25 | *** join/#asterisk evharten (~evharten@oracle.evertje.net) |
| 17:14.38 | Ineedabettername | irroot: i own one end of the trunk and had IAX working fine there prevously with another asterisk server...had to mgrate to a provider...they say they support iax but won't allow traffic on 4569 for whatever security reason |
| 17:14.58 | Katty | i just read the cli is your best friend. |
| 17:15.10 | Katty | but my brain didn't interpret properly |
| 17:15.12 | Katty | bwahehehehe |
| 17:15.14 | irroot | Ineedabettername ah ok then it should work sorry for confusion |
| 17:15.25 | irroot | Katty explain i dont get it :P |
| 17:15.33 | Katty | it does not need explaining! |
| 17:15.40 | Katty | RE: Gutter. |
| 17:16.25 | p3nguin | How about a T, Pat? |
| 17:16.47 | *** join/#asterisk MoreAllLess (~Justo@adsl-99-41-7-219.dsl.lsan03.sbcglobal.net) |
| 17:20.16 | *** part/#asterisk zealiod (~zealiod@94-30-97-220.xdsl.murphx.net) |
| 17:20.38 | *** join/#asterisk celord (~celord@201.198.102.2) |
| 17:21.07 | *** join/#asterisk singler (~singler@78-60-139-121.static.zebra.lt) |
| 17:21.53 | SuperNull | Alright so for anyone who needs to know.. how to get regserver to update, you must enable it in sip.conf [general] rtsavesysname=yes |
| 17:22.04 | SuperNull | that took me about 3 days to find. |
| 17:23.27 | Rroet | p3nguin: so if I'm to setup a sip trunk for voipbuster... the old "bleg we don't speak about gui" put it as a type=peer in users.conf. Should I keep it there or move it to the sip.conf? |
| 17:23.44 | p3nguin | ~trunk |
| 17:23.44 | infobot | trunk is probably a word with varying definitions. In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment."; in Ethernet a trunk carries more than one 802.1q VLAN. There is no such thing as a "SIP Trunk" -- Don't use the term. the nose of an elephant |
| 17:23.56 | p3nguin | ~users.conf |
| 17:23.57 | infobot | [~users.conf] users.conf is a flaming pile of sh1t that takes the fine control of several perfectly usable asterisk config files and reduces them to the lowest common denominator and makes your system behave like a "toaster grade" PBX system. |
| 17:24.16 | Rroet | ~rofl |
| 17:24.16 | infobot | well, rofl is rolling on the floor laughing |
| 17:24.23 | Rroet | ;-) |
| 17:24.55 | Rroet | so. in other nicely used words: do not ever use users.conf .. thanks |
| 17:25.01 | p3nguin | My recommendations is to delete users.conf. Save from it what you want to save, first. |
| 17:25.18 | Rroet | all I need to save / know is in the grey mass between my ears. |
| 17:26.17 | Rroet | in all fairness then I should probably delete most files there and start from scratch with empty files. |
| 17:26.21 | Rroet | ~sip.conf |
| 17:26.37 | Rroet | ... .. |
| 17:26.37 | irroot | folks im going offline catch you arround going out for daughters birthday |
| 17:27.53 | p3nguin | Yes, I agree. If you've allowed a GUI to touch your files, there is a chance they have been mangled. |
| 17:28.34 | p3nguin | Although, for what it's worth, the Digium GUI is probably less intrusive than FreePBX. |
| 17:28.42 | Rroet | apt-get --purge remove asterisk && apt-get install asterisk |
| 17:28.48 | Rroet | I wasn't using freepbx |
| 17:34.00 | *** join/#asterisk navaismo (~navaismo@fixed-203-96-202.iusacell.net) |
| 17:34.51 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 17:35.14 | *** join/#asterisk justdave (~dave@unaffiliated/justdave) |
| 17:36.49 | *** join/#asterisk mirco (~mirco@p5B283BBA.dip.t-dialin.net) |
| 17:37.36 | *** join/#asterisk umay (~chris@64.92.218.72) |
| 17:40.25 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 17:40.48 | *** join/#asterisk GreatSUN (~greatsun@88-117-1-167.adsl.highway.telekom.at) |
| 17:40.55 | GreatSUN | re |
| 17:47.07 | *** join/#asterisk tm1000 (~tm1000@li251-245.members.linode.com) |
| 17:50.08 | *** join/#asterisk StaRetji (~BigAll@80.93.240.171) |
| 17:50.43 | *** join/#asterisk mateu (~mateu@missoula.org) |
| 17:51.06 | *** join/#asterisk michael-i (~anonymous@204.11.230.58.static.etheric.net) |
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| 17:53.52 | *** join/#asterisk pc-m (~pascal@modemcable094.94-70-69.static.videotron.ca) |
| 17:54.39 | *** join/#asterisk thehar (thehar@diddlebox.thehar.com) |
| 18:01.34 | mirco | Hi hi, I've got a prob with my 1.8.3.2 on Voyage-ONE Linux. When dialing into VM CLI tells: File digits/1F does not exist in any format |
| 18:02.34 | mirco | When I use find I see the file 1F.gsm exist's in "/usr/share/asterisk/sounds/digits/de/1F.gsm |
| 18:06.08 | [TK]D-Fender | mirco, Go verify your asterisk.conf varlibdir points there |
| 18:06.53 | *** join/#asterisk drift- (ae3001a4@gateway/web/freenode/ip.174.48.1.164) |
| 18:07.07 | atan2 | What is 999 <555> 1212 555 called? |
| 18:07.22 | drift- | hey how do i reset polycom to default settings? |
| 18:07.31 | drift- | isint there a button on boot i put in? |
| 18:07.34 | atan2 | Not the area code but the exchange bits :D |
| 18:07.39 | drift- | i got error blabla.cfg |
| 18:07.41 | drift- | polycom 501 |
| 18:12.51 | mirco | [TK]D-Fender: I've deleted "/var/lib/asterisk/sounds/" and set a link to "ln -s /usr/share/asterisk/sounds /var/lib/asterisk/sounds" cause in asterisk.conf I've: astvarlibdir => /var/lib/asterisk |
| 18:13.11 | mirco | but it didn't change the behavior |
| 18:17.37 | [TK]D-Fender | You should simply make sure the path is set |
| 18:17.59 | ruied_ | [TK]D-Fender, About the BLF status, is this the configuration needed? I'm making some tests but didn'd managed to change the led state yet. http://pastebin.com/gV8XMw7i |
| 18:18.44 | [TK]D-Fender | ruied_, exten => 802,hint,SIP/${EXTEN} <----------- not a SIP device you're pointing to. you need to point it to the device_state entry |
| 18:18.47 | mirco | [TK]D-Fender: how can I check which path is set for astvarlibdir? |
| 18:19.25 | [TK]D-Fender | mirco, "cause in asterisk.conf I've: astvarlibdir => /var/lib/asterisk" <- you already showed it to us. That is it right there |
| 18:19.35 | [TK]D-Fender | mirco, Then restart * |
| 18:19.51 | mirco | I already restarted * |
| 18:20.08 | [TK]D-Fender | ruied_, ${DEVICE_STATE(sip/802)}) <-- no, this is where you set your custom state |
| 18:20.12 | [TK]D-Fender | ~devstate |
| 18:20.13 | infobot | func_devstate is a module included with Asterisk 1.6 and above. A 1.4 backport is available here: http://svncommunity.digium.com/svn/russell/asterisk-1.4/func_devstate-1.4/func_devstate.c |
| 18:20.16 | [TK]D-Fender | ^^^ |
| 18:20.27 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 18:20.28 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 18:20.31 | [TK]D-Fender | rough docs on this |
| 18:20.48 | [TK]D-Fender | as it has chanegd through versions. There is not supposd to be any reference to a SIP device anywhere in yours |
| 18:21.31 | [TK]D-Fender | ruied_, and lines 3-8 in your PB ( http://pastebin.com/gV8XMw7i ) are dialplan lines, not sip.conf lines |
| 18:21.42 | [TK]D-Fender | ruied_, Yuo do not need to be messing with sip.conf for this |
| 18:23.23 | mirco | ls -l /var/lib/asterisk/sounds/digits/de/1F.gsm |
| 18:23.23 | mirco | -rw-r--r-- 1 asterisk asterisk 957 2005-11-04 11:24 /var/lib/asterisk/sounds/digits/de/1F.gsm |
| 18:23.44 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 18:23.51 | ruied_ | [TK]D-Fender, mistake, lines 3-8 are in extensions.conf, not in sip.conf I've changed sip.conf with extensions.conf in the dialplan... |
| 18:23.53 | mirco | is it possible that the locale/LANG can be the prob? |
| 18:24.27 | ruied_ | in the pastbin.... |
| 18:24.41 | mirco | I mean the de in the path I've pasted above? |
| 18:28.44 | mirco | yes the "de" in the path is the prob, after setting a link it work's |
| 18:30.14 | *** join/#asterisk shadowapex (~William@adsl-99-36-142-6.dsl.irvnca.sbcglobal.net) |
| 18:30.31 | shadowapex | Is it possible to set the RTP port range on a per SIP peer basis? |
| 18:30.43 | mirco | but the LANG seems to be set: -- <SIP/25-00000001> Playing 'digits/1F.gsm' (language 'de') |
| 18:30.44 | Katty | hi |
| 18:32.47 | *** join/#asterisk snaserowski (~samer@198.Red-88-31-128.staticIP.rima-tde.net) |
| 18:34.21 | [TK]D-Fender | shadowapex, No |
| 18:34.27 | ruied_ | [TK]D-Fender, 802,n,Set(DEVICE_STATE(Custom:lamp1)=INUSE) <----- isn't this to set the device custom state in the dialplan? |
| 18:34.29 | shadowapex | :< |
| 18:34.45 | [TK]D-Fender | mirco, You showed a folder from 2005. The folder layout changed during 1.6 |
| 18:34.49 | ruied_ | what is lam1 and lam2? green and red lamps? |
| 18:34.58 | [TK]D-Fender | ruiedyes, and your hint has to point to it |
| 18:35.14 | [TK]D-Fender | ruied_, not to "sip/123456". |
| 18:35.25 | [TK]D-Fender | Your hint is for the dynamic one you're inventing |
| 18:36.09 | *** join/#asterisk cerienjean (~iper@95.138.77.91) |
| 18:36.31 | *** join/#asterisk navaismo (~navaismo@189.146.55.106) |
| 18:38.48 | mirco | [TK]D-Fender: but after "ln -s /var/lib/asterisk/sounds/digits/de/1F.gsm /var/lib/asterisk/sounds/digits/" nearly all the other file's have been played: http://paste.ubuntu.com/701868/ |
| 18:39.56 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 18:40.08 | Rroet | I'm starting to hate the cli ;) |
| 18:40.18 | ruied_ | I have the hint in sip.conf, so, should I change it in the sip.conf? |
| 18:41.31 | Rienzilla | hi |
| 18:41.42 | p3nguin | hints in sip.conf? Sounds new. |
| 18:41.44 | ruied_ | I'll be back... have to go out.... |
| 18:42.13 | Rienzilla | I have a probably stupid question. If my user is put in a queue he gets a default music on hold. Is it possible for me to make it a normal ringing-tone instead of music? Or do I just need to replace all music on hold files with an mp3 containing that tone? |
| 18:42.19 | ruied_ | [TK]D-Fender,I'll continue to test it later, have to go out.... thanks! |
| 18:42.49 | [TK]D-Fender | ruied_ hints are dialplan. This has nothing to do with sip.conf. |
| 18:43.23 | [TK]D-Fender | Rienzilla, "core show application queue" |
| 18:43.57 | *** join/#asterisk f2knight (~ben@c-76-115-43-21.hsd1.or.comcast.net) |
| 18:44.00 | ruied_ | [TK]D-Fender, Ok, I'll try it... |
| 18:44.02 | ruied_ | thanks |
| 18:44.41 | *** join/#asterisk vinhdizzo (~vinh@dhcp-053181.ics.uci.edu) |
| 18:45.52 | Rroet | p3nguin: I start to hate the cli. with the gui (no comment) I had a voicemail system up & running including passwords etc.. but cli configs don't seem to like it. |
| 18:46.56 | p3nguin | You're doing it wrong. |
| 18:47.08 | Rroet | ofc, I'm a user. |
| 18:48.12 | p3nguin | Users should not be administering asterisk. |
| 18:48.20 | Rroet | ;) |
| 18:48.34 | p3nguin | If you want to configure voice mail, configure it in voicemail.conf. |
| 18:48.53 | Rroet | moved my users to sip.conf as they're sip softphones and made something that resembles a exten |
| 18:48.55 | p3nguin | Assign a mailbox to each peer if you want MWI. |
| 18:49.07 | Rroet | ~MWI |
| 18:49.07 | infobot | Message Waiting Indicator |
| 18:50.04 | *** join/#asterisk michael-i (~anonymous@204.11.230.58.static.etheric.net) |
| 18:50.19 | Rroet | let me guess, voicemail.conf is one of those things that the gui failed to use as well and they probably tossed the voicemail config somewhere in extensions.conf ? |
| 18:50.44 | p3nguin | extensions.conf isn't for voice mail. |
| 18:50.58 | p3nguin | But the GUI probably did not touch voicemail.conf. |
| 18:51.18 | michael-i | Hi all, just wondering if anyone noticed the inconsistencies in string escaping used in the examples on asterisk.org vs wiki.asterisk.org? Just compare https://wiki.asterisk.org/wiki/display/AST/Function_SHIFT vs http://www.asterisk.org/docs/asterisk/trunk/functions/shift |
| 18:52.31 | michael-i | probably an overzealous addslashes or escape when pulling from the db |
| 18:55.01 | Rroet | p3nguin: so I notice |
| 18:55.15 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 18:56.16 | [TK]D-Fender | Rroet, The GUI generated run-time-only config data based on users.conf. |
| 18:56.28 | Naikrovek | i am cursed. |
| 18:57.10 | Rroet | odd that I can't recreate the voicemail functionality... |
| 18:57.20 | *** join/#asterisk ferdna (~yup@cpe-67-10-220-35.elp.res.rr.com) |
| 18:57.22 | [TK]D-Fender | Rroet, Pastebin is your friend... |
| 18:57.23 | [TK]D-Fender | ~pb |
| 18:57.23 | infobot | [~pb] 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.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 18:57.50 | Rroet | I know pastebin, diffing right now with the old and new asterisk config dir |
| 18:59.36 | *** part/#asterisk shadowapex (~William@adsl-99-36-142-6.dsl.irvnca.sbcglobal.net) |
| 19:01.40 | p3nguin | I don't use users.conf, so I don't know how your GUI configured voicemail... but if you open the sample voicemail.conf, you'll see examples of how to set up mail boxes. |
| 19:06.33 | Naikrovek | i wonder if i'm the only person that is having issues with asterisk & snmp... i can NOT seem to make this work |
| 19:11.51 | r0m|u | Naikrovek, you want snmp for monitoring? |
| 19:12.20 | Naikrovek | indeed i do. getting snmp working for everything except asterisk is easy. can't get asterisk monitoring to work, though. |
| 19:12.27 | r0m|u | I am using nagios and works very well with asterisk |
| 19:12.33 | Naikrovek | i'm sure it does. |
| 19:13.02 | *** join/#asterisk bchia (~Adium@nat/digium/x-hzmwmikvtqxzjuyx) |
| 19:14.14 | Rroet | these previous 4 lines feel like a dejavu |
| 19:14.22 | Naikrovek | i can't walk the asterisk mibs though. no matter what i do, it will not work. most recently, I'm following these instructions: http://ofps.oreilly.com/titles/9780596517342/asterisk-Monitoring.html |
| 19:14.31 | Naikrovek | and i'm giving up. I'm going to hire a consultant to make this work. |
| 19:14.35 | Naikrovek | I've spent FAR too much time on this. |
| 19:14.46 | Naikrovek | I'd rather spend a few thousand to pay someone else to do it at this point |
| 19:15.02 | p3nguin | 0.0 |
| 19:15.03 | Naikrovek | actually |
| 19:15.13 | Naikrovek | I'd rather get a cisco phone system and pay someone to install that. |
| 19:15.19 | Naikrovek | but mgmt wont' help me there. |
| 19:15.33 | Naikrovek | i'm going home |
| 19:15.45 | [TK]D-Fender | Wow... smell the burnout... |
| 19:15.56 | Rroet | <-- done evil, ran -gui once more et voila, all things working.. now to diff against the vanilla install config directory and see the differences. |
| 19:16.37 | *** join/#asterisk blizzow (~jburns@67.50.165.58) |
| 19:18.05 | [TK]D-Fender | users.conf generates stuff you can't diff... |
| 19:18.22 | Rroet | I know |
| 19:18.29 | [TK]D-Fender | and what it does do is a poor start to learn from |
| 19:19.24 | Rroet | I understand where you're coming from, yet my twisted world works the other way arround. I learn by examples.. not by first ploughing through a pile of books to generate a first config file :( |
| 19:22.15 | *** join/#asterisk dirkD (~dirkD|not@84-245-20-6.dsl.cambrium.nl) |
| 19:22.45 | Rroet | I know, I'm weird. |
| 19:23.14 | p3nguin | The sample files provide good examples of how things work. |
| 19:23.32 | p3nguin | users.conf created by some GUI... does not. |
| 19:23.54 | p3nguin | There's really nothing you can take from users.conf and plug in anywhere else. |
| 19:23.56 | *** join/#asterisk gxdssoft (~gxdssoft@200.106.102.231) |
| 19:24.44 | Rroet | p3nguin: I wouldn't dare to argue that they're perfect for a complete and full understanding of all advanced possibilities the system has to offer. But any thing about 60 lines in sample files for a simple setup, in my humble opinion, means that it somehow lacks a certain useability. |
| 19:25.14 | Rroet | :s/about/beyond/ |
| 19:25.18 | Katty | hugs p3nguin |
| 19:25.21 | p3nguin | Suit yourself. All of us here have working systems without users.conf being involved. |
| 19:25.30 | Katty | p3nguin: how you doin |
| 19:25.35 | p3nguin | And we're not vim. :) |
| 19:25.37 | p3nguin | katty: Normal. |
| 19:25.50 | p3nguin | s/Normal/Typical/ |
| 19:25.53 | Rroet | p3nguin: the six editor I have never gained an m |
| 19:25.58 | Rroet | :s/an/a/ |
| 19:29.25 | *** join/#asterisk pc-m (~pascal@modemcable094.94-70-69.static.videotron.ca) |
| 19:31.46 | Rroet | p3nguin: where would you suggest setting up the user configuration? |
| 19:32.14 | p3nguin | If by "user configuration," you mean SIP phones... sip.conf is a good place. |
| 19:32.36 | [TK]D-Fender | And parts of dialplan required, so extensions.conf. And voicemail.conf for that. |
| 19:32.53 | [TK]D-Fender | Those are the 3 key basic "get off the ground" files. |
| 19:34.22 | Rroet | and how would I link them together? All using the same context name? |
| 19:35.14 | p3nguin | I like to use a context for all my phones called, not surprisingly, "phones" which includes all contexts that phones should have direct access to. |
| 19:36.15 | p3nguin | My "phones" context includes other contexts like internal, outbound, voicemail, etc. |
| 19:37.13 | [TK]D-Fender | Rroet, sip.conf defines the peers you will connect with (phones). Each peer will point to some context in extensions.conf that determines what they can dial and what will happen when they do. Voicemail.conf defines boxes you send calls to depending on what they dial in the dialplan. sip.conf also has the "mailbox" parameter for telling a phone that a box it should care about has messages |
| 19:37.39 | *** join/#asterisk devcoder (~leemelnyk@216.18.243.44) |
| 19:38.49 | *** join/#asterisk gxdssoft (~gxdssoft@200.106.102.231) |
| 19:39.26 | Rroet | then the whole idea of a dialplan as came from the -gui is nothing but a global context you can use in the sip.conf to refer a set of extension rules defined in the extensions.conf? |
| 19:42.02 | p3nguin | I can't even understand what you're asking there. |
| 19:43.15 | Rroet | my bad. I used the letter g u and i in that field. |
| 19:43.19 | Rroet | I'll try again: |
| 19:44.27 | Rroet | setting up a sip phone in the sip.conf I can attach a context= field to it. The value corresponds with a [value] field in the extentions.conf in which I can setup specific exten fields for this sip phone? |
| 19:45.04 | p3nguin | That's correct. |
| 19:46.17 | p3nguin | Here's an example of a dial plan: http://pastebin.com/Piqv4Egj |
| 19:46.53 | Rroet | if I'd paste something evil into a pastebin for sake of splitting up this sip phone entry into the proper files. Could you help me understand which entries whent into the wrong file and help me direct them to the right onces? only needed for 1 phone entry. the rest I would be able to figure out myself.. |
| 19:47.25 | p3nguin | I can certainly try. |
| 19:49.07 | Rroet | http://pastebin.com/Ka1vq8nv |
| 19:49.17 | p3nguin | Here's an example of a sip.conf: http://pastebin.com/tER2jGnY |
| 19:49.19 | Rroet | I've managed to filter out the passwords. |
| 19:50.09 | Rroet | My eye spotting some 0031 and Amsterdam area codes there ? :) |
| 19:52.46 | p3nguin | Here's how I'd change what you have: http://pastebin.com/jYWxnKen |
| 19:53.08 | p3nguin | Hmm? 0031 area code? Where? |
| 19:53.43 | p3nguin | I don't think I have any 0031 numbers in my dial plan anywhere. |
| 19:54.15 | Rroet | k |
| 19:54.19 | *** join/#asterisk blizzow (~jburns@184-96-97-124.hlrn.qwest.net) |
| 19:54.22 | Rroet | I'll have a look, thanks for your effort |
| 19:55.12 | [TK]D-Fender | p3nguin, PM |
| 19:55.19 | *** join/#asterisk ruied_ (~ruied@19.198.108.93.rev.vodafone.pt) |
| 19:55.21 | p3nguin | I responded. |
| 19:56.25 | Rroet | p3nguin: so thanks, this would be the sip.conf information. thanks. and gathering from your remark on the nat. I plan to use a softphone on my mobile, which I'll use both internally and externally but using the same number. would it be wise to keep nat=yes. Or would it be wiser to create 2 softphones for it? |
| 19:56.45 | p3nguin | I'd leave nat=yes for it. |
| 19:59.11 | mirco | I need to use the number of the phone I'm calling from, and this is not ${CALLERID(all)}, num or dnid! I'm talking about what is for example [24] in sip.conf |
| 19:59.27 | Rroet | how do I link this phone to the voicemail system? do I need to create the context= value as a [value] label in voicemail and add the hasvoicemail=yes line to it? |
| 19:59.44 | Rroet | voicemail.conf* |
| 20:01.08 | p3nguin | ~devicenames |
| 20:01.08 | infobot | Devices, extensions, and people should be entirely abstracted. Extension numbers are applied to people, and people are applied to devices. This means you should name your devices something unique to each device, such as an ID tag or asset tag number, or a MAC address. |
| 20:01.09 | [TK]D-Fender | Rroet, hasvoicemail is users.conf and has no parallel |
| 20:01.32 | [TK]D-Fender | Rroet, http://pastebin.com/8RWhKL8S <- here is a modded and reduced version of p3nguin's dialplan |
| 20:01.44 | p3nguin | Choose an extension for your phone to be called via. It looks like you've selected 6001. |
| 20:01.55 | p3nguin | Choose a voicemail box number. I'd probably use 6001 again. |
| 20:01.56 | mirco | ${SIPCALLID} is not what I expected: 24 but does the trick |
| 20:02.01 | [TK]D-Fender | Rroet, Where you can see that when you dial an extension for one of the 2 sample phones in there it can fall through to voicemail. |
| 20:02.02 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 20:02.40 | [TK]D-Fender | mirco, ${CHANNEL} + CUT() |
| 20:02.56 | Rroet | it calls the sip phone for 30 sec, then falls through to Voicemail. That's how I'm reading it. |
| 20:03.01 | Rroet | option 3, disconnect. |
| 20:03.02 | mirco | Thx, but why the cut()? |
| 20:05.12 | [TK]D-Fender | mirco, Go look at what is in that varibale and you tell me. |
| 20:05.30 | mirco | [TK]D-Fender: |
| 20:05.40 | [TK]D-Fender | Rroet, Correct. If dial fails for any reason then it will go into the box so they can leave a message |
| 20:05.53 | Rroet | I'll try |
| 20:05.54 | mirco | Sorry, was a mouse error |
| 20:07.11 | [TK]D-Fender | Rroet, Again, just a hacked up mod of p3nguin's sample |
| 20:07.37 | [TK]D-Fender | Rroet, now you'd need a sip.conf for the peers to match, and 2 netires in voicemail.conf, but this isn't really hard. |
| 20:07.40 | Rroet | I know, I don't intent to use it in production. I'm using it as a base sample to see where I'm going. |
| 20:07.45 | p3nguin | I tried to touch on several concepts in my example. Maybe a bit much for your case right now. |
| 20:07.46 | [TK]D-Fender | I should probably write a small book on it... |
| 20:08.05 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202) |
| 20:09.58 | mirco | [TK]D-Fender: how shall I use cut() |
| 20:11.35 | pabelanger | michael-i: looks like a bug in the wiki software. ${foo(blah)} needs to be escaped otherwise the wiki thinks it is a function call. But looks to break when using multiple together |
| 20:12.51 | michael-i | pabelanger: I've spent the better part of days messing with escaping errorsâ¦not fun. |
| 20:14.55 | *** join/#asterisk scubes13 (~scubes13@cpe-024-168-196-000.sc.res.rr.com) |
| 20:15.05 | pabelanger | nods |
| 20:15.40 | Rroet | [TK]D-Fender: you deduced a line like exten => 100,1,Dial(SIP/100,30); <-- could I change that into: exten = _#1XX,1,Dial(SIP/${EXTEN}/100,30); |
| 20:15.56 | Rroet | or is it imperative that I keep using the exact numbers? |
| 20:16.19 | [TK]D-Fender | mirco, So what do you see in the variable? |
| 20:16.21 | p3nguin | I don't recommend using # before extension numbers because it breaks dialing of several types of phones. |
| 20:16.25 | p3nguin | It's the "dial now" key. |
| 20:16.52 | mirco | [TK]D-Fender: for example: SIP/25-00000020 |
| 20:17.10 | p3nguin | What are you trying to turn that into? |
| 20:17.16 | Rroet | so _1XX would be better. but that would be a valid line for having multiple phones in the 1XX range to fall under this exten rule? |
| 20:17.28 | [TK]D-Fender | <Rroet> [TK]D-Fender: you deduced a line like exten => 100,1,Dial(SIP/100,30); <-- could I change that into: exten = _#1XX,1,Dial(SIP/${EXTEN}/100,30); <-the # is a literal char in the dial. Next lose the /100 |
| 20:17.52 | [TK]D-Fender | Rroet, EXTEN => _1XX,1,Dial(SIP/${EXTEN},30) |
| 20:18.14 | Rroet | p3nguin: from the deduced file [TK]D-Fender created in pastebin I see you have 2 phones with each a seperate 3 lines of exten. While they're mostly the same. I was hoping to create a general rule that would work for both phones. |
| 20:18.16 | [TK]D-Fender | Rroet, "_" denotes that what follows is a pattern (x's, n's, etc). |
| 20:18.50 | p3nguin | You can use a pattern, but I don't because extensions really should not match the device name. |
| 20:18.50 | [TK]D-Fender | Rroet, using range patterns as opposed to hard numbered is a start, then comes macro's and gosub's |
| 20:19.06 | [TK]D-Fender | Rroet, You seem to have something of a programming background, which is good |
| 20:19.10 | Rroet | ok, so in my case using 6XXX range numbers for internal calling I should skip the _ and # |
| 20:19.22 | [TK]D-Fender | Rroet, no, "_" is required |
| 20:19.24 | Rroet | [TK]D-Fender: I'm in It, true. But not a programmer. |
| 20:19.26 | p3nguin | XXX is part of a pattern. |
| 20:19.39 | Rroet | ok. |
| 20:20.01 | p3nguin | exten => 6XXX is litterally a 6 and three Xs. |
| 20:20.23 | p3nguin | exten => _6XXX is a 6 and three other digits 0-9. |
| 20:20.40 | Rroet | gotcha. |
| 20:21.41 | p3nguin | But with phones having names such as 000011112222, you're going to need to hard-code a Dial() for every phone. I do it in a Gosub(). |
| 20:21.59 | [TK]D-Fender | Macro() is your friend... |
| 20:22.16 | p3nguin | or use astDB. |
| 20:22.38 | [TK]D-Fender | p3nguin, Which is defnct with 1.10 from what was written |
| 20:22.46 | p3nguin | Set(DEVICE=${DB(phones/${EXTEN}/device)}) |
| 20:22.50 | [TK]D-Fender | p3nguin, that is going to FUBAR FreePBX and so many user's dialplans |
| 20:22.54 | p3nguin | Dial(${DEVICE}) |
| 20:23.09 | Rroet | which would explain why the webinterface, which we will not refer to ever again, uses the extensions as phonenames |
| 20:23.48 | [TK]D-Fender | Rroet, you could do [john] in sip.conf and use EXTEN => 100,1,Dial(DIP/john, 30) |
| 20:23.56 | [TK]D-Fender | SIP* |
| 20:24.07 | *** join/#asterisk kerx (~kerx@76-240-161-60.lightspeed.irvnca.sbcglobal.net) |
| 20:24.07 | p3nguin | chortles |
| 20:24.52 | scubes13 | p3nguin - a few days ago, you helped me setup my iptables to restrict SIP/etc access to certain dynamic dns entriesâ¦. if I only have ports open to certain dns/ip's⦠should I have a need to run fail2ban? |
| 20:24.55 | Rroet | I got the grasp of that. |
| 20:26.13 | p3nguin | scubes13: Theoretically you'll already be blocking everything that you don't want to specifically allow, so no. |
| 20:26.14 | [TK]D-Fender | ok, hcekcout time here, BBIAB |
| 20:26.25 | [TK]D-Fender | ok, checkout time here, BBIAB |
| 20:26.27 | [TK]D-Fender | AGH |
| 20:26.32 | Rroet | p3nguin: for the exten line towards the voicemail.. |
| 20:26.46 | scubes13 | p3nguin - thanks mate! saved me +100MB ram |
| 20:27.05 | Rroet | does it require the phonenumber, or phonename? |
| 20:27.23 | p3nguin | Nope, it requires a voice mail mail box name. |
| 20:27.32 | p3nguin | fail2ban is most often used where you have lots of things open and unwanted people often try to get in. |
| 20:27.42 | Rroet | ok, the voicemailbox name we gave it in sip.conf |
| 20:27.59 | p3nguin | The name you gave it in voicemail.conf, since that's where voice mail is configured. |
| 20:28.34 | Rroet | didn't I configure it in sip.conf with : mailbox = 6001 ? |
| 20:28.44 | p3nguin | That's only for MWI. |
| 20:28.51 | Rroet | ~MWI |
| 20:28.51 | infobot | Message Waiting Indicator |
| 20:28.54 | Rroet | ahh. |
| 20:30.45 | *** part/#asterisk tuxxie (~Ryan@rrcs-70-63-90-226.midsouth.biz.rr.com) |
| 20:30.45 | Rroet | then I think I've not received a voicemail.conf example.. let me try to figure this out ;) |
| 20:30.48 | p3nguin | Let me give you a quick rundown of how I do my own office system. I have a phone. The MAC address is 000011112222, so the phone's name in sip.conf is 000011112222. Now my name is Rob, so my extension is 762 (ROB on the keypad). My voicemailbox is also 762. This allows my extension and my voicemailbox number to be tied to me, and I can use it no matter what phone I decide to use by just changing the phone that Dial() dials. |
| 20:31.44 | Rroet | ok, I'll try |
| 20:31.44 | p3nguin | So if I move to a new office with a new phone, I can change extension 762 to dial AAAABBBBCCCC instead of 000011112222. |
| 20:32.09 | p3nguin | And my extension will still be 762 and my voicemailbox number will still be 762 as well. |
| 20:32.26 | p3nguin | Those are personal for me, where phones are just devices that people use sometimes. |
| 20:33.04 | Rroet | so my voicemail rundown would be: 6001 => 6001,Full Name,e@ma.il,,timezone ? |
| 20:34.36 | p3nguin | I'd have to look to confirm exact syntax, but it looks pretty close. |
| 20:34.44 | p3nguin | (if not exactly right) |
| 20:35.13 | Rroet | I'm looking at the sample voicemail.conf and the first 2 values (eg. 6001 => 6001) are both differnt value's in each sample. |
| 20:35.24 | Rroet | I haven't figured that part out yet. |
| 20:35.33 | p3nguin | 1234 => 4321,Name,email |
| 20:35.47 | p3nguin | 1234 is the mailbox number; 4321 is the password for mailbox 1234. |
| 20:36.18 | Rroet | aha! |
| 20:36.25 | p3nguin | There's a setting to make it so that when the mailbox passwd is the same as the mailbox number, the user has to change the passwd on first call in. |
| 20:36.39 | p3nguin | Change means change, too. It does not mean set it to the same thing again. |
| 20:37.07 | p3nguin | I wasted a lot of some with someone insisting he was changing the password when instead he was setting it to the same thing. |
| 20:37.27 | p3nguin | s/some/time/ |
| 20:39.35 | Rroet | ;) |
| 20:43.10 | *** join/#asterisk singler (~singler@78-60-139-121.static.zebra.lt) |
| 20:43.24 | Rroet | Ohh, the one thing I hadn't figured out yet though.. where do I set the phonenumber of a SIP phone? I have a mailbox id, but I don't have a phonenumber yet.. or is that the hardcoded exten = 6001 in the extensions.conf ? |
| 20:43.43 | p3nguin | ~devicenames |
| 20:43.44 | infobot | Devices, extensions, and people should be entirely abstracted. Extension numbers are applied to people, and people are applied to devices. This means you should name your devices something unique to each device, such as an ID tag or asset tag number, or a MAC address. |
| 20:43.48 | p3nguin | This ^ |
| 20:44.19 | p3nguin | You choose the extension for the person, then apply it to the phone you want the person to be using. |
| 20:44.29 | p3nguin | See my example above. |
| 20:44.39 | p3nguin | extension 762 ... |
| 20:44.46 | p3nguin | I tried to explain it. |
| 20:44.52 | Rroet | yes, but the example is a bit abstract.. in my simple mind I'm the phone |
| 20:45.01 | p3nguin | You're not a phone. |
| 20:45.17 | p3nguin | At least I assume that you're a human. |
| 20:45.27 | devil_evoxxx | hi, there is something using * from 1.8.5 to 1.8.7...and can make me a test? |
| 20:45.50 | Rroet | litterally it's true. but I have only 1 mobile ip phone I'd like to use so hence I'd make the statement, I'm the phone. wherever the mobile goes, I go ;) |
| 20:46.20 | p3nguin | You're still not a phone. |
| 20:46.35 | p3nguin | You can implant the phone into your person, then we'll consider calling you a phone. |
| 20:47.02 | p3nguin | Did you decide what extension YOU want to use for people to call YOU on your phone? |
| 20:47.25 | p3nguin | I thought you decided on 6001. Was I incorrect? |
| 20:47.26 | Rroet | I'm not up for that yet.. current implementations of phone implants are considered fetishisms |
| 20:48.00 | Rroet | yes, I'm for clear example sake always talking about 6001 |
| 20:48.50 | p3nguin | Does your phone/device have a unique identifier of any sort? Asset tag number, MAC address? Something that NO OTHER device will have. |
| 20:49.20 | p3nguin | serial number? |
| 20:49.32 | p3nguin | service tag number (in the case of Dell devices). |
| 20:49.50 | Rroet | computer mac address, but I'm considering a unique name I'd give it.. like mymobile or hermobile |
| 20:50.34 | Rroet | I know it's invalid.. it's like programing C++ and using Marvell comicbook superhero's as variables. |
| 20:50.36 | p3nguin | Okay, I can live with that without having too much anguish. |
| 20:50.49 | *** join/#asterisk nighty^ (~nighty@TOROON12-1279662182.sdsl.bell.ca) |
| 20:51.07 | p3nguin | So you know what extension you're going to use; we'll set that up in extensions.conf. |
| 20:51.25 | p3nguin | You know what name you're going to give your SIP device; we'll set that up in sip.conf. |
| 20:51.32 | *** join/#asterisk fskrotzki (~fskrotzki@cpe-67-253-245-174.rochester.res.rr.com) |
| 20:51.49 | p3nguin | You'll probably end up using the extension number as the voicemail number; we'll set that up in voicemail.conf. |
| 20:51.56 | p3nguin | Seeing a pattern? |
| 20:52.18 | Rroet | ok.. |
| 20:53.29 | *** join/#asterisk atan (~atan@unaffiliated/atan) |
| 20:53.37 | Rroet | so exten => 6001,1,Dial(SIP/mymobile,30) && exten => 6001,2,Voicemail(6001@default,u) |
| 20:55.26 | p3nguin | That seems reasonable. |
| 20:56.37 | Rroet | and now my desktop hates me. logging in with mostprobably the worst softphone software it doesn't allow me to log on. |
| 20:57.45 | Rroet | (3cx) |
| 20:57.48 | *** part/#asterisk vinhdizzo (~vinh@dhcp-053181.ics.uci.edu) |
| 20:58.02 | p3nguin | Here's something I do: http://pastebin.com/JbSFf4Tb |
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| 20:59.00 | Rroet | DONTCALl... TORTURE, CONGESTION.. i'd better go hide |
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| 21:00.04 | p3nguin | Those are possible DIALSTATUS values. |
| 21:01.27 | p3nguin | The only place I hard-code my device names is in the sip.conf and in the astdb. |
| 21:02.41 | p3nguin | Based on the convention I described earlier, in the DB, phones/762/device would have a value of 000011112222. |
| 21:02.44 | Rroet | I don't know what the astdb is just yet. But I'll leave that for another night. |
| 21:02.53 | p3nguin | Wait, that's not accurate... |
| 21:02.57 | p3nguin | Based on the convention I described earlier, in the DB, phones/762/device would have a value of SCCP/000011112222. |
| 21:03.08 | p3nguin | or in your case SIP/000011112222. |
| 21:04.27 | p3nguin | Sure, you don't necessarily need to use the astdb. But if you want, it is there for the using. |
| 21:04.35 | Rroet | [Oct 3 23:04:13] NOTICE[16286] chan_sip.c: Call from 'JelleDT' to extension '6001' rejected because extension not found in context 'incoming'. |
| 21:04.43 | Rroet | I tortured the beast a bit too much. |
| 21:05.08 | p3nguin | So you have a phone named 'JelleDT' configured in sip.conf. You gave it a context of incoming. |
| 21:05.39 | p3nguin | Certainly not my preference of a context, but we can still make calls go if you continue using it. |
| 21:06.00 | p3nguin | Where did you put extension 6001? |
| 21:06.03 | Rroet | no, it had the right context name to it... but context was conext |
| 21:06.04 | p3nguin | under what context? |
| 21:06.36 | Rroet | phones |
| 21:06.45 | p3nguin | Oh, so you have incoming as the context set in the general section. |
| 21:06.58 | Rroet | I for example sake borrowed the example from [T....fender |
| 21:07.17 | *** part/#asterisk gabeh (~gabeh@67.133.93.182) |
| 21:07.18 | Rroet | now at the end it didn't pass to voicemail. I have something to look into ;) |
| 21:07.41 | p3nguin | Typically you'll be putting only includes in the phone context so that you don't ever create overlaps between incoming and outgoing contexts. |
| 21:08.10 | p3nguin | Did you remember to reload the voice mail module? "voicemail reload" should do it. |
| 21:08.10 | Rroet | yes. |
| 21:08.16 | p3nguin | voicemail show users |
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| 21:08.25 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 21:08.25 | Rroet | [phones] is a includes only section |
| 21:08.26 | *** join/#asterisk chazzam (~chazz@173-24-236-90.client.mchsi.com) |
| 21:08.35 | p3nguin | If you see yours listed, that part should be good. |
| 21:08.36 | *** join/#asterisk fipu (~fipu@fipu.usr.sh) |
| 21:08.53 | Rroet | pp_voicemail.c: No entry in voicemail config file for '6002' |
| 21:09.02 | Rroet | logging teaches a lot ;) |
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| 21:09.16 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 21:09.23 | Rroet | using 2 softphones.. mobile and desktop to call eachother. |
| 21:09.43 | dym | evening! |
| 21:10.03 | Rroet | for the 2 600X numbers I use [phones] section in the voicemail.conf. I take it it needs a context like that as well? |
| 21:10.18 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 21:10.58 | p3nguin | No. If you have a voicemail context (in voicemail.conf), you have to make sure you call VoiceMail(6001@phones) rather than @default. |
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| 21:11.18 | *** join/#asterisk micols (~0x2AA7F64@rlogin.dk) |
| 21:11.23 | Rroet | ouch, that's a definet error between keyboard and chair.. will fix |
| 21:11.30 | dym | http://www.voip-info.org/wiki/view/Asterisk+config+voicemail.conf |
| 21:11.35 | Rroet | :s/definet/definete/ |
| 21:11.35 | dym | for further enlightment |
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| 21:12.21 | *** mode/#asterisk [+o Qwell] by ChanServ |
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| 21:15.08 | Rroet | p3nguin: by decoupling the user from its phone, as you described might be a good thing, I know get the issue that when I'm calling my voicemailbox (999) it can't read the username which I need to get to my voicemails |
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| 21:22.30 | p3nguin | What method are you using to get your voicemails? |
| 21:23.02 | Rroet | I'll beat you to it: You're doing it wrong |
| 21:23.12 | Rroet | exten = 999,1,VoiceMailMain(${CALLERID(num)}@default) |
| 21:23.53 | p3nguin | You want 999 to be your "voicemail extension" if I get you correctly. |
| 21:24.10 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:24.19 | p3nguin | Unless you moved you mailboxes into the default voicemail context (in voicemail.conf), you're looking in the wrong context. |
| 21:24.21 | Rroet | correct. |
| 21:24.30 | Rroet | all are in @default |
| 21:25.10 | Rroet | but my phones have a devicename now or in my case a JelleDT or iPhoneJelle .. which is ofcourse different from the extention numbers. |
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| 21:25.44 | p3nguin | Do you want 999 to be only for your own voicemails (no matter which phone is calling it), or do you want that to be the extension where you have a chance to enter any mailbox number? |
| 21:26.44 | Rroet | in a perfect world I'd log on to any phone with a given name and password and it'll automatically allow me access to the mailbox |
| 21:27.10 | p3nguin | I use *86 (*VM) for that: http://pastebin.com/UHYXLbjj |
| 21:27.31 | p3nguin | Change mine to 999 if you want 999 instead. |
| 21:28.23 | p3nguin | This assumes you have correctly configured callerid on the phone's entry in sip.conf. |
| 21:29.09 | Rroet | I don't have a callerid setup in the sip.conf |
| 21:29.25 | Rroet | just the [name] and a mailbox=600X |
| 21:29.53 | p3nguin | Just to see what caller id information you have without setting it... |
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| 21:30.31 | p3nguin | Put a line into the extension to show you the value of ${CALLERID(all)}. |
| 21:31.01 | p3nguin | Verbose(${CALLERID(all)}) or NoOp(${CALLERID(all)}) will be fine for that. |
| 21:32.10 | Rroet | in the logging all it states is app_voicemail.c: Couldn't read username |
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| 21:32.47 | p3nguin | Yeah, when you don't have a caller id number and you are parsing the mailbox based on the caller id number, that's bound to happen. |
| 21:33.01 | p3nguin | So take a look at what is actually in callerid. |
| 21:33.34 | Rroet | devicename was in it. |
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| 21:35.49 | Rroet | odd thing is that if I change my extension or ID in the Sip credentials of the softphone it won't allow me to reconnect to asterisk. I'm doing it wrong |
| 21:37.26 | Rroet | I'm calling it a night for now. Thanks for your patience p3nguin. It help me grasp the mere basics of the software pbx. |
| 21:42.33 | *** part/#asterisk Rroet (~Rroet@5354C380.cm-6-5d.dynamic.ziggo.nl) |
| 21:42.38 | [TK]D-Fender | Rroet: Great progress for your start |
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| 21:51.15 | ChannelZ | Look at you, being all positive :P |
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| 21:59.53 | [TK]D-Fender | <PROTECTED> |
| 22:00.00 | [TK]D-Fender | ChannelZ: ^ |
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| 22:04.16 | beek | Evenin' [TK]D-Fender |
| 22:08.29 | *** join/#asterisk KingDavidNYC (~Chris1232@pool-74-96-172-188.washdc.fios.verizon.net) |
| 22:08.45 | KingDavidNYC | Hello everyone |
| 22:09.17 | [TK]D-Fender | beek: y0 |
| 22:09.24 | KingDavidNYC | yo |
| 22:10.08 | beek | [TK]D-Fender: ...Nice to have you back where you belong... |
| 22:11.00 | KingDavidNYC | TK- Hey pal, I am glad you remember me, I liked hanging here |
| 22:11.29 | KingDavidNYC | Can someone please help me, how do I get to monitor a digium port?, I mean, I will like to see all 24 channels, and it looks like the card is working, but I just |
| 22:11.50 | KingDavidNYC | just want to see bits passing through the channels |
| 22:12.13 | KingDavidNYC | there is dahdi_monitor, but seems to show only one channel at a time |
| 22:12.47 | KingDavidNYC | I will like to monitot all 24 channels since I dont know in what channel the call is coming in |
| 22:13.03 | KingDavidNYC | I dont know how to use dahdi_maint |
| 22:16.25 | [TK]D-Fender | KingDavidNYC: straight CLI |
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| 22:16.31 | [TK]D-Fender | dahdi show channels |
| 22:16.40 | [TK]D-Fender | dahdi show channel [X] |
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| 22:20.20 | dym | when compiling asterisk from source and then wanting to do a version upgrade - can i simply unpack the new source and compile it? |
| 22:20.26 | KingDavidNYC | really?, wow, i am gonna do it right now... Thanks TK, you are a genious |
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| 23:00.15 | p3nguin | What files should I be using for Polycom phones, combined or split? |
| 23:03.05 | p3nguin | Useragent field from sip show peer says PolycomSoundPointIP-SPIP_500-UA/2.1.2.0078 |
| 23:03.27 | p3nguin | This means it's a SoundPoint IP 500 using 2.1.2, right? |
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| 23:33.13 | p3nguin | How can I tell what BootROM I should be using on this old Polycom 500? |
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| 23:33.41 | p3nguin | It won't boot up, so I need to supply the appropriate BootROM files to it. |
| 23:33.52 | Naikrovek | p3nguin: there is a compatability chart somewhere on their website. you'll want the latest supported version of the bootrom and sip |
| 23:33.58 | p3nguin | I just don't know which one to use. |
| 23:34.48 | p3nguin | I didn't necessarily want to upgrade the SIP app if I didn't need to. Since it just won't boot up, I'd like to just give it a good BootROM and let it go for the time being. |
| 23:35.32 | Naikrovek | http://downloads.polycom.com/voice/voip/sip_sw_releases_matrix.html |
| 23:35.54 | p3nguin | I guess I'll start with 4.1.2 and see if it mentions the 500 in it. |
| 23:36.17 | p3nguin | Well no I won't... there is no readme in the zip. |
| 23:37.18 | p3nguin | It's not on there. |
| 23:37.21 | p3nguin | It's a 500. |
| 23:37.41 | p3nguin | The release notes for BootROM 4.1.2 says it'll work on a 500. |
| 23:38.01 | Naikrovek | there ya go then |
| 23:38.28 | Naikrovek | http://support.polycom.com/PolycomService/support/us/support/voice/soundpoint_ip/soundpoint_ip500.html |
| 23:38.31 | p3nguin | But if I don't give it the same version, it'll format the phone when it loads the new version. Since the phone was configured by hand, there is no backup of config settings. |
| 23:40.28 | p3nguin | Let's hope 3.2.3 is what was on it. |
| 23:44.29 | p3nguin | My first attempt at recovering a Polycom, and the phone is 1000 miles away. I'd so much rather be touching it to do this. |
| 23:45.52 | p3nguin | When a 500, a 501, and a 550 are all on the same LAN, do I need to have a common BootROM and SIP app that is compatible with all of them, or can each phone look for its own versions of files? |
| 23:47.21 | Naikrovek | wouldn't it be easier to give the person a spare, then ship the busted phone to you? |
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| 23:47.46 | Naikrovek | if you have a windows dhcp server you can set up a reservation so that one sees it's own boot file location, unmolesting the other phones |
| 23:48.08 | Naikrovek | the other phones won't bother with firmware updates until they reboot, so as long as there's no power outage while you're doing this, you'll be fine. |
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| 23:50.45 | p3nguin | oh, and add 650 to that. |
| 23:52.44 | p3nguin | It looks like SIP 2.1.2 would support all those phones anyway, and all of them are using that version or a lower version anyway. |
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| 23:58.41 | [TK]D-Fender | Never touch the bootrom unless you absolutely have to |
| 23:58.46 | p3nguin | I have to. |
| 23:58.57 | [TK]D-Fender | Why is that? |