| 00:10.42 | *** join/#asterisk koffel (~ekoffel@c-68-82-146-18.hsd1.nj.comcast.net) |
| 00:11.02 | koffel | is asterisk now part of this channel? |
| 00:28.11 | navaismo | which GUI are you using? |
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| 03:05.19 | *** join/#asterisk axscode (axscode@110.55.118.82) |
| 03:05.56 | axscode | hi guys, is there a plugin/module thats function almost trixbox... |
| 03:07.32 | lorsungcu_ | no. |
| 03:07.44 | lorsungcu_ | i think you are looking for #freepbx |
| 03:10.10 | axscode | freepbx is not asterisk? |
| 03:10.39 | lorsungcu_ | it uses asterisk |
| 03:11.03 | lorsungcu_ | it's just a bunch of scripts that automate normal pbx functions |
| 03:15.06 | axscode | great, i hope it can be install not as an iso |
| 03:31.20 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 03:32.08 | [TK]D-Fender | axscode: How "almost" are you looking for? |
| 03:33.25 | [TK]D-Fender | axscode: FreePBX is basically a comlpete front end that takes over all the responsibilities of configuring * for you and you usually stick to doing things it's way. So if youre loking for too much custom work then GUI's are not for you. |
| 03:35.12 | axscode | im trying to install freepbx now.. |
| 03:35.33 | axscode | some trouble during installation, hope i can get through with it. |
| 03:39.35 | *** join/#asterisk dijib (~dijib@208-96-84-35.eastlink.ca) |
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| 03:42.40 | lorsungcu_ | axscode, you need to bring this to the proper channel |
| 03:42.47 | lorsungcu_ | #freepbx will be able to help you |
| 03:43.34 | *** join/#asterisk AlfE_ (~quassel@83-215-36-12.bruck.dyn.salzburg-online.at) |
| 03:44.01 | xbp | what is the issue |
| 03:44.35 | xbp | if you look on powerpbx.com you will find a good installation guide for asterisk + freepbx |
| 03:45.07 | xbp | start from scratch if you're not too familiar with linux and have gotten yourself into something halfway and gone back and forth among bugs |
| 03:45.26 | xbp | you have to be pretty savvy to go in and look up logs and find errors from taking shortcuts |
| 03:45.33 | xbp | shortcuts cost ya everytime.. Are you using a vps? |
| 03:45.55 | xbp | i install asterisk on vps slices so I deal with headaches all th etime |
| 03:45.56 | xbp | OH. |
| 03:45.59 | xbp | run along. |
| 03:46.17 | xbp | types to the floor which never listens |
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| 05:41.40 | waxstone | hi all, I have asterisk setup with sip clients working for my home office im now trying to setup landline should i go for external setup like http://tinyurl.com/c2sw9nf for $52 or with an internal card? only one line required |
| 05:45.49 | jpsharp | Well, if you're looking to connect to a phone line that particular box won't work. |
| 05:46.03 | jpsharp | But I'd go with an external SIP gateway myself. Much less hassle. |
| 05:47.15 | jpsharp | But if you're looking to connect plain telephones to Asterisk, you could use that box. |
| 05:48.27 | waxstone | jpsharp: could you explain the difference between phone line and plain telephone? |
| 05:48.47 | waxstone | jpsharp : trying to hook up normal home phone line |
| 05:49.11 | waxstone | ok i understand now i believe |
| 05:49.20 | waxstone | normal handsets |
| 05:49.39 | waxstone | wow thanks saved me some $$$. how about http://tinyurl.com/ceq3l3 do you have any you know work or could recommend? |
| 05:50.42 | waxstone | i want an otsid ecall to my home to pass into asterisk so phone line |
| 05:50.47 | waxstone | outside** |
| 05:51.21 | *** join/#asterisk ThomasLocke (~ThomasLoc@pdpc/supporter/active/thomaslocke) |
| 05:56.17 | kaldemar | waxstone: FXO is what you need in that case. FXS provides battery and is used to connect phones. FXO is for lines. |
| 05:57.25 | waxstone | kaldemar: thanks ill search of an external gateway with one FXO line |
| 05:57.27 | *** join/#asterisk deo_ (~dnepangue@112.198.79.31) |
| 06:02.50 | waxstone | kaldemar: http://tinyurl.com/ceq3l3 has 1 FXO port and 1 FXS port so that should work correct? |
| 06:04.31 | kaldemar | the site you link to does not answer. |
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| 06:06.09 | waxstone | sorry dk what happened ther |
| 06:06.54 | waxstone | http://www.amazon.com/Cisco-SPA3102-Voice-Gateway-Router/dp/B000FKP55U/ref=sr_1_1?ie=UTF8&qid=1346738802&sr=8-1&keywords=spa3102 |
| 06:09.17 | kaldemar | sure, that will do the job. |
| 06:10.37 | waxstone | kaldemar: thanks! |
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| 06:30.12 | mirela666 | mornin' all |
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| 06:35.33 | v0lZy | morning |
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| 07:32.47 | reisi | hi everyone! trying to use AMI over asterisk-java; any ideas why does not the client library receive information on any channels on the server? (server is 1.8; not sure if I should configure this with more than allowing login with username/pw) |
| 07:38.05 | *** join/#asterisk bulkorok (~bulkorok@85.183.36.36) |
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| 07:46.14 | kaldemar | reisi: you should configure the read/write parameters in manager.conf to allow the manager user to actually get some data. |
| 07:50.27 | reisi | kaldemar: that i have done |
| 07:51.05 | reisi | kaldemar: i am able to originate calls and execute for example "sip show peers" but no access to channels (via listener or ManagerConnection) |
| 07:51.06 | kaldemar | attach to CLI and do "manager set debug on" and see what happens when you try to do something. |
| 07:51.12 | reisi | kaldemar: roger that |
| 07:51.32 | kaldemar | how are you trying to get the channel information? |
| 07:53.32 | reisi | kaldemar: there are multiple built-in ways inside https://blogs.reucon.com/asterisk-java/ ; or actually just one: it should receive a StatusEvent (not sure what this is in AMI terms) and then update it's in-java caches according to that event |
| 07:53.56 | reisi | kaldemar: apparently those events are not being sent; it could though be that this asterisk-java library is just broken at the moment |
| 07:54.40 | kaldemar | events are something you need to enable while loggin in to the AMI with "Events: On". |
| 07:54.57 | *** join/#asterisk vlad_starkov (~vlad_star@ppp91-79-87-44.pppoe.mtu-net.ru) |
| 07:57.41 | kaldemar | ManagerConnection.addEventListener() and what goes along with it is probably what you're missing. |
| 07:58.07 | reisi | actually not sure if that is it, but there's no event mask being sent |
| 07:58.43 | kaldemar | if you don't want to use events, a mere CommandAction("core show channels concise") and some parsing gets you the info (unless there's a class that handles core channel information on demand nowadays). |
| 08:00.56 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 08:05.01 | reisi | still does not work (at least through the abstraction i'm trying to use); originate is successfully queued, but no actual unique channel id gets returned ever |
| 08:07.27 | reisi | so no OriginateEvent is received |
| 08:08.03 | reisi | set manager debug on does not seem to be debugging anything |
| 08:14.17 | kaldemar | it's not too verbose. you can always use tcpdump to verify what goes on. |
| 08:16.23 | *** join/#asterisk Vince-0 (c4d7bcf4@gateway/web/freenode/ip.196.215.188.244) |
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| 08:24.20 | *** join/#asterisk v0lZy (~Thunderbi@mail.silk-group.net) |
| 08:33.12 | v0lZy | hey people |
| 08:33.16 | v0lZy | cananyone clue me in on CDR |
| 08:33.42 | v0lZy | Do i need to do anything in the dialplan |
| 08:33.48 | v0lZy | or just configure cdr.conf |
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| 08:39.27 | *** part/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 08:44.25 | *** join/#asterisk DREMA (c98d9c56@gateway/web/freenode/ip.201.141.156.86) |
| 08:44.42 | DREMA | Hello everyone! |
| 08:45.32 | DREMA | Is anyone alive? |
| 08:46.02 | fenrus | yea |
| 08:46.14 | DREMA | How are you man! |
| 08:46.26 | DREMA | I'm looking for some help with my asterisk box |
| 08:46.30 | DREMA | :/ |
| 08:46.37 | fenrus | im quite all right |
| 08:46.39 | ChannelZ | Holy shit! |
| 08:46.48 | fenrus | if you ask your question instead of asking if you can ask.. ;) |
| 08:46.52 | ChannelZ | No one ever comes here for that! |
| 08:47.04 | DREMA | Hahahaha, sorry sorry |
| 08:47.30 | DREMA | to the point |
| 08:48.04 | DREMA | inbound PSTN calls work fine |
| 08:48.16 | DREMA | trying to dial out goes to busy tone |
| 08:48.20 | DREMA | this is on my logs |
| 08:48.29 | DREMA | WARNING[1386]: chan_sip.c:8435 process_sdp: Unsupported SDP media type in offer: audio 8024 RTP/SAVP 4 0 8 2 97 102 100 101 |
| 08:49.03 | *** join/#asterisk brdude (~brdude@c-24-7-76-160.hsd1.ca.comcast.net) |
| 08:49.19 | ChannelZ | Well it's only a warning but not necessarily why your call is failing |
| 08:49.33 | DREMA | to be precise, on the core debug I got this |
| 08:49.34 | DREMA | DEBUG[1386]: chan_sip.c:8334 process_sdp: Processing session-level SDP v=0... UNSUPPORTED. [Sep 1 13:04:46] DEBUG[1386]: chan_sip.c:8334 process_sdp: Processing session-level SDP o=1005 8010 8000 IN IP4 192.168.2.132... UNSUPPORTED. [Sep 1 13:04:46] DEBUG[1386]: chan_sip.c:8334 process_sdp: Processing session-level SDP s=SIP Call... UNSUPPORTED. [Sep 1 13:04:46] DEBUG[1386]: chan_sip.c:8334 process_sdp: Processing session-level S |
| 08:49.47 | DREMA | on the sip debug |
| 08:50.56 | DREMA | http://pastebin.com/EirLQg87 |
| 08:51.00 | DREMA | that |
| 08:51.01 | jpsharp | RTP/SAVP says something wants to negotiate encryption. |
| 08:51.21 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 08:52.10 | jpsharp | Yep. Your grandstream is trying to negotiate encryption but Asterisk is telling it that it has no idea what that's about. |
| 08:52.41 | DREMA | I found some problem like that on google. Something about requesting encryption but asterisk not supporting it |
| 08:52.45 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 08:53.03 | DREMA | Can't find any option about encryption in my Grandstream |
| 08:53.21 | jpsharp | Encryption, TLS, SRTP |
| 08:53.51 | DREMA | Let me check one more time |
| 08:54.58 | DREMA | I should mention it was working perfectly fine |
| 08:55.12 | DREMA | No changes made to the grandstrem nor the asterisk |
| 08:55.40 | jpsharp | Or you could go through the oh-so-fun-and-exciting evolution of configuring SRTP :) |
| 08:57.09 | DREMA | I would really like to do it that way to learn about it |
| 08:57.22 | DREMA | but is my work pbx |
| 08:57.28 | DREMA | cant play with it right now |
| 09:02.12 | DREMA | Maybe I'm blind but I can't fine anything about encryption besides SIP transport |
| 09:02.17 | DREMA | and its on UDP |
| 09:04.22 | *** join/#asterisk enoch (~unix@unaffiliated/enoch) |
| 09:04.23 | enoch | hi all |
| 09:04.33 | enoch | the best way to make an sms gateway? |
| 09:04.37 | DREMA | Yeah, I'm BLIND |
| 09:04.57 | DREMA | SRTP Mode: Enabled but not forced |
| 09:05.05 | DREMA | Could that be the problem? |
| 09:06.31 | DREMA | How do I forward a call between extensions from the Asterisk-CLI ? |
| 09:07.38 | DREMA | "ext 1005 dial to 1006" for example |
| 09:10.49 | DREMA | or how I place a call on another phone from the asterisk-cli |
| 09:18.34 | bulkorok | DREMA: originate |
| 09:20.18 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:15b7:451d:63fd:b1ae) |
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| 09:45.05 | *** join/#asterisk mathi (~Matthew@240.83-240-81.adsl-dyn.isp.belgacom.be) |
| 09:45.07 | mathi | hi |
| 09:46.49 | mathi | some functions are uppercase like ISNULL(), DB(), DB_DELETE(), ... and others are camel case like GotoIf(), Dial(), Set(), ... |
| 09:47.16 | mathi | why don't they all follow the same convention? is there a difference among these functions? |
| 09:47.53 | kaldemar | mathi: all functions are in upper case. GotoIf(), Dial() and Set() are not functions, they are applications. |
| 09:48.38 | mathi | I'm not sure I udnerstand the difference between an application and a function :( |
| 09:49.02 | tomodachi | a function is something you program yourselve, an application is a program made by someone else |
| 09:49.34 | kaldemar | mathi: exten => s,1,Application() |
| 09:49.53 | kaldemar | mathi: exten => s,n,Set(FUNCTION(field)=value) |
| 09:50.22 | reisi | when a sip channel is up, using speex, how can i Playback a ".spx" file? (currently the .sln file seems to be always selected) |
| 09:50.45 | kaldemar | mathi: you see? applications are something that are executed in every priority in an extension. functions are something that can be used inside applications. |
| 09:51.06 | kaldemar | mathi: functions cannot be used like exten => s,1,FUNCTION() |
| 09:51.55 | mathi | kaldemar, how can I use then DB_DELETE(test/count) |
| 09:58.40 | *** join/#asterisk kresp0 (~kresp0@213.37.150.59.dyn.user.ono.com) |
| 10:03.00 | v0lZy | hi all |
| 10:03.56 | *** join/#asterisk mquevedob (~mquevedob@201.217.51.220) |
| 10:04.19 | v0lZy | kaldemar: I noticed that when I put somone on hold (or switch to another line to make a second call) ... after 2 minutes of being on hold, asterisk hangs up the original caller (it seends some beeps to me before that). What option controls this hangup behaviour? |
| 10:05.21 | mquevedob | does anyone know if it is possible to configure SDP/connection information ? I need to set a different RTP IP address |
| 10:09.56 | *** join/#asterisk wonderworld (~ww@dsdf-4db55cc3.pool.mediaWays.net) |
| 10:12.45 | v0lZy | mathi: you want to DB_DELETE something' |
| 10:12.46 | v0lZy | ? |
| 10:13.08 | mathi | v0lZy, I was just wondering why DB_DELETE is a function and not an application |
| 10:13.41 | mathi | it was an application until version 1.4 |
| 10:15.17 | v0lZy | u use it like exten => n,NoOp(${DB_DELETE(field/value)}) |
| 10:16.00 | mathi | but didn't it make more sense using it as an applicatio, ? exten => Dbdel(field/value) ? |
| 10:16.12 | v0lZy | As to why its an application rather than a function, I dont know. But to me it makes more sense as an a function since its like Set ... sets unsets something |
| 10:16.41 | mathi | v0lZy, well to delete a family, you use an application |
| 10:16.42 | v0lZy | mathi: I agree, it would make more sense ... using it that way |
| 10:17.05 | mathi | v0lZy, maybe the reason is that an application cannot return anything ? |
| 10:17.31 | v0lZy | mathi: I honestly dont know. |
| 10:17.31 | mathi | as DB_DELETE returns the old value |
| 10:18.07 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-diechfgluynrcowh) |
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| 10:42.02 | kaldemar | mathi: the function can also used to retrieve a value from the database, like Set(var=${DB_DELETE(family/key)}) |
| 10:42.24 | mathi | kaldemar, yep I am just reading it here: https://issues.asterisk.org/jira/browse/ASTERISK-7025?page=com.atlassian.jira.plugin.system.issuetabpanels%3Aall-tabpanel |
| 10:43.09 | kaldemar | mathi: you should be reading it in CLI with "core show function DB_DELETE". |
| 10:43.41 | v0lZy | hm..... rtpholdtimeout ... |
| 10:43.49 | reisi | should sox input.wav -t raw -s -2 -c 1 -r 8000 output.sln produce something that should be playable by Playback? |
| 10:47.30 | kaldemar | reisi: yes. |
| 10:54.35 | *** join/#asterisk vlad_starkov (~vlad_star@ppp91-79-87-44.pppoe.mtu-net.ru) |
| 10:57.41 | v0lZy | kaldemar: I noticed that when I put somone on hold (or switch to another line to make a second call) ... after 2 minutes of being on hold, asterisk hangs up the original caller (it seends some beeps to me before that). What option controls this hangup behaviour? I dont have a rtpholdtimeout value specified in my spi.conf general section... is it 120sec by default, could that be the case? |
| 10:57.50 | v0lZy | (i have my rtptimeout set to 120 sec |
| 10:57.52 | v0lZy | ) |
| 10:58.11 | v0lZy | or at least i thought i did, i cant seem to find it now |
| 10:59.55 | *** part/#asterisk mirela666 (~Thunderbi@212.200.146.253) |
| 11:00.13 | *** join/#asterisk teff (~teff@client-80-1-162-43.bsh-bng-011.adsl.virginmedia.net) |
| 11:00.15 | *** join/#asterisk mirela666 (~Thunderbi@212.200.146.253) |
| 11:03.13 | v0lZy | is there a way to see what my current rtptimeout value is? |
| 11:03.20 | v0lZy | from asterisk CLI? |
| 11:03.32 | kaldemar | v0lZy: rtpholdtimeout is 0 by default, which means disabled. |
| 11:03.35 | kaldemar | "sip show settings" |
| 11:04.05 | reisi | kaldemar: is there documentation somewhere on how Playback() selects the file, and what file suffices are acceptable? |
| 11:05.15 | mirela666 | reisi: http://www.voip-info.org/ |
| 11:06.00 | v0lZy | hm |
| 11:06.04 | v0lZy | my rtptimeout = 0 |
| 11:06.12 | *** join/#asterisk gusto (~gusto@2001:a60:11ff:1200::42:4) |
| 11:06.19 | mirela666 | reisi: or here : https://wiki.asterisk.org/wiki/display/AST/Application_Playback |
| 11:06.29 | v0lZy | rtpholdtimeout also 0 |
| 11:06.36 | v0lZy | why do my calls drop when on hold for 120 sec? |
| 11:07.05 | *** join/#asterisk vlad_starkov (~vlad_star@ppp91-79-87-44.pppoe.mtu-net.ru) |
| 11:07.25 | reisi | mirela666: that does not quite cover the supported formats and their suffices? |
| 11:08.06 | kaldemar | reisi: "core show application Playback" don't waste your time reading voip-info. |
| 11:08.07 | mirela666 | reisi: than read about sound files |
| 11:08.07 | mirela666 | http://www.voip-info.org/wiki/view/Asterisk+sound+files |
| 11:11.14 | v0lZy | hm... |
| 11:11.29 | v0lZy | My calls seem to drop when on hold at exactly 120 seconds |
| 11:11.38 | v0lZy | this is probably some setting or lack of thereof |
| 11:12.02 | *** join/#asterisk darksk1ez (~mhb@fsf/member/darkskiez) |
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| 11:13.22 | v0lZy | if i set to verbose 9 |
| 11:13.33 | v0lZy | I just get that dial on the original caller exited non zero |
| 11:14.22 | v0lZy | my dial parameter is Dial(SIP/16,,Tt) |
| 11:14.34 | v0lZy | am i missing something.. ring time, something? |
| 11:16.03 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 11:16.03 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 11:19.57 | *** mode/#asterisk [+o mjordan] by ChanServ |
| 11:31.06 | Digweed | I've always used the IP address of my SIP proxy in the sip.conf but would like to use the DNS name. That does not seem to work however, I get this error: |
| 11:31.08 | Digweed | [Sep 4 14:52:56] WARNING[9623]: chan_sip.c:2921 create_addr: No such host: sip.myprovider.com |
| 11:31.08 | Digweed | [Sep 4 14:52:56] WARNING[9623]: chan_sip.c:7599 transmit_register: Probably a DNS error for registration to username@sip.myprovider.com, trying REGISTER again (after 20 seconds) |
| 11:31.23 | Digweed | I replaced the real address with an example. |
| 11:31.26 | Digweed | ;) |
| 11:32.11 | Digweed | This is the working configuration: register=>username:password:username@ipaddress:5060 |
| 11:32.18 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 11:32.29 | Digweed | This is not working : register=>username:password:username@sip.myprovider.com:5060 |
| 11:32.43 | Digweed | The machine is able to resolve the domain. |
| 11:36.15 | *** join/#asterisk italorossi (~textual@189.124.200.92) |
| 11:37.15 | Digweed | Anyone a clue? |
| 11:37.27 | Digweed | Asterisk 1.4.21.1 |
| 11:47.23 | *** join/#asterisk darksk1ez (~mhb@fsf/member/darkskiez) |
| 11:49.23 | Faustov | Digweed: well, is your resolver working? |
| 11:49.42 | Digweed | When I ping the domain it resolves to the correct IP. |
| 11:50.14 | Faustov | from the same box? |
| 11:50.18 | Digweed | Correct. |
| 11:50.28 | Digweed | I exit the Asterisk console and ping, resolves fine. |
| 11:51.22 | Faustov | incorrect settings in /etc/nsswitch.conf ? |
| 11:52.11 | Digweed | Hm, I have never edited that file. |
| 11:52.48 | Faustov | I am out of ideas |
| 11:53.03 | Digweed | Me as well hah :) |
| 11:53.23 | Faustov | you can jump on strace though, maybe it will spit some hints |
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| 12:14.21 | carrar | *Y*A*W*N* |
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| 12:34.58 | v0lZy | kaldemar, or anyone else, could session-expires= parameter be dropping the calls i have on hold after 2 min? |
| 12:36.00 | v0lZy | i have session-expires=120 in my sip.conf and everythings fine except when i put the caller on hold, he drops after 2 min |
| 12:40.03 | mirela666 | v0lZy: best way to find out is to try to modify and see what happens :D |
| 12:41.45 | v0lZy | i guess |
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| 12:45.54 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 12:52.30 | v0lZy | hm |
| 12:52.33 | v0lZy | apparently that is the case |
| 12:52.37 | v0lZy | i increased the number to 180 |
| 12:52.47 | v0lZy | i expect to be disconnected at 3 min |
| 12:52.52 | v0lZy | will see in 20 sec |
| 12:53.11 | v0lZy | yeah |
| 12:53.13 | v0lZy | i got disconnected |
| 12:53.31 | v0lZy | apparently session-expires parameter is causing people on hold to be disconnected after a certain amount of time |
| 12:53.39 | v0lZy | how can i prevent this? |
| 12:58.12 | *** join/#asterisk TheCompWiz (~TheCompWi@63.214.236.169) |
| 12:58.24 | [TK]D-Fender | v0lZy, http://www.google.ca/#hl=en&sclient=psy-ab&q=asterisk+disable+sip+session-timers&oq=asterisk+disable+SIP+se&gs_l=hp.3.0.0i30.1094.7897.0.9572.29.26.2.0.0.2.1118.15434.5-15j5j2.22.0...0.0...1c.2LNAxr0G-YI&pbx=1&bav=on.2,or.r_gc.r_pw.r_qf.&fp=7fed4507dd273c61&biw=1600&bih=883 |
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| 13:00.24 | TheCompWiz | Anyone know how to keep the manager timeout from ever timing out? |
| 13:00.48 | TheCompWiz | ... or why a connection would automatically terminate itself after a while. |
| 13:06.44 | WIMPy | There is no timeout on manager. |
| 13:10.00 | *** join/#asterisk bchia (~Adium@nat/digium/x-jdpznnwgfgofwxkr) |
| 13:14.42 | TheCompWiz | WIMPy: so... what is "writetimeout"? |
| 13:15.06 | v0lZy | [TK]D-Fender: I dont know exactly what u mean but thanks |
| 13:15.21 | v0lZy | I understand that session-timers have to do something with this |
| 13:15.49 | v0lZy | I deliberately used them to clean dead channels that arent bridged etc |
| 13:15.58 | v0lZy | but now apparently this becomes a problem |
| 13:18.00 | kaldemar | if you're rude enough to put people on hold for over two minutes, you deserve some issues. |
| 13:18.03 | [TK]D-Fender | v0lZy, What is there to not understand? You asked how to stop it. This is a TEN SECOND Google search. |
| 13:18.27 | mirela666 | lol kaldemar is right lol |
| 13:19.35 | v0lZy | I already googled... |
| 13:20.41 | v0lZy | kaldemar: well its not rude if its legit... i have someone on the phone that calls and wants to speak to someone. we dont know what office that person is in at the moment |
| 13:20.45 | [TK]D-Fender | v0lZy, yes and that exact search shows me the paramter value to set in the first page without even drilling the results. |
| 13:21.18 | v0lZy | session-timers=refuse |
| 13:21.33 | v0lZy | but that means it wont be cleaning up after itself, if 1 channel somehow stays up |
| 13:22.06 | WIMPy | TheCompWiz: As th sample states that's for clients that don't listen. |
| 13:22.25 | TheCompWiz | don't listen to what? |
| 13:22.28 | WIMPy | But there's no idle timeout. |
| 13:22.37 | WIMPy | To the data that is sent to them. |
| 13:22.39 | v0lZy | Point is that they have the person on hold until the reception locates them then transfers the call |
| 13:22.58 | v0lZy | takes a few minutes to locate some staff that tends to run around a relatively big building |
| 13:23.18 | TheCompWiz | grrr... then I'm back to square one. What would cause a connection to the manager interface to terminate immediately after a call is made. |
| 13:23.28 | *** join/#asterisk deo (~dnepangue@222.127.13.226) |
| 13:23.34 | *** part/#asterisk deo (~dnepangue@222.127.13.226) |
| 13:24.04 | WIMPy | TheCompWiz: Falure to process the data sent from Asterisk. |
| 13:24.08 | WIMPy | Or an evil firewall. |
| 13:24.28 | WIMPy | Firewalls may time out idle connections. |
| 13:25.09 | TheCompWiz | WIMPy: it stays connected continuously ... until a call is made. I can send all sorts of actions & get responses without a problem. I doubt that's a firewall. What sent by ami would REQUIRE a response? |
| 13:27.18 | leifmadsen | a PING probably |
| 13:27.35 | TheCompWiz | ping doesn't require a response... a ping is sent... and replied to by a ping. |
| 13:27.37 | TheCompWiz | *pong |
| 13:28.55 | TheCompWiz | and just flipping through the log... no pings/pongs. |
| 13:29.11 | TheCompWiz | buncha "Newexten" and such. |
| 13:29.27 | v0lZy | im looking at http://www.asterisk.org/doxygen/trunk/Config_sip.html [TK]D-Fender. Session-timers seems like a good thing to have. But why is this a problem with onhold stuff.. i mean why does it work ok when 2 lines talk and why is it a problem when someone is on hold, whats the difference here? |
| 13:29.58 | *** join/#asterisk dms (~dms@nat/digium/x-zstghetircyutqdy) |
| 13:34.40 | [TK]D-Fender | v0lZy, either disable them or set them to "arbitrarily large". Take your pick. |
| 13:35.05 | v0lZy | Yeah, i got that part |
| 13:35.17 | v0lZy | Is there another way to do cleanup then? |
| 13:35.31 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
| 13:35.37 | v0lZy | or some other parameter that exempts onhold stuff from session-timers ? |
| 13:36.08 | WIMPy | TheCompWiz: Sounds wrong. I guess you should consult wireshark. It could even be something going wrong within Asterisk. |
| 13:36.35 | TheCompWiz | WIMPy: I have the full dialog. I am leaning towards the asterisk problem.... /sigh |
| 13:36.38 | TheCompWiz | thanks anyway. |
| 13:41.41 | *** join/#asterisk vlad_starkov (~vlad_star@213.79.102.163) |
| 13:43.20 | v0lZy | gotta run |
| 13:44.21 | v0lZy | thanks for the help guys |
| 13:44.25 | v0lZy | c u tomorrow |
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| 14:13.22 | *** join/#asterisk Twitchnln (~Adium@adsl-184-36-49-212.asm.bellsouth.net) |
| 14:13.29 | Twitchnln | morning |
| 14:13.53 | Twitchnln | anyone know how to default a cisco/linksys spa303 from web interface? |
| 14:15.53 | *** join/#asterisk MaDiNfO_ (~madinfo@81.92.217.159) |
| 14:15.57 | MaDiNfO_ | hi |
| 14:16.17 | MaDiNfO_ | does asterisk suports cid lookup ? |
| 14:16.29 | MaDiNfO_ | on what file should i configure for that ? |
| 14:17.01 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 14:18.16 | leifmadsen | MaDiNfO_: cid lookup in what way? |
| 14:18.34 | MaDiNfO_ | web one... |
| 14:18.44 | leifmadsen | like a CNAM dip? that would be configured outside of asterisk -- likely triggered via CURL() function or AGI() application |
| 14:18.48 | MaDiNfO_ | somescript.php?number=[number] |
| 14:18.57 | leifmadsen | AGI() |
| 14:19.10 | MaDiNfO_ | any documentation for that ? |
| 14:19.12 | MaDiNfO_ | how to ? |
| 14:19.16 | leifmadsen | lots |
| 14:19.17 | MaDiNfO_ | or something that i can check out ? |
| 14:19.19 | leifmadsen | look for Asterisk AGI |
| 14:19.25 | leifmadsen | use the google |
| 14:22.11 | carrar | THE |
| 14:22.50 | MaDiNfO_ | :P |
| 14:23.25 | *** join/#asterisk hrolf (~manzoor@unaffiliated/hrolf) |
| 14:23.28 | hrolf | Hi #asterisk |
| 14:23.41 | TheCompWiz | I am not... I just like doritos. |
| 14:24.33 | *** join/#asterisk ks3 (ks3@2600:3c03::f03c:91ff:feae:97e3) |
| 14:25.20 | hrolf | I have a question regarding the Originate action. Can any one tell what I need to specify for the 'Channel' field if I need to dial the number 4647766 through DAHDI, something like DAHDI/1/4647766 etc? |
| 14:25.44 | hrolf | I cannot found any documentation explaining this. |
| 14:26.35 | [TK]D-Fender | hrolf, Exactly what you'd put in Dial() |
| 14:27.14 | leifmadsen | same string you'd use for Dial() |
| 14:27.36 | leifmadsen | perhaps you can show what you're attempting to do and what the output is giving you -- use pastebin please |
| 14:28.02 | hrolf | leifmadsen, [TK]D-Fender: What string would I use for Dial()? |
| 14:28.24 | [TK]D-Fender | hrolf, You ahve no idea how to dial normally in the dialplan with it in the first place? |
| 14:28.28 | hrolf | leifmadsen: I'm looking for any document describing the format. |
| 14:28.59 | hrolf | with version 1.6 |
| 14:29.16 | leifmadsen | 1.6 isn't a version, but it's basically the same as all versions of Asterisk |
| 14:29.37 | hrolf | leifmadsen: 1.6.2.22 |
| 14:29.52 | hrolf | leifmadsen: So isn't it documented? |
| 14:31.16 | leifmadsen | hrolf: you didn't read the dialplan basics chapter at asteriskdocs.org? |
| 14:31.17 | leifmadsen | http://asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-DP-Basics-SECT-3.html#asterisk-DP-Basics-SECT-3.3 |
| 14:35.37 | hrolf | leifmadsen: Nope, I learned (still am) by studying the already existing system we have. (I think I'll have to begin with the basics first.) Thanks for the link, it seems like it will answer my questions. |
| 14:36.00 | leifmadsen | hrolf: ya, documentation will be key... just looking at the system isn't going to be a huge help likely |
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| 15:15.30 | Katty | hi |
| 15:29.04 | carrar | Hi |
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| 16:09.36 | bmoraca_work | what would cause DPMA to not return the voicemail messages or folder lists for visual voicemail? |
| 16:15.34 | *** join/#asterisk deo (~dnepangue@112.198.79.94) |
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| 16:20.21 | bmoraca_work | ok, so it's a 1.2 issue |
| 16:21.46 | Qwell | pretty hard to say without knowing more |
| 16:22.12 | bmoraca_work | well, it works in 1.0.2 and not with 1.2 |
| 16:22.26 | bmoraca_work | no errors on the phone, dialplan reads the voicemail .txt file |
| 16:22.28 | bmoraca_work | weird |
| 16:22.42 | bmoraca_work | guess the sweet spot is 10.6 with 1.0.2 |
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| 16:33.02 | WIMPy | I've got an issue with sip peer configuration. Asterisk seems to use the name supplied in host= as the domain in the to field of the invite. Is it possible to change that? |
| 16:37.05 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v012-133.mobile.uci.edu) |
| 16:37.50 | [TK]D-Fender | WIMPy, fromdomain <- |
| 16:38.39 | WIMPy | That doesn't change the *to* domain. |
| 16:39.33 | [TK]D-Fender | Hrm.... |
| 16:40.06 | [TK]D-Fender | Not sure if there was a secondary "domain" in addition to "realm" |
| 16:40.11 | [TK]D-Fender | I'd check for those next |
| 16:40.39 | WIMPy | I didn't find anything. |
| 16:41.17 | WIMPy | I'm not sure how that could work syntactically, but could it be possible to put the domain in to the Dial()? |
| 16:42.06 | [TK]D-Fender | Not when using a proper ppeer entry... |
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| 17:10.27 | WIMPy | Oh, I didn't know there are Android smartphones with DECT (only). |
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| 17:16.49 | Katty | hi. |
| 17:16.59 | WIMPy | lo |
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| 17:25.27 | Nephfl | anybody familiar with monast and the Twisted framework? |
| 17:25.40 | Nephfl | I'm just trying to set it up and it isn't cooperating |
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| 17:50.21 | Qwell | WIMPy: no wifi or anything? |
| 17:52.20 | WIMPy | Yes, wifi, but no sort of PLMN support. |
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| 18:10.04 | autofsckk | hi |
| 18:10.24 | autofsckk | im trying to use at least 3 sip phones behind pfsense |
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| 18:17.37 | navaismo | can i tag everyone here I need a d40 phone |
| 18:19.56 | leifmadsen | no |
| 18:20.53 | navaismo | :( |
| 18:20.54 | *** join/#asterisk _Corey_ (~chatzilla@64.215.11.114) |
| 18:26.20 | autofsckk | i receive this [Sep 4 14:25:17] NOTICE[16295]: res_rtp_asterisk.c:2334 ast_rtp_read: Unknown RTP codec 126 received from '189.151.229.76:43194' |
| 18:27.14 | autofsckk | i can contact to somebody outside my lan connected to a vlan * server, i can hear but they can not hear me |
| 18:28.40 | MLNoah | I'm working with XMPP in Asterisk 11, ran into a behavior I didn't really expect from the documentation. The scenario I'm doing is this: end user sends (unsolicited) XMPP message to Asterisk, which processes it and sends back a request for further information to the end user, waiting for the response with JABBER_RECEIVE(). When that response comes back, it receives back into the call that |
| 18:28.41 | MLNoah | was waiting for it, but also starts off a new call (via sendtodialplan). Is that expected behavior? Is there something I can do to configure so it doesn't happen? |
| 18:35.49 | WIMPy | just shot an Astibank |
| 18:35.59 | *** part/#asterisk rrittgarn (~rrittgarn@75-150-221-196-Illinois.hfc.comcastbusiness.net) |
| 18:39.38 | ChannelZ | With a large caliber I hope |
| 18:46.56 | WIMPy | Not really. |
| 18:49.00 | navaismo | Throw through the window? |
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| 18:57.04 | adeel|work | is there a downside to enabling the usereqphone value in sip.conf? |
| 18:59.45 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 19:00.08 | jaytee | anyone know if there's a 3Com VCX irc channel? |
| 19:00.37 | navaismo | adeel|work, that is defined in the general section of the sip.conf but dont know if still using in new versions |
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| 19:02.01 | adeel|work | navaismo, yup, i know it's in the general section...but is there a downside of using it as the default is/was no |
| 19:04.06 | navaismo | no idea, never used it and documentation only says it add "phone" to the URI |
| 19:04.23 | adeel|work | well, it adds ';user=phone' |
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| 19:13.48 | acidfoo | many people here use sccp(skinny) ? -- just being curious |
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| 19:23.08 | TheCompWiz | why would anyone *choose* to use sccp? |
| 19:24.17 | acidfoo | I guess that's another question |
| 19:24.54 | TheCompWiz | I guess.. the answer is... "as few as possible" |
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| 20:42.36 | pigpen | Oooo...10.7.1. I am so excited. |
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| 20:49.34 | nny | odd question: I have one remote peer that is showing up as a PFC address on the same subnet as the local network. I have externip set as well as localnet (have also tired with it disabled). I have another remote peer who properly shows up as the remote public IP. Both have nat=yes in sip.conf. |
| 20:51.20 | pigpen | does anybody know if the dpma works with asterisk realtime? |
| 20:51.27 | pigpen | I would think so..but you never know |
| 20:52.36 | navaismo | pigpen, you need to define the user&password in the dpma and it will work |
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| 20:54.36 | nny | Odd, in spite of having nat=yes in sip.conf, the peer shows up as NAT N ---> 105/105 192.168.1.115 D N 5060 UNREACHABLE |
| 20:54.43 | nny | it also shows up as Nat Always in sip show peer 105 |
| 20:56.08 | nny | making some adjustments, this may be a localip delclaration issue, one sec |
| 20:56.15 | pigpen | navaismo, k, thnaks! |
| 20:56.21 | pigpen | geesh, I can't type. Thanks! |
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| 20:59.02 | nny | ahh got it. heh |
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| 21:18.06 | sarsaeol | remedial question: what makes some extensions register from 5060, but others register from ports like '13422 and 15114' |
| 21:18.10 | sarsaeol | i thought it was a NAT thing |
| 21:18.17 | sarsaeol | but am now not so sure |
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| 21:20.31 | Qwell | sarsaeol: Source port. It's unimportant. |
| 21:26.10 | sarsaeol | ty Qwell |
| 21:26.54 | leifmadsen | some clients let you change the source port |
| 21:27.14 | leifmadsen | which is important if you're running asterisk and the client on the same machine (which happens on development environments sometimes, like in the Asterisk training classes) |
| 21:30.50 | sarsaeol | leifmadsen: i'm just trying to track down an issue where i can only register one extension at a time from a certain public IP address… it seems the source port stuff really is unrelated |
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| 21:31.24 | leifmadsen | what is the symptom of being unable to register? |
| 21:31.28 | leifmadsen | is asterisk rejecting it, or what? |
| 21:31.57 | leifmadsen | certain firewalls/NATs can wreak havoc |
| 21:32.49 | sarsaeol | the 'bad' ip space registers from 5060, but another ip space where i can register many extensions comes from source ports like 13000 or 15000…symptoms are that the * box says the extension is unauthorized, but nothing in the /var/log/asterisk/full saying bad auth or bad acl |
| 21:33.25 | sarsaeol | im having network engineers make sure there is nothing screwy in firewall |
| 21:33.48 | sarsaeol | but its weird that the request would ever make it to the pbx if the firewall was bad |
| 21:33.59 | sarsaeol | so i might be chasing my tail on that |
| 21:39.12 | jpsharp | Unless the firewall has a SIP rewriting system. |
| 21:40.00 | sarsaeol | well hopefully the network guys will come back to me with something. |
| 21:40.35 | sarsaeol | because i certainly cannot find any PBX settting which would limit the amt of registrations |
| 21:40.51 | leifmadsen | right -- turn off the ALG sip stuff |
| 21:40.57 | leifmadsen | it won't be on asterisk |
| 21:41.22 | leifmadsen | if all phones are coming from 5060, then the firewall/nat system is doing something to alter the packets likely |
| 21:41.42 | leifmadsen | SIP ALG stuff almost always screws stuff up |
| 21:41.51 | sarsaeol | aint that the truth |
| 21:41.55 | sarsaeol | :-/ |
| 21:42.05 | leifmadsen | turn it off, and it'll likely just magically start working |
| 21:42.35 | sarsaeol | word… thanks for the insight… ill try to update yall with what worked |
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| 23:15.23 | lorsungcu | I've still got an issue with odic making asterisk hang |
| 23:15.28 | lorsungcu | has anyone ever seen that? |
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| 23:34.51 | wheel | I am trying to create an ACD queue, using a SIP peer as the member. The peer is a SIP gateway, with agents connected, but appears as a single 'trunk group' to Asterisk. Is it possible to place a channel limit on that peer, so that only 10 calls can be active? As a call ends, the next caller in queue would be passed to the peer, with never more than 10 calls at any given time? |
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| 23:38.41 | paulc | wheel: so you send calls to a single destination and it distributes to 10 phones at the other end? |
| 23:38.54 | wheel | right |
| 23:39.20 | paulc | Hmm.. seems kinda "wrong".. because don't you want Asterisk to distribute calls to agents, so you can get stats on agent performance, know who's available versus not, etc? |
| 23:40.01 | paulc | ideally each "agent" (in the ACD sense of the word) is known to Asterisk, so it can distribute calls efficiently (and give meaningful stats) |
| 23:42.13 | wheel | I agree, but in this case, the phones are hidden (don't belong to me). I was trying to build pseudo channels, i.e. SIP-peer-000001, SIP-peer-00002, etc., and treat the channel like an agent. That way, I would have device state even though I don't have agent stats. |
| 23:42.33 | wheel | The gateway is connected to a PBX which provides the actual agent stats. |