IRC log for #asterisk on 20140928

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01:03.05KevinRobertsGreetings all
01:19.48ChannelZahoy
01:31.25KevinRobertsI would like some help I am a blind user of asterisk, and I want to learn how to configure it with just the config files but I have looked at the example files, but I am trying to to make make sence. I want to have a conference ext, also a way for users to have their own exts and maybe some other stuff but mainly the conf and user ext and voicemail.
01:32.09WIMPyDid you read the
01:32.11WIMPy~book
01:32.11infobotAsterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook
01:32.13WIMPy?
01:38.04KevinRobertsOk first of all I am not useing real hardware so I will not need the hardware software all of this is being done through a virtual private server by the way all internet based.
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02:17.31Penguinkevinroberts: That doesn't really matter.  You can use strictly VoIP from your Internet-based server.  You'll probably want to use SIP.  Read the book and learn about configuring devices and then extensions.
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02:21.02ChannelZ~primer
02:21.02infoboti guess primer is http://burner.com/asterisk-primer
02:29.41ChannelZwanders off to find some dinner
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09:18.07KevinRobertsset view
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12:11.50KevinRobertsHello, I have a question, I was following the guide at http://burner.com/asterisk-primer/ but rang in to a problem I can dial my ext 555 and I see that it is doing the commands in the asterisk consal but am n ot hereing any audio I tryed with two different softphones one on my Iphone, and one on my computer.
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12:58.31Neo`hi al
12:58.33Neo`*all
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13:08.25KevinRobertsHi
13:08.58Neo`how are you? )
13:09.06KevinRobertsdoing good and you?
13:09.18Neo`me too, thx )
13:10.35KevinRobertsI am wondering you might be able to help me out here I am playing with a asterisk system am playing with and I just installed. But I am not hereing any audio from the system at all and I am doing this over a vps server so I do not have access to the local computer that the asterisk is running on so am testing it through softphones
13:13.43KevinRobertsI was following the guide on http://burner.com/asterisk-primer and was on the part of makeing a dialplan got it the thing to see the ext and I see it doin the commands through the playback, but know audio at all
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13:33.28hlueseahello, i need some help about 2 nic and 2 different static ip address issue. I have configured on Centos and each nic get their own static - subnet correctly and ping correctly to own side blocks. Actually 1 interface is connected directly voip vendor with their own dedicated g.shdsl line and other nic is intent access i want to send calls from internet and connect to vendor's account
13:34.20hlueseawhen i configured as iax soft phone and registered asterisk then try to call sip with that vendor line i got retransmission errors and i have seen Anonymous on FROM field on sip dialogue
13:35.19hlueseahow can i solve this issue ? have any body experiences before  ?
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14:08.03ertyuihi
14:08.09ertyuiiu can't make asterisk
14:08.34ertyuii got this pseudo error : The configure script must be executed before running 'make'
14:09.11hlueseadid u run this command before make = ./configure
14:09.30ertyui<PROTECTED>
14:09.56ertyuiwell i have tried several time ./configure before make
14:10.06ertyuinot solving at all
14:10.39WIMPySounds like configure fails. What does it give you?
14:11.14ertyuisounds like configure works for me without any error
14:11.27ertyuijust only make not working
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14:39.53hlueseaanybody have experienced about 2 different nic and static ips binding for 1 is sip and other is iax ?
14:41.09[TK]D-Fenderset the bindport in each
14:41.57hlueseai have configured like that and iax user registered but i have failed to send calls from iax user to sip provider. When i send calls From field seen Anonymous in sip trace
14:43.00[TK]D-FenderThen you have a privacy flag set on the incoming call
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14:43.38hlueseado you mean set off all flags to solve issue ?
14:43.51[TK]D-Fendercore show function CALLERPRES
14:44.00[TK]D-FenderAnd lok at your peer
14:44.49hlueseaok i am trying it know ;) thanks
14:50.40hlueseai got below like trace now, xxxxxxxx is my callerid for this line that is correct but vendor side gave sip reason 486 ::  From: "Anonymous" <sip:xxxxxxxxxxx@anonymous.invalid>;tag=as401c2131
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18:42.27asteriskfanAnyone work with extensions.conf in realtime? It appears I can't refernce dialplan context directly in the database. I need to create a switch statement in extensions.conf for every context created in the database?
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19:00.42[TK]D-Fenderasteriskfan: yes
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19:04.34asteriskfan@[TK]D-Fender Sounds like my application will have to update the extensions.conf file directly and reload the dialplan as new users are added.
19:05.44[TK]D-Fenderasteriskfan: Depends on what you actually require.  Because "users" is  not a "thing".  If each DEVICE you define needs separate contexts, etc, that's another matter.  You should reconsider how you lay out your dialplan.
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19:42.24ApteryxHello! I noticed the conversation is not always terminated automatically when one of the party hangs up. I'm using a simple Dial command in my extensions.conf. Should I explicitely define a Hangup() for every extension? This happens using chan_pjsip, Bria (iPhone) and Jitsi (Ubuntu).
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19:52.45ApteryxSeems like I have to set the autofallthrough optionÉ
19:52.48Apteryx?
19:56.12ApteryxSeems to have fixed it :)
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20:34.29ApteryxHi! Could someone tell me what "switching from simple_bridge technology to native_rtp" means? Is Asterisk trying to connect directly (without relaying) the RTP streams between my endpoints?
20:34.54ApteryxI've specifically told it not to using direct_media=no in my endpoints definitions (pjsip.conf).
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20:54.55ApteryxAnother question: can the external_media_address be set to some domain name like: sip.mydomain.com, or does it only accept IP values?
21:48.31ApteryxAnswer to my previous question is: No. And that caused my issue without reporting anything wrong about not accepting domain names as valid values for external_media_address in pjsip.conf
21:55.02ApteryxThe good news about it is that some fine folk already wrote a 70 lines bash script to refresh the IP values of pjsip.conf or sip.conf :). See https://github.com/hayesey/asterisk-externhost.
21:55.37WIMPywonders what that is so long.
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22:13.04KevinRobertsHi all
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22:56.21ChannelZahoy
22:57.30[TK]D-Fenderchips
22:59.54ChannelZmmmm
23:02.54KevinRobertschannelz do you have a few? I have  a few questions about the asterisk primer
23:04.20PenguinLots of other people here, too.
23:05.18ChannelZYes
23:05.30KevinRobertsYeah I know but am haveing a audio problem with the system to where when I dial a ext am not anything through the playback application I see through the cli that it is doing the commands, but am not hereing what it is doing
23:05.32ChannelZYour problems are probably NAT/firewall related
23:05.54ChannelZFirst, is asterisk behind NAT?
23:06.15KevinRobertsKnow cause am doing through a vps server
23:06.18ChannelZ(or a firewall?)
23:06.47ChannelZSo your asterisk box has a public IP and no ports are being blocked?
23:06.59KevinRobertsknow it has a public Ip yes
23:07.15KevinRobertsknow ports are being blocked
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23:07.34ChannelZOK. Then the device you're trying to test, is it behind NAT?
23:07.44PenguinThe phone could also be behind a NAT and the configuration for the phone is not set to nat=yes.
23:08.10KevinRobertsKnow. it is just a softphone I tryed it on my computer and my Iphone it will connect just fine but I do not here any audio
23:08.41ChannelZSo the softphone is on a computer with a public IP directly connected to the internet?  Unlikely
23:09.43ChannelZIf you're at home on a cable modem or DSL you are almost certainly behind NAT
23:10.15KevinRobertsThe softphone is connected to the internet yes and it is on my local network but I made sure nothing is being blocked if stuff was blocked, it wouldn't connect right?
23:10.34ChannelZYes and no
23:10.56PenguinYou have to configure your peer in sip.conf for proper operation behind a NAT when the device is behind a NAT.
23:11.02ChannelZIn sip.conf for the peer, set nat=auto_force_rport,comedia
23:12.32KevinRobertswould that be after the [general] section? and I am on a wireless router not a direct calbel modom
23:12.48KevinRobertscable
23:13.58ChannelZIt would be under the [whatever] section, where [whatever] is the device you're configuring
23:14.35KevinRobertsso it would be under the [Testphone-A] section
23:15.00ChannelZ(though you could set it in [general] since 90% of the time your remote peers are behind NAT firewalls)
23:15.21KevinRobertsOk will try that and see what happens
23:20.59KevinRobertsOk that was the problem
23:21.32KevinRobertsnow i noticed you didn't go through how to set up the voicemail thing
23:21.51PenguinIt's in the book.
23:21.55Penguin~book
23:21.55infobotAsterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook
23:22.58KevinRobertsOk will take a look.
23:23.23ChannelZYeah it's not a complete setup guide, it's a primer to take away the scarriness of getting started
23:23.56ChannelZI will write more sections in the future to cover some things (like NAT setup), but I've just not had a lot of time lately.
23:24.39ChannelZBut now that you have the basics, other information you find on the net should hopefully make more sense.
23:25.47KevinRobertsis it safe to keep that nat=auto_force_rport,comedia in the config or what just makeing sure
23:27.05ChannelZYeah
23:27.23ChannelZIf you have peers later which aren't behind NAT, you can set them to nat=no individually.  Or do the opposite.
23:34.10KevinRobertsOk good I am happy I got somewhere with this you did a good job on that guide
23:35.09ChannelZThanks, glad you got it running. Sometimes NAT issues can be torture with SIP
23:44.50KevinRobertsNow with the book I know it is going to be refering to phone device hardware witch I don't need.
23:45.38ChannelZWell that's just one section. The book covers a bunch of topics you probably don't care about.
23:46.10KevinRobertsdoes the book cover the c onfbridge application?
23:52.28KevinRobertshere is my plan I have I plan to have people answer calls through sip but I also want a number to where people who do not have a softphone tobe able to call with a normal phone but yet intern it will connect them to the sip clients.
23:53.33ChannelZI think ConfBridge came after.
23:53.54ChannelZOr rather ConfBridge appeared in 1.8 (I think) but the first version of it was very basic.
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23:57.55KevinRobertsdo you know of a guide that will talk about the conf bridge application? cause I plan to have a ext for a conference
23:58.51KevinRoberts~book
23:58.51infobotAsterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook

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