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| 01:03.05 | KevinRoberts | Greetings all |
| 01:19.48 | ChannelZ | ahoy |
| 01:31.25 | KevinRoberts | I 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. |
| 01:32.09 | WIMPy | Did you read the |
| 01:32.11 | WIMPy | ~book |
| 01:32.11 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 01:32.13 | WIMPy | ? |
| 01:38.04 | KevinRoberts | Ok 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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| 02:17.31 | Penguin | kevinroberts: 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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| 02:21.02 | ChannelZ | ~primer |
| 02:21.02 | infobot | i guess primer is http://burner.com/asterisk-primer |
| 02:29.41 | ChannelZ | wanders off to find some dinner |
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| 09:18.07 | KevinRoberts | set view |
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| 12:11.50 | KevinRoberts | Hello, 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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| 12:58.31 | Neo` | hi al |
| 12:58.33 | Neo` | *all |
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| 13:08.25 | KevinRoberts | Hi |
| 13:08.58 | Neo` | how are you? ) |
| 13:09.06 | KevinRoberts | doing good and you? |
| 13:09.18 | Neo` | me too, thx ) |
| 13:10.35 | KevinRoberts | I 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 |
| 13:13.43 | KevinRoberts | I 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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| 13:33.28 | hluesea | hello, 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 |
| 13:34.20 | hluesea | when 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 |
| 13:35.19 | hluesea | how can i solve this issue ? have any body experiences before ? |
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| 13:57.43 | file | moo |
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| 14:08.03 | ertyui | hi |
| 14:08.09 | ertyui | iu can't make asterisk |
| 14:08.34 | ertyui | i got this pseudo error : The configure script must be executed before running 'make' |
| 14:09.11 | hluesea | did u run this command before make = ./configure |
| 14:09.30 | ertyui | <PROTECTED> |
| 14:09.56 | ertyui | well i have tried several time ./configure before make |
| 14:10.06 | ertyui | not solving at all |
| 14:10.39 | WIMPy | Sounds like configure fails. What does it give you? |
| 14:11.14 | ertyui | sounds like configure works for me without any error |
| 14:11.27 | ertyui | just only make not working |
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| 14:39.53 | hluesea | anybody have experienced about 2 different nic and static ips binding for 1 is sip and other is iax ? |
| 14:41.09 | [TK]D-Fender | set the bindport in each |
| 14:41.57 | hluesea | i 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-Fender | Then you have a privacy flag set on the incoming call |
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| 14:43.38 | hluesea | do you mean set off all flags to solve issue ? |
| 14:43.51 | [TK]D-Fender | core show function CALLERPRES |
| 14:44.00 | [TK]D-Fender | And lok at your peer |
| 14:44.49 | hluesea | ok i am trying it know ;) thanks |
| 14:50.40 | hluesea | i 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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| 18:42.27 | asteriskfan | Anyone 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-Fender | asteriskfan: yes |
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| 19:04.34 | asteriskfan | @[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-Fender | asteriskfan: 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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| 19:42.24 | Apteryx | Hello! 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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| 19:52.45 | Apteryx | Seems like I have to set the autofallthrough optionà |
| 19:52.48 | Apteryx | ? |
| 19:56.12 | Apteryx | Seems to have fixed it :) |
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| 20:34.29 | Apteryx | Hi! 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? |
| 20:34.54 | Apteryx | I've specifically told it not to using direct_media=no in my endpoints definitions (pjsip.conf). |
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| 20:54.55 | Apteryx | Another question: can the external_media_address be set to some domain name like: sip.mydomain.com, or does it only accept IP values? |
| 21:48.31 | Apteryx | Answer 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 |
| 21:55.02 | Apteryx | The 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. |
| 21:55.37 | WIMPy | wonders what that is so long. |
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| 22:13.04 | KevinRoberts | Hi all |
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| 22:56.21 | ChannelZ | ahoy |
| 22:57.30 | [TK]D-Fender | chips |
| 22:59.54 | ChannelZ | mmmm |
| 23:02.54 | KevinRoberts | channelz do you have a few? I have a few questions about the asterisk primer |
| 23:04.20 | Penguin | Lots of other people here, too. |
| 23:05.18 | ChannelZ | Yes |
| 23:05.30 | KevinRoberts | Yeah 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 |
| 23:05.32 | ChannelZ | Your problems are probably NAT/firewall related |
| 23:05.54 | ChannelZ | First, is asterisk behind NAT? |
| 23:06.15 | KevinRoberts | Know cause am doing through a vps server |
| 23:06.18 | ChannelZ | (or a firewall?) |
| 23:06.47 | ChannelZ | So your asterisk box has a public IP and no ports are being blocked? |
| 23:06.59 | KevinRoberts | know it has a public Ip yes |
| 23:07.15 | KevinRoberts | know ports are being blocked |
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| 23:07.34 | ChannelZ | OK. Then the device you're trying to test, is it behind NAT? |
| 23:07.44 | Penguin | The phone could also be behind a NAT and the configuration for the phone is not set to nat=yes. |
| 23:08.10 | KevinRoberts | Know. 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 |
| 23:08.41 | ChannelZ | So the softphone is on a computer with a public IP directly connected to the internet? Unlikely |
| 23:09.43 | ChannelZ | If you're at home on a cable modem or DSL you are almost certainly behind NAT |
| 23:10.15 | KevinRoberts | The 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? |
| 23:10.34 | ChannelZ | Yes and no |
| 23:10.56 | Penguin | You have to configure your peer in sip.conf for proper operation behind a NAT when the device is behind a NAT. |
| 23:11.02 | ChannelZ | In sip.conf for the peer, set nat=auto_force_rport,comedia |
| 23:12.32 | KevinRoberts | would that be after the [general] section? and I am on a wireless router not a direct calbel modom |
| 23:12.48 | KevinRoberts | cable |
| 23:13.58 | ChannelZ | It would be under the [whatever] section, where [whatever] is the device you're configuring |
| 23:14.35 | KevinRoberts | so it would be under the [Testphone-A] section |
| 23:15.00 | ChannelZ | (though you could set it in [general] since 90% of the time your remote peers are behind NAT firewalls) |
| 23:15.21 | KevinRoberts | Ok will try that and see what happens |
| 23:20.59 | KevinRoberts | Ok that was the problem |
| 23:21.32 | KevinRoberts | now i noticed you didn't go through how to set up the voicemail thing |
| 23:21.51 | Penguin | It's in the book. |
| 23:21.55 | Penguin | ~book |
| 23:21.55 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 23:22.58 | KevinRoberts | Ok will take a look. |
| 23:23.23 | ChannelZ | Yeah it's not a complete setup guide, it's a primer to take away the scarriness of getting started |
| 23:23.56 | ChannelZ | I 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. |
| 23:24.39 | ChannelZ | But now that you have the basics, other information you find on the net should hopefully make more sense. |
| 23:25.47 | KevinRoberts | is it safe to keep that nat=auto_force_rport,comedia in the config or what just makeing sure |
| 23:27.05 | ChannelZ | Yeah |
| 23:27.23 | ChannelZ | If you have peers later which aren't behind NAT, you can set them to nat=no individually. Or do the opposite. |
| 23:34.10 | KevinRoberts | Ok good I am happy I got somewhere with this you did a good job on that guide |
| 23:35.09 | ChannelZ | Thanks, glad you got it running. Sometimes NAT issues can be torture with SIP |
| 23:44.50 | KevinRoberts | Now with the book I know it is going to be refering to phone device hardware witch I don't need. |
| 23:45.38 | ChannelZ | Well that's just one section. The book covers a bunch of topics you probably don't care about. |
| 23:46.10 | KevinRoberts | does the book cover the c onfbridge application? |
| 23:52.28 | KevinRoberts | here 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. |
| 23:53.33 | ChannelZ | I think ConfBridge came after. |
| 23:53.54 | ChannelZ | Or rather ConfBridge appeared in 1.8 (I think) but the first version of it was very basic. |
| 23:56.26 | *** join/#asterisk tristero (~al.f.zero@unaffiliated/transfinite) |
| 23:57.55 | KevinRoberts | do you know of a guide that will talk about the conf bridge application? cause I plan to have a ext for a conference |
| 23:58.51 | KevinRoberts | ~book |
| 23:58.51 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |