Asterisk Tutorial 05 - Asterisk PBX SIP Phone Peers [english]
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- เผยแพร่เมื่อ 14 ต.ค. 2024
- Welcome to episode of 5 of our Introducing Asterisk video tutorials. Today's topic covers how to add and register SIP peers to your Asterisk services which is an essential step in building your Asterisk VoIP Server.
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You guys are the best!!! Mathias, your explications are so acurate and complete. Very comprehensive step by step tutorials!!!
thanks guys, I've been reading a lot of tutorials, but none of them explain the procedures well. I finally it working and making calls!. Thanks a lot!
+Hamilton Jimenez Vasquez glad we could help out!
Could you please help me setting this up
Thank you guys for en best tutorial about VoIP tech. I do some private research regard connectivity cp 6900 from Cisco and Asterisk 16.12.0. The question is: assume cp 6941 has a SIP 9.4.1.3.SR3 firmware. For connection with Asterisk, I need to upload a new sepmac.cnf.xml. If this xml file configured manually correctly, for upload purposes can I use just FTTP server (e.g. tftpd64), right? Or I need to install CUCM from Cisco?
Welcome to the latest instalment from the VoIP Guys Introducing #Asterisk series. In today's episode we cover how to add #SIP devices to your asterisk services.
th-cam.com/video/3UEBclNDDUE/w-d-xo.html
Great Channel, thank you, please continue recording in English for more wide public
Hi Giorgi, do not worry - we will continue in English. Our next Asterisk Video will be online next week. Should you have any topics you would like us to cover, then let us know by replying here.
pascom Netzwerktechnik Hello, It will be great if you record CAPsMAN video in English and also to dedicate several videos about how to build large wireless networks with multiple access points, roaming functionality and central control based on Mikrotik.
thank you in advance
Giorgi Bakhtadze Hey Giorgi, just wanted to let you know that while I was filming our next video, I had a chat with Mathias and in answer to your question - yes we can do some tutorials on MikroTik and CAPsMAN. We will get round to them in the near future - so watch this space.
Thank you, it will be great
Thanks. These videos were so useful!
Hi Wayne, glad you find them useful. The next video will be online next week! Just wondering - do you have any topics you would like us to cover?
pascom Netzwerktechnik Any topics on Asterisk would be great. Thanks again
Can you provide a link to the softphone that you are using, please?
This is a great. But i have a question , in sip.conf file you wrote :g/^\s*;/d this comment and after that which keyboard need press in order to entering sip peers configuration, i can not entered. I dont understand this way.
This is a very good online school. Could you please make a video on how to register to a SIP provider? Many thanks in advance.
Hi rey talattad , thank you for your kind feedback. No question about that, we will be making a tutorial on integrating SIP providers.
We are having a short summer break at the moment and will be back in a few weeks or so when we start filming more video, so if you have any more topics you would like us to cover, then let us know and we will get round to them.
All the best,
The VoIP Guys
Hi Mathias, i am working on a sip project presently, and i really love the videos. How can i download the telephone soft phone you are using for Ubuntu. I already have Zoiper but wish to simulate the whole traffic on one PC. Thanks alot
hi some of these commands are not available on asterisk 17.5.1
The 'sip show peers' command no longer works in version 13 of Asterisk - can you provide me with the replacement?
I've checked and not sure - is the replacement in this later version 'module show like chan_sip.so' ?
Also 'sip reload' command is no longer working in version 13 - can you update the command for that as well please?
So far - excellent video's - really good explanations - but I've installed Xlite on another PC - same network.
My problem is that my router is the DHCP server - and therefore have not set Asterisk as DHCP server.
So the Xlite has a static IP address - and I've put that IP address in the 'sip.conf' - so the line reads host=172.28.xxx.xxx
When I try to connect I get sip error 408 - authentication error - and it will not show in >CLI
What am I doing wrong? - thanks
Hello thanks for the videos they are great! But I have 1 doubt: what if you don't have a hardware phone at the moment to log in that interfaces where you add the peers accounts?? is there any alternative?
I been trying to fin a tutorial about connecting two asterisk pbx how would I go about that any episode that mentions that?
"why are we using a softphone?", "because it's free". ^_^
Hello,
Thanks for the videos.
I have installed asterisk server where my softphone clients are registered. I am making a call to another server.I have configured the destination server as a peer however when I re-load the file I get an error
ERROR[2456]: chan_sip.c:4263 __sip_reliable_xmit: Serious Network Trouble; __sip_xmit returns error for pkt data
Any Suggestions ?
Hi VoIP Guys. Trying to integrate Avaya SIP trunk with asterisk to use it just as a voicemail server.
Should I have all the extensions in sip.conf and also in voicemail.conf?
Also when I dial from Avaya to a sip trunk in asterisk I wish to get the "from" header and pass it to the correct peer, because Avaya always dial the same number to go to the SIP trunk, so it always have the same callerID when reaches the asterisk. Any clue?
Is there any GUI Available for Asterisk ? any video about it ? this complex thing should have a gui.
Hey +daDfarSpanTaistheKin,
If you're looking for a system that is Asterisk based, but has a GUI available then try out our mobydick phone system. You can download our Free community edition from our website (www.pascom.net) and clicking on download. In order to set up your mobydick system I would also recommend using our mobydick school videos:
th-cam.com/video/laF0s-alisk/w-d-xo.html
as well as signing up to our community forum (community.pascom.net) where free support is available!
All the best,
The VoIP Guys
Great series
Hello Pascom ,
this videos are great . I have question when i write sip show peers status of agent doesn't appear and write unmonitored
+بلال أبو العلا Hi, thank you! unmonitored means that the sip peer is not qualified. Maybe there is a "qualify=no" in your sip.conf
Great tutorial. Please find this link i have created asterisk sip peer by watching your tutorial
hi please can you provide a link to asterisk 11 version you used for this tutorial.thanks
Thanks so much !
You're welcome!
Hi Guys Please i have ip phone "ipecs lip 8002E" and i want to connect it with the PBX please Help me how can i do that
Im getting an NOTICE : [2016-03-31 17:20:08] NOTICE[1820][C-0000001a]: chan_sip.c:25865 handle_request_invite: Call from 'person1' (192.168.0.3:42513) to extension '100' rejected because extension not found in context 'from-sip-external'.
(my context is also named as 'phones')
And this just happens when I call from person1 to person2. Person2 to person1 works fine.
My peers are like this:
Name/username Host Dyn Forcerport Comedia ACL Port Status Description
person1/person1 192.168.0.3 D Yes Yes 42513 OK (14 ms)
person2/perso2 192.168.0.6 D Auto (No) No 48663 OK (5 ms)
Do you have any idea why this is hapenning?
Anyway, really thanks to these tutorials.
Hi There, I am constantly getting No such command "sip show peers". Can someone please support me?
Update the videos to showcase pjsip on asterisk 16
@12:31 when i reload sip i get below error
asterisk*CLI> sip reload
[Oct 30 00:59:07] ERROR[3055]: netsock2.c:271 ast_sockaddr_resolve: getaddrinfo("asterisk", "(null)", ...): Name or service not known
[Oct 30 00:59:07] WARNING[3055]: acl.c:833 resolve_first: Unable to lookup 'asterisk'
asterisk*CLI>
i changed hostname after installaiton of asterisk from generic installation name (which now i do not remember)
+daDfarSpanTaistheKin Your DNS is not working.
whats the softphone are you use???
Hey +Dy Monteiro, for the purposes of these tutorials we are using "Telephone" which is available to download from the mac App Store. Other alternatives include X-lite or Bria.
Or even better simply use "mobydick" as it comes complete with a fully integrated softphone ;-)
thanks..!! now how to conect my phone to asterisk??
and when i do the tutorial 04 my virtual machine lost the internet Connection.!!
Exclent Channel ... And Exclent Tutorial... So Good Guys...
Hey +Dy Monteiro, tutorials 5 and 6 cover SIP peers and dial plans. Regarding your VM losing internet connection, did you check whether your DNS server is working properly?
ok. thanks guys...
NIICE
X-Lite is not platform independent, It doesn't work on Linux
echo "" > /etc/asterisk/sip.conf works better for me... to empty the file.
Hopeless company, with no support or contact!
Hi Mark,
I am sorry that you have experienced some support issues. Quick question, are these issues related to Asterisk or to our pascom phone system? If they are related to a pascom solution, firstly let me apologise again and please feel free to contact us via our website: www.pascom.net where you will find plenty of contact and support options available. For starters, feel free to try our forum: www.pascom.net/forum/ and depending on your licensing, you can benefit from having access to our support team via our support portal.
Yours Sincerely,
your pascom team.