How to establish SIP connection, when SIP-proxy is required?

Posted by LA_ on Super User See other posts from Super User or by LA_
Published on 2012-11-10T17:53:03Z Indexed on 2012/11/16 23:03 UTC
Read the original article Hit count: 290

Filed under:

voip

|

asterisk

I have

Asterisk/1.8.13.1
Asterisk GUI-version : SVN--r

Yes, quite old one, but I can not update it since this is installed on my Synology NAS. NAS is connected to internet thru router Asus RT-N16.

I should use the following data to connect to the server:

Auth name – 7499952XXXX
User name/User ID/Display Name – nickname
Authorization user name - [email protected]
Domain - sip.beeline.ru
SIP proxy server - msk.sip.beeline.ru

I've also found the following string:

[email protected]:password:[email protected]@msk.sip.beeline.ru:5060/7499952XXXX

I've tested the parameters on my PC thru X-Lite and it works well (so, assume there is no any problem with the router, no need to do anything with router's NAS settings).

But since I am quite new to Asterisk, I can not understand where to input all these data. Asterisk GUI doesn't have fields for proxy:

Can somebody please help me with step-by-step instruction? Thank you in advance!

© Super User or respective owner

Related posts about voip

pfsense multi-site VPN VOIP deployment

as seen on Server Fault - Search for 'Server Fault'
have main office pfsense firewall configured like this: local networks WAN - internet LAN - local network VOIP - IP phones need to connect remote offices (multi-users) and single remote users (from home) use IPSEC or OpenVPN to build "permanent" automatically connecting tunnels from remote location… >>> More
pfsense multi-site VPN VOIP deployment

as seen on Server Fault - Search for 'Server Fault'
have main office pfsense firewall configured like this: local networks WAN - internet LAN - local network VOIP - IP phones need to connect remote offices (multi-users) and single remote users (from home) use IPSEC or OpenVPN to build "permanent" automatically connecting tunnels from remote location… >>> More
How to implement VOIP + SIP in iPhone?

as seen on Stack Overflow - Search for 'Stack Overflow'
Hello all, I want to develop a VOIP application for iPhone . But I don't know the basics of VOIP concepts and also If there are any sources or library available that can I use in my application . So If anyone can provide me VOIP learning resources or library or anything that is useful in terms… >>> More
Simplest VoIP Solution Yet?

as seen on Internet.com - Search for 'Internet.com'
MyPhone2Go is fully contained on a portable USB flash drive, can be used with just about any Windows computer. >>> More
Study: Mobile VoIP Explosion on the Way

as seen on Internet.com - Search for 'Internet.com'
Recent research from InStat predicts there will be nearly 300 million active mobile VoIP users by end of 2013. >>> More

Related posts about asterisk

How to send REGISTER request periodically from my SIP client to Asterisk server using Asterisk Manag

as seen on Stack Overflow - Search for 'Stack Overflow'
Hi, I am using Asterisk 1.4 server and I have created a desktop client using the Asterisk.NET Library. I am able to log into the AMI (as a manager) using Asterisk.Net, but I cannot find a way to send the REGISTER command using the AMI, to share my client's location information with the server. I… >>> More
asterisk : Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?)

as seen on Stack Overflow - Search for 'Stack Overflow'
Hi all, I am learning asterisk. After I have installed asterisk I have tried to connect with it using asterisk -rvvvvc. But it gave me an following error message. Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) How can I solve this issue.Please help me. Thanks in advance. >>> More
Ubuntu: package installed, but files missing?

as seen on Server Fault - Search for 'Server Fault'
I have been playing around with the /etc/asterisk directory, installing the related pacakge (asterisk-config), removing it, removing the directory manually (just playing around to get the configuration synced to my configuration repo). Now I just want to reinstall the official package, so I do: root@tethys:/etc#… >>> More
Can we manage incoming call number landing on asterisk?

as seen on Server Fault - Search for 'Server Fault'
I am using Asterisk 1.4.2 in two different machine. I have configured some extensions in asterisk. When any caller, dial my extension number with local number then if I see asterisk console (asterisk -r) then incoming number is starting with 0, but if caller dial same extension number using STD number… >>> More
Asterisk SIP digest authentication username mismatch

as seen on Super User - Search for 'Super User'
I have an asterisk system that I'm attempting to get to work as a backup for our 3com system. We already use it for a conference bridge. Our phones are the 3com 3C10402B, so I don't have the issue of older 3com phones that come without a SIP image. The 3com phones are communicating SIP with the… >>> More