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VOiP Setup - Outgoing Ok, No Incoming

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  #1  
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Junior Guru Wannabe
 
Join Date: Oct 2010
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VOiP Setup - Outgoing Ok, No Incoming


Hi,

I have setup Elastix on my own VPS. I have setup all the configuration alright as per my knowledge. I can make Outgoing calls fine but I cannot receive Incoming calls. I am using a softphone.
Let me know if anyone of you can resolve this Incoming calls issue or guide me through the process.

Looking forward.
Thanks!



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  #2  
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Web Hosting Evangelist
 
Join Date: Oct 2007
Location: Northampton, UK
Posts: 548
Some if it may depend on what provider you are using. I wrote a guide that may help - http://sysadminman.net/blog/2009/get...did-number-386

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  #3  
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WHT Addict
 
Join Date: Nov 2004
Location: Marietta PA
Posts: 137
First with out seeing your configuration I would suggest that you make sure you setup the inbound route correctly.

Next I would do a tail -f 10 /var/log/asterisk/full and watch the logs for a bit to make sure that the trunk is fully registered.

If you are running a firewall I would verify that you have the correct ports open.

At a min ports 5060 - 6000 for registration of SIP UDP
10000-20000 UDP for SIP communication. That is inbound and outbound.

Finally do a asterisk -r and place a inbound call and see if your asterisk system even sees the call and if so watch the call flow to see where it is failing.

We provide VoIP Consulting if there is still a problem.

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  #4  
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Join Date: Oct 2002
Posts: 120
Quote:
Originally Posted by webhostinggeek View Post
Hi,

I have setup Elastix on my own VPS. I have setup all the configuration alright as per my knowledge. I can make Outgoing calls fine but I cannot receive Incoming calls. I am using a softphone.
Let me know if anyone of you can resolve this Incoming calls issue or guide me through the process.

Looking forward.
Thanks!
You are behind a NAT firewall? (your home or office router) it is probably that Network Address Translation is the issue

You need to use some form of NAT Traversal, such as STUN, Mediaproxy, rtpproxy, ICE, SER/Kamailio/Opensips depending on your softphone support

Kamailio and OpenSIPS are forks of OpenSER (which is no longer developed)

They are used at production environments in conjunction with mediaproxy or rtpproxy to deal with NAT Traversal


it is also possible to use Client Side NAT Traversal solutions such as STUN and ICE (depending on your softphone support)

  #5  
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Junior Guru Wannabe
 
Join Date: Jan 2004
Location: Lakewood, NJ
Posts: 87
Just use STUN in Asterisk. Also make sure you register pretty frequent, use qualify=2000 and forward UDP ports 5060, 10000-20000 to your device.

Also look at the Asterisk logs to see if it is a config issue.

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