asterisk users - Jul 2005 - [Asterisk-Dev] Digium to Sponsor a Pizza party at Cluecon

Digium, the creator and primary developer of Asterisk, the industrys  
first Open Source PBX, will be hosting a pizza party from 4pm to 6pm  
on the first day of Cluecon. We look forward to everyone coming out  
to enjoy this opportunity to meet fellow developers and users in a  
more casual environment.

I would like to personally thank Mark Spencer and Digium for their  
support.

Thanks,
Brian West
Asterlink.com
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Brian West [brian.west@mac.com] wrote:
> Digium, the creator and primary developer of Asterisk, the industrys
> first Open Source PBX, will be hosting a pizza party from 4pm to 6pm
> on the first day of Cluecon. We look forward to everyone coming out
> to enjoy this opportunity to meet fellow developers and users in a more
> casual environment. 
> 
Thanks.  I missed the first 600 copies of that announcement. :-)

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  _/_/_/   _/_/      _/    _/    _/    _/_/  _/   K e v i n   W a l s h
 _/ _/    _/          _/ _/     _/    _/  _/_/    kevin@cursor.biz
_/   _/  _/_/_/_/      _/    _/_/_/  _/    _/

Hi,

I've got two Yuxin phones that I want to use to make calls between two 
branch offices - ip to ip calling.

I've given both phones (behind a firewall) static IPs. I've set up port 
forwarding on the firewall for udp on ports 5060-5061 to connect both 
phones.

I'm using the sip protocol and have set up the register,  signal and 
control port on both to 5060 and the rcp port to 5601.

The problem is that each phone can ring each other and a connection is 
made when the other party picks up but there is no sound. It all worked 
fine when they were on the same IP segment.

I'm not sure if I I have some ports wrong or if for IP to IP calling on 
these phones I should try h323 instead..

Does anyone have any suggestions as to what I should try to get this 
working ?