Nabeel,
I suppose you OpenSIPS seats on a public IP, right ? The callee looks to
have a private IP. And, at IP level, it is impossible to route from a
public IP to a private one.
I see your script has NAT traversal support. My question is - did the
callee properly registered via this script ? can you do an "opensipsctl
ul show" to see the callee's registration ?
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 06.08.2015 07:14, Nabeel wrote:
Hi,
Yes, the destination IP is 192.168.0.19:60912
<http://192.168.0.19:60912/> and both phones are registered to
OpenSIPS. In this case, the callee is connected to Wi-Fi (hence
192.xx IP address) and the caller is connected to a mobile network.
The opensips.cfg I am using was generated from 'make menuconfig',
except with the addition of "alias=domain.com <http://domain.com>". I
have attached my config file at this link:
http://pastebin.com/0QRyC938
On 6 August 2015 at 05:00, SamyGo <[email protected]
<mailto:[email protected]>> wrote:
Hi Nabeel,
Quick question; what is this destination ip? 192.168.0.19:60912
<http://192.168.0.19:60912> ? - Destination User Agent Registered
on OpenSIPS?
Can you share the opensips.cfg code snippet for this call ?
On Wed, Aug 5, 2015 at 11:55 PM, Nabeel <[email protected]
<mailto:[email protected]>> wrote:
Hi,
I am using the residential script generated by 'make
menuconfig', with UDP and NAT support enabled. I added
"alias=domain.com <http://domain.com>" to the config because
otherwise the UA did not register with my domain
([email protected] <mailto:[email protected]>). When I
attempt to make a call, I see '408 Request Timeout' in the sip
trace and the call does not connect. Please check the
log/trace below and advise how to fix this.
SIP trace:
http://pastebin.com/u5h9qGNr
OpenSIPS log:
http://pastebin.com/B8PUCKh0
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