outgoing calls transfer not work

[jsebasmz]

hi everyone

When I try to transfer an outgoing call to a internal extension, the call cuts suddenly.

IPoffice 500V2 R8
SIP Trunk

thanks for your help.

Regards

 

[Bas1234]

Enable rtp keep alives on lan1 or 2 also disable SIP ALG or for Cisco No fixups on sip on the router side.

___________________________________________
It works! Now if only I could remember what I did...

Dain Bramaged (Avaya Search tool

http://tinyurl.com/bas1234

)
______________________________________

 

[tlpeter]

RTP is more for when you have no speech.
I think you are not sending the correct URI to the provider (read number)


BAZINGA!

I'm not insane, my mother had me tested!

 

[Bas1234]

· RTP keepalives: Release 6.1+
These settings can be used with SIP trunks associated with the LAN through their Use Network Topology Info setting (Line | Transport). For some scenarios, with frequent call forwarding on the same SIP trunk, speech path may be lost during the connection. The use of periodic keepalive packets may prevent the issue.

· Scope: Default = Disabled
Select whether the sending of keepalive packets should be disabled or sent for RTP or for both RTP and RTCP.

· Periodic timeout: Default = 0 (Off). Range = 0 to 180 seconds.
Sets how long the system will wait before sending a keepalive if no other packets of the select SCOPE are seen.

· Initial keepalives: Default = Disabled.
If enabled, keepalives can also been sent during the initial connection setup.

___________________________________________
It works! Now if only I could remember what I did...

Dain Bramaged (Avaya Search tool

http://tinyurl.com/bas1234

)
______________________________________