SIP FORWARD ISSUE

[VoiceonData]

Hi,

I have a IP 406v2 set up on 4.2 20 software.

They have sip lines coming into the system which are working fine.

Incoming Calls work - speech path works
Outgoing Calls work - speech path works

the problem lies when someone wants to fwd there calls off site.

The forwarding part of the call works and the destination rings but there is no speech path.


Any ideas ???

Regards

Tyron

voiceondata

http://www.voiceondata.co.uk

 

[PlatGuy]

Known issue.

Do this............

To help find the port pairs used for SIP Calls – use SysMon, select the SIP Filters and also check the “Development Tracing” checkbox in the System tab.

In the Status Tab open a RTP Streams window.

Make some test calls and capture them on the SysMon. Then select highlight “StunInfo:” then press F4 and you will get a window like Fig 2. You will see there are ports starting with 491xx (Private) and 564xx (Public) being mapped by Stun against the IP Address C0a8fee4 (192.168.254.228 – My IP Office address)

In your router, you need to create port pairs to resolve what ports Stun is using when these calls are routing in then out of the system to the same SIP Provider and external IP Address.

If you look at Fig 1, you can see the typical internal ports the IPOffice is using – 49152 (inbound call) and 49154 (Outbound call) for this particular call. Looking in figure 2, you can see how each port resolves.
ie Stun creates port 49152
Then a paired port is created 49153 (there is always an odd and an even port)
It then resolves these to external ports 56448 & 56449

So there is a pairing of ports:
49152 56448
49153 56449

This needs to be allowed in the Firewall/Port redirection in the ADSL router and pointed at the IPOffice so we point the RTP in the right direction when a call is received in and out of the system.

 

[Bas1234]

Or try to tick "Re-Invite" on the SIP tab, you provider needs to support this option to make it work.

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

Dain Bramaged
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[VoiceonData]

Thanks for the reply very useful.

Apart from we are not using stun



voiceondata

http://www.voiceondata.co.uk

 

[DavidCT]

VoiceonData,

What firewall/router are you using?

On the SIP config, find "binding refresh" and set it to something like 30. Make some test calls. If it works, INCREASE the value to 45 and make some test calls. You want binding refresh as high as possible, as long as your calls work.