SIP call on wait results in 'dead' line

[DJSmiley]

I'm having some issues with an OmniPCX (R7.0)

It has been setup for use with a SIP trunk. This works ok, I'm able to dail and can be dailed on the trunk.
However, putting a call on hold results in a dead line. I can get back to the call, but there is no sound anymore (both sides). The line is not disconnected, only totally silent.

I had similar issues upon initiating a call: I had sound for 2sec. This has been solved by disabling the reinvite on multiple-codec answer (Noteworthy adress MultAnsReinv as discussed in Ed.W820 SIP Public R7.0 from the BPWS)

After contacting the voip operator, it seems there is another invite send when I put the call on hold. However, I don't know the noteworthy adress refering to this invite. Its not on any file as far as I can see.
The Noteworthy overview from R6.1 (Article 1 N°80 - Ed 07) doesn't refer to anything regarding this (even the MultAnsReinv isnt mentioned, despite its already available in R6.1)

Does anyone have an updated overview of the noteworthy adresses from R7.0, or another file where these are discussed, or does anyone know how to disable the re-invite upon putting a call on hold?

[DJSmiley]

*kick*

Still having these problems.

Meanwhile, I have connected the PBX directly to the internet (assigned it a public IP adress), and the problems are solved. Offcourse, this is way-of a preferred solution. (Just to verify the problem).

This also confirms my initial thought on this problem: NAT port issues.

Anyone with an updated view how to bypass this re-invite (causing NAT to release the connection) or another solution?

[tuxalize]

Hi DJ,
I had the same issue.
Actually the problem was on Router configuration.
It is correct that the OXO sends another invitation.
What's happened was for any reason the router we were using was changing the ports. In other words, if you see the details of the Wireshark traces, from internal side and external site, you will see that OXO sends invitation on one port, then the router sends to external in another port (SIP NAT/T), the external SIP server reply on this port, but the router sends in another port to OXO.
Basically it happened because we were using Firewall, VPN, port forwarding. After talking to the router tech support they suggest us to change the order of Firewall rules on the router. It worked fine.

Regards,
Tux

[jay_tee_ef]

Hi we're having the same problem too... if the oxo is not behind the firewall, will it work if we work on the config of router?

[jay_tee_ef]

oxo is connected directly to public IP. we have the same problem though. the call drops when there is an attempt to transfer it from 1 local to local. a 2nd invite is recognized on our platform via wireshark but it doesnt have the same cli

[royce.clark]

I have the same problem. Putting a call on hold or an attempt to transfer results in dead air for both parties. This is also affecting calls being answered by the voicemail system and AA. If someone releases the call during the AA message the call is not released in the OXO. Is this because of the port changing in the ADSL router? or is this a setting I need to change in the OXO? Please help...

[Saleemi]

i have same problem problem on OXO connect with release 6.0/128.001. please help....