OXO R910 and Avaya G650 CM6 via SIP Trunk

A H323 and SIP forum only !
User avatar
tot3nkopf
Alcatel Unleashed Certified Guru
Alcatel Unleashed Certified Guru
Posts: 4001
Joined: 02 Feb 2006 10:41
Location: Germany & Romania
Contact:

Re: OXO R910 and Avaya G650 CM6 via SIP Trunk

Post by tot3nkopf » 25 Sep 2013 08:29

ray_III wrote: I think tot3nkopf is correct that destination port is fixed
I did not said that it is fixed. I said you can configure the destination port (so you can use whatever destination port: 5060, 5066, 7777 --> this is the port on which the other pbx is listening as stated).

User avatar
enio.eltz
Senior Member
Posts: 1037
Joined: 28 Jul 2009 09:48
Location: Taquara - RS - Brazil

Re: OXO R910 and Avaya G650 CM6 via SIP Trunk

Post by enio.eltz » 25 Sep 2013 08:41

Hi guys

I SAID that in my thought destination port could be fixed as 5060, but ok, Tot3nkopf explained that is possible by configuring remote SIP port in gateway parameters as I wrote before, asking Ray_III to change that parameter. But he said he did but no success...
Regarding to TCP transport mode, I am not sure about this. I dind't know any parameter to check this.

Best regards.
Enio Eltz Filho
ACFE OT R1
ACFE OXE R11/R10
ACFE OXO R10/R9

User avatar
tot3nkopf
Alcatel Unleashed Certified Guru
Alcatel Unleashed Certified Guru
Posts: 4001
Joined: 02 Feb 2006 10:41
Location: Germany & Romania
Contact:

Re: OXO R910 and Avaya G650 CM6 via SIP Trunk

Post by tot3nkopf » 25 Sep 2013 09:02

ray_III wrote: Is there's a way to force OXO to use TCP transport mode?
Maybe:

UDP to TCP Switching Deactivation Noteworthy Address

no_tcpswitching (default 00)


switching of UDP to TCP is enabled


Description
This parameter allows to disable the automatic switching of transport type (from UDP to TCP)
for outgoing SIP messages.
When not disabled, SIP messages are switched to TCP.
The default value 00 activates the switching of UDP to TCP.
The 01 value deactivates the switching of UDP to TCP.

By the way this is from Sys doc. (which means we are doing your homework now)

User avatar
ray_III
Member
Posts: 113
Joined: 12 Sep 2010 20:03

Re: OXO R910 and Avaya G650 CM6 via SIP Trunk

Post by ray_III » 25 Sep 2013 21:42

Hi All,

Thanks for your replies, I already read that "no_tcpswitching", but unable locate this parameter.

User avatar
enio.eltz
Senior Member
Posts: 1037
Joined: 28 Jul 2009 09:48
Location: Taquara - RS - Brazil

Re: OXO R910 and Avaya G650 CM6 via SIP Trunk

Post by enio.eltz » 25 Sep 2013 22:26

This is not the name, it is a byte of many ones. In the technical manual there is the right name that is something like VOIPntwaddr. Tomorrow morning, my local time I post the right name.
Enio Eltz Filho
ACFE OT R1
ACFE OXE R11/R10
ACFE OXO R10/R9

User avatar
enio.eltz
Senior Member
Posts: 1037
Joined: 28 Jul 2009 09:48
Location: Taquara - RS - Brazil

Re: OXO R910 and Avaya G650 CM6 via SIP Trunk

Post by enio.eltz » 26 Sep 2013 06:38

Hi

The right name is VOIPnwaddr offset/byte 18. Default is 00 Hex. In the picture attached you can see the description.
You do not have the required permissions to view the files attached to this post.
Enio Eltz Filho
ACFE OT R1
ACFE OXE R11/R10
ACFE OXO R10/R9

Post Reply

Return to “H323 / Sip”