Dear all,
Concerning the REINVITE discussion...
Does anybody can confirm if the Cisco ATA186 accepts the REINVITE Sip
Message, and if is compatible with this feature?
In other words, is it possible to make the ATA186 change the RTP destination
and start sending the media packets straight to
:23 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] RTP session traversing Asterisk server...
Dave,
You can use a sniffer to view the contact field in the INVITE Message
that
the Originating Phone sends to *. Then look at the INVITE Message that
*
sends to the remote phone and compare
[EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, August 01, 2003 4:20 PM
Subject: RE: [Asterisk-Users] RTP session traversing Asterisk server...
Ricardo,
You are right about the contact field in the INVITE message. It does
display the address or our Asterisk proxy. It seems to me
Hi,
Cisco 7940/60 does P2P with FWD.
BR,
Dan
- Original Message -
From: Dave Packham [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, July 29, 2003 5:30 AM
Subject: Re: [Asterisk-Users] RTP session traversing Asterisk server...
Check out this bug
http://bugs.digium.com
that are NAT'd behind ADSL/cable connections.
I don't seem to be hitting the bug that Dave mentioned below ...
-Original Message-
From: Dave Packham [mailto:[EMAIL PROTECTED]
Sent: 29 July 2003 04:30
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] RTP session traversing Asterisk
server
Packham [mailto:[EMAIL PROTECTED]
Sent: 29 July 2003 15:43
To: [EMAIL PROTECTED]; [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] RTP session traversing Asterisk
server ...
can you share the SIP conf entries that you are using to get
this to work? I have played with the canreinvite
PROTECTED]
Sent: 29 July 2003 15:43
To: [EMAIL PROTECTED]; [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] RTP session traversing Asterisk
server ...
can you share the SIP conf entries that you are using to get
this to work? I have played with the canreinvite and
reinvite entries
-
From: Dave Packham [mailto:[EMAIL PROTECTED]
Sent: 29 July 2003 15:43
To: [EMAIL PROTECTED]; [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] RTP session traversing Asterisk
server ...
can you share the SIP conf entries that you are using to get
this to work? I have played
Yes, i've observed the same operation :|, Adam.
I've the last CVS Asterisk, and two softphones (Linphone 1.12 and X-lite
v2 last version), both with speex code active.
When i call from one to another ... ringing ok but ... when try to talk
... the Asterisk go crazy warming out of memory (i
]
Sent: Monday, July 28, 2003 11:29 AM
Subject: [Asterisk-Users] RTP session traversing Asterisk server ...
I've been reading up on the SIP and related (SDP/RTP) RFC's and as I would
expect the RTP session should ideally be between the two end points of the
call, in my case the AS5300
environment.
- Original Message -
From: Low, Adam [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, July 28, 2003 11:29 AM
Subject: [Asterisk-Users] RTP session traversing Asterisk server ...
I've been reading up on the SIP and related (SDP/RTP) RFC's and as I would
expect the RTP
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