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
On your sip.conf for each sip endopoint set canreinvite = yes.
That way the rtp stream won´t go through *. The only problem though is for
ATA 186. They need canreinvite = No when they are in a NAT environment.
- Original Message -
From: Low, Adam [EMAIL PROTECTED]
To: [EMAIL
Check out this bug
http://bugs.digium.com/bug_view_page.php?bug_id=005
its a know problem. I have played with the canreinvite stuff to no end and have never
gotten my Cisco Phones to do P2P RTP. I am going to try free world dialup to see if
it does P2P with my Cisco Phones then it might
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