http://www.google.com/search?q=port+5000+asterisk
answer is in the first hit :)
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Robert Thomas
Sent: Wednesday, April 13, 2011 5:46 PM
To: Asterisk Users Mailing List - Non-Commercial
Each media stream will use two, one for RTP and one for RTCP. In your case
10200/10201 are an RTP/RTCP pair, and 10504/10505 are another pair. RTP is
always and even numbered port, and RTCP is always RTP port + 1. Yes, it's
in the RFC for RTP.
The fact that you have two pairs means that two
We have two models of Cisco phone that do this, but Asterisk handles it fine.
I don't recall doing anything special to make it work. We're using Asterisk
v1.4.17.
What is the error message on Asterisk?
-Original Message-
From: asterisk-users-boun...@lists.digium.com on behalf of
-Original Message-
From: asterisk-users-boun...@lists.digium.com on behalf of Noah Miller
Sent: Thu 5/6/2010 10:41 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] OT: NAT in SPA922
It is a building, with 24 separated rooms, each room will
James,
I'm assuming your talking SIP here.
It's not the Contact header that is important here but the Via header.
Responses should be going back to whatever port is specified there.
Contact header is used for incoming requests.
Via header is used for responses to outgoing requests.
7965:
in the SIP/2.0 180 Ringing, the SDP shows:
a=sendonly
this is hold by rfc 3264. then when the other end picks up, a new SDP is
probably sent with
a=sendrecv
I believe your server is acting correctly.
-Original Message-
From: asterisk-users-boun...@lists.digium.com on behalf of
I don't know in your particular case, but if I call a PSTN endpoint via my
provider, the SIP signaling is different than if I'm calling a remote SIP
endpoint. This is because PSTN gateways have to make decisions (about codecs,
eg) independently of the remote endpoints.
In other words,
all you need to do is make the configurations mirror each other.
in the example below, all of the endpoints are SIP, but it doesn't matter if
you move the endpoints to another protocol, like Fxs:
on serverA
extesions.conf:
[phones]
include = localphones
include = to_serverB
[localphones]
call-id doesn't match?
SIP/2.0 200 OK
...
Call-ID: 2117388659-506...@82.158.83.xxx
...
ACK sip:6615xx...@130.117.xxx.xxx SIP/2.0
...
Call-ID: 2117388659-506...@192.168.1.100
...
I'm not sure, but I think that the part after the '@' must also match
throughout the dialog. A Grandstream bug?
