> canreinvite=no
[general]
port = 5060
srvlookup = yes
nat = yes
tos = lowdelay
disallow = all
allow = ulaw
allow = gsm
allow = alaw
context= INVALID
Currently my IP phones haves this in the sip.conf
[4403]
type= friend
username
Since I always use canreinvite=no, you're probably right.
John
Bruce Komito wrote:
Not to beat a dead horse, but I had the problem even with the two lines on
different ports. The canreinvite=no thing solved the problem.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 284-5800 ex
Not to beat a dead horse, but I had the problem even with the two lines on
different ports. The canreinvite=no thing solved the problem.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 284-5800 ext 115
On Mon, 24 May 2004, John Fraizer wrote:
> Bruce Komito wrote:
>
> > In
Bruce Komito wrote:
> In sip.conf, try setting canreinvite=no for both lines.
>
> Bruce Komito
> High Sierra Networks, Inc.
> www.servers-r-us.com
> (775) 284-5800 ext 115
canreinvite=no will sometimes make a difference but, I believe that what
most people are running into is what I described in m
Barry Fawthrop wrote:
The problem is probably that both of your SIP phones are using the same
port. I played with two 7960's behind a Linksys on Saturday and finally
got them playing right when I changed the following:
In Phone 1's SIP[macaddr].cnf:
voip_control_port: 5061
In Phone 2's SIP[maca
PROTECTED]
Subject: Re: [Asterisk-Users] 2 Sip phones behind un-natted Asterisk
What does the Xten diagnostic log say about a single
session?
Also, what does the * SIP debug output say? I'd be
very interested to see what IPs and ports SIP is
trying to set the RTP connection on. (Since SIP
ap
>
> I'm having a similar problem with snom 200s would changing the port
> work there also or is that just a 7960 issue? Do you or any other
> know where I would change that on a snom 200 ??
>
> thanks in advance
>
> Barry
> ___
try adding
Canreinvi
In sip.conf, try setting canreinvite=no for both lines.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 284-5800 ext 115
On Mon, 24 May 2004, Barry Fawthrop wrote:
> > The problem is probably that both of your SIP phones are using the same
> > port. I played with two 7960's
> The problem is probably that both of your SIP phones are using the same
> port. I played with two 7960's behind a Linksys on Saturday and finally
> got them playing right when I changed the following:
>
> In Phone 1's SIP[macaddr].cnf:
>
> voip_control_port: 5061
>
> In Phone 2's SIP[macadd
What does the Xten diagnostic log say about a single
session?
Also, what does the * SIP debug output say? I'd be
very interested to see what IPs and ports SIP is
trying to set the RTP connection on. (Since SIP
appears to be working fine, it's the RTP part that is
breaking).
Are both the Xten an
John, In my case, the two ports are not using the same IP port (one is on
5060, the other on 5061), but of course, they are on the same IP address.
I think that is what is confusing the NAT server, but I don't know what to
do about it.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(
Chad Brown wrote:
I have 2 SIP phones (Cisco 7960 & XTen) behind a NAT provided by a
Linksys firewall that supports UPnP. The Asterisk server has a public
IP. Here are the problems that I am having with this configuration…
1. The 2 SIP phones can call MeetMe and have a conference but cann
I'm not the original poster, but I have the same problem with a Sipura.
In my configuration, I have line 1 set to port 5060 and line 2 set to port
5061. I assume that is what you are suggesting, right?
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 284-5800 ext 115
On Mon, 2
>> I have 2 SIP phones (Cisco 7960 & XTen) behind a NAT provided by a
>> Linksys firewall that supports UPnP. The Asterisk server has a
>> public IP. Here are the problems that I am having with this
>> configuration...
>>
>>
>>
>> 1. The 2 SIP phones can call MeetMe and have a conference but
I am having exactly the same problem with two phnes connected to a Sipura
behind a Linksys. I'm sure this is NAT, because it works fine when I move
the Sipura out from behind the Linksys.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 284-5800 ext 115
On Mon, 24 May 2004, Ch
On Mon, 24 May 2004, Chad Brown wrote:
> 1.The 2 SIP phones can call MeetMe and have a conference but
> cannot call each other. (Yes, they connect but no audio either
> direction)
> 2.I have verify=yes in the sip.conf for both phones. Both phones
> constantly go Unreachable. (However, th
I have 2 SIP phones (Cisco 7960 & XTen) behind a NAT
provided by a Linksys firewall that supports UPnP. The Asterisk server
has a public IP. Here are the problems that I am having with this configuration…
The 2 SIP phones can call
MeetMe and have a conference but cannot call eac
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