How do you overcome the 302 moved temporarily messages? I tried using
a wireless bridge with one of my Cisco phones Polycom 301's and when
I tried to receive a call it would give me the 302 error.
On 4/27/07, Michael Graves [EMAIL PROTECTED] wrote:
--Original Message Text---
From: Mike
Date:
Which one has video for the mac?On 9/7/06, Nick Ellson [EMAIL PROTECTED] wrote:
Hello Michael,I just had both Mom and my brother up as extensions on my Asterisk pbxusing IAX2, the Cubix phone for now, but I downloaded and tried several. Iloke multiple lines, but a clean GUI is better for my
I've never experienced any of those problems. I can send video to other eyeBeam versions, 3.0 is the only version that supports video on the lite side. I've never lost any SIP information and only one registration isn't a big deal to me. If you need more than one buy the full eyebeam version.
On
What's wrong with X-Lite 3.0? I haven't had any issues with it and find it to be one of the best SIP video software choices, and it's free.On 7/27/06, Joao Pereira
[EMAIL PROTECTED] wrote:Hello to all
can someone recommend me a nice SIP client with video for windows??I tried X-Lite 3.0 but it's a
Hey everyone I have my Asterisk server setup as the DMZ on my Linksys router. If I use the internal IP as the domain in Xlite clients will register and work, however, if I use the FQDN for my asterisk server the clients will not register. I have all the extensions set to NAT=yes and have modified
Anyway around that? It's a PITA to have to change that all the time with my PDA laptop.On 12/29/05, Kerry Garrison
[EMAIL PROTECTED] wrote:
If the machines with X-Lite are on the local network, use
the private ip, if they are outside the network, use the public
ip.
-Kerry
From: [EMAIL
I'm running AAH 2.2 and *1 works from my eyebeam sip phones to do on demand recording.
You need to set the DIAL_OPTIONS of wW in order to utilize this
feature. lower case w means called person can initiate, upper case
means callee can initiate, I think that is the order.
They show up as
Good lord, it's part asterisk part goiax.com
If you have an issue with it ignore the thread. At first I thought I had asterisk config'd wrong.
Now I know better than to waste my time with a list that has people like you on it.On 10/21/05, Robert Webb
[EMAIL PROTECTED] wrote:On Fri, 21 Oct 2005
ine is 339
- Original Message -
From:
Blake Krone
To:
[EMAIL PROTECTED]
Cc:
Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Friday, October
21, 2005 1:38 AM
Subject: Re:
[Asterisk-Users] Goiax.com DID not working anymore?
What is your prefix? M
I've been using my goiax.com DID for a few days now and it is no longer
working. I get the number or code you dialed can not be found. I
haven't touched any configs or anything on the asterisk box since it
was working last night.
Anyone else having problems using the DID from goiax?
Thanks
not working
- Original Message -
From:
Blake
Krone
To:
Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Friday, October 21, 2005 12:32
AM
Subject: [Asterisk-Users]
Goiax.com DID
not working anymore?
I've been using my goiax.com DID
for a few days now
What is your prefix? Mine is 978, maybe only certain ones are having problems?On 10/20/05, Robert Webb [EMAIL PROTECTED]
wrote:
Just tested mine and it is working
fine.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] On Behalf Of Blake
KroneSent: Thursday, October 20,
Well it sounds like there's a few of us that are having problems. When
it was working I would see entries in the log so I'm assuming something
at goiax.com is not working anymore and the DID simply isn't able to
know who to call.
On 10/20/05, Tom Lynn [EMAIL PROTECTED] wrote:
I've always received
What is the best solution? I dont want to have modify firewall's at all or do port fowarding. Ideally I would like a solution that with either a softphone or wireless hardphone one could connect via friends, family, or hotspots without reconfiguring their devices.
What are people using? STUN?
So I won one of these on ebay, in the auction it says it has the RJ45
ports on it but it doesn't :(
If I were to get an analog adapter would I be able to use the video
portion of this or am I SOL? The auction requires me to pay for
shipping back, so I end up losing money unless I sell it on my
]
[mailto:[EMAIL PROTECTED] On Behalf Of
Blake Krone
Sent: Tuesday, 19 July 2005 11:08 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Vizufon Video Phone
So I won one of these on ebay, in the auction it says it has
the RJ45 ports
acceleration.
Cheers.
S.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Blake Krone
Sent: Friday, July 08, 2005 5:53 AM
To: Matt Riddell
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] SIP Xten eyeBeam Video
Which is what I was finding out, but once I set it to the DMZ
everything was fine seen as I didn't have to worry about the ports at
all.
Now if I could only get video to work again I'll be all set!
I'll have to look into the IAX2 protocol also.
-blake
On 7/7/05, Rich Adamson [EMAIL PROTECTED]
I only have the basic h.263 enabled in Xten.
Everytime I start sending video it just shows noise, I can see in the
log that it's trying to use the 263 codec.
On 7/7/05, Matt Riddell [EMAIL PROTECTED] wrote:
Blake Krone wrote:
Hello all, I HAD video working before I upgraded to 1.08 (latest
=On-Demand
qualify=no
port=5060
nat=never
[EMAIL PROTECTED]
host=dynamic
dtmfmode=rfc2833
context=from-internal
canreinvite=no
callerid=Blake Krone 200
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=h261
allow=h263
allow=h263p
In the full log I see the following when I start video:
Ooh, format
Ronald I've been struggling with the same problem. So far I have set
my asterisk server to be in the DMZ and still nothing with sound, no
music on hold or anything. From what I'm reading if your SIP clients
connect from behind a NAT/Firewall/Private LAN there are problems and
it doesn't work well
forgot to include the list
-- Forwarded message --
From: Blake Krone [EMAIL PROTECTED]
Date: Jul 6, 2005 9:07 PM
Subject: Re: [Asterisk-Users] Re: Remote SIP Connections
To: dbruce [EMAIL PROTECTED]
Just had my brother connect from his time warner cable in minnesota to
my
Well it's now working somehow magically, heh.
Must have been the DMZ settings instead of the port forwarding.
On 7/6/05, Blake Krone [EMAIL PROTECTED] wrote:
forgot to include the list
-- Forwarded message --
From: Blake Krone [EMAIL PROTECTED]
Date: Jul 6, 2005 9:07 PM
I just set my asterisk to the NAT and it works magically.
On 7/6/05, Blake Krone [EMAIL PROTECTED] wrote:
Ronald I've been struggling with the same problem. So far I have set
my asterisk server to be in the DMZ and still nothing with sound, no
music on hold or anything. From what I'm reading
Hello all, I have my * server setup behind a Linksys WRT54G on
Adelphia cable. I have forwarded 5060,1-10020, and another port
set can't remember off the top of my head but I can't seem to connect
to the * server from any locations that are direct connects to the
Internet. Am I missing a
I have gotten them to be able to connect but I am unable to hear the
other person and they can't hear me either.
What else am I missing?
On 7/5/05, Blake Krone [EMAIL PROTECTED] wrote:
Hello all, I have my * server setup behind a Linksys WRT54G on
Adelphia cable. I have forwarded 5060,1
on the
WRT54G.
You could also play with port triggering settings, but that is also a very
dificult process.
Regards,
Derek Bruce
- Original Message -
From: Blake Krone [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Tuesday, July 05, 2005 7:10 PM
Subject: [Asterisk
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