Hello Jan,
It depends how many telephone lines you will connect with asterisk - the
more lines you need the more expensive hardware you will need. For a single
line the hardware can be as cheap as $20 (used) and for more lines the price
increases rapidly.
Hello,
I'm having intermittent STUN trouble. Every one out of perhaps 5 reboots
the PAP2 contacts STUN ... on the other attempts it just skips that step all
together. I have been verifying this using ethereal which shows the
distinctive STUN server DNS lookup followed by about 10 STUN queries
Depends what the beep sounds like ... but I've been having this system on a
busy system which has XP100 and the Ethernet cards or other devices sharing
one IRQ. You might need to spread out the IRQs so that XP100 get's its own
and Ethernet gets another one of its own.
How fast of a system is it?
Has anyone made this work? For me everything is fine until I switch
canreinvite form no to yes. What happens is that asterisk hangs on
attempting native bridge ... from what I understand attempting native
bridge means that the RTP is routed through asterisk (just without any
codec translation)
Hello,
Im having trouble figuring out how to setup Asterisk
so that its only a registrar not passing any RTP data during
phone calls.
So far I got this far:
Asterisk server holds registration information for phones
Phones register with Asterisk giving it their ip+port where
they
: [Asterisk-Users]
Registrar only setup
have you tried in the sip.conf for the devices
canreinvite=yes
- Original Message -
From: Tomas Florian
To: asterisk-users@lists.digium.com
Sent: Tuesday, August
30, 2005 8:48 PM
Subject: [Asterisk-Users
Hello,
All Im looking for is a yes/no answer here. I have
heard that the following scenario is possible (reasonably easy to implement as
well) but I just dont get it :-) if it is possible Ill
go ahead and learn on my own, I just dont want to waste time on
something that will not work.
I had a recent bad experience with Compaq
with asterisk. For some reason it could not be forced to put my PCI cards
on anything other than IRQ 11. This is a major problem because when all
the cards (2 xnetwork,2x FXS) are active and generate interrupts it chops up
the sound quality.
I
Hello,
I know that I can have DID on a single line, but will AMP support 2+ lines
with DID?
Has anyone tried this? Straight forward?
Thank you,
Tomas
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Hello,
Is it possible to make BT100 phones ring in different ways based on where
the call is coming from?
The general idea is that I need the BT100 ring in 2 different ways depending
on whether the call come from Zap1 or Zap2.
It's because this system is for a receptionist answering two
was not changing my underwear in the morning :-) LOL
On the Asterisk side it's just the usual:
Nat = yes
Qualify = yes
Tomas
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tomas Florian
Sent: Saturday, April 23, 2005 11:56 PM
To: 'Asterisk Users Mailing
Hello,
I'm having some major problems getting SIP phones to register whenever I put
them behind a Linksys router. The same phones will register behind any other
NAT (I've tried 3 others without problems)
I've been debugging using Ethereal and these are the differences that I
found between
?locationid=US-AK
Tomas Florian wrote:
Hello,
I'm having some major problems getting SIP phones to register whenever I
put
them behind a Linksys router. The same phones will register behind any
other
NAT (I've tried 3
I'm trying to register BT100s ... (doesn't work)
X-Lite seems to work though
Tomas
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mojo-Jojo
Sent: Saturday, April 23, 2005 8:48 PM
To: Luki; Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
on the Grandstream?
On 4/23/05, Tomas Florian [EMAIL PROTECTED] wrote:
I'm trying to register BT100s ... (doesn't work)
X-Lite seems to work though
Tomas
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mojo-Jojo
Sent: Saturday, April 23, 2005 8:48 PM
: no
- Local sip port: 5060
- Use NAT ip: no
- Proxy require: no
And in my sip.conf I have
Nat=yes
Qualify=yes
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tomas Florian
Sent: Saturday, April 23, 2005 11:04 PM
To: 'Pedro'; 'Asterisk Users Mailing List
Hello,
I'm in the process of implementing the following setup
External SIP phones at another location(s) (nat = yes)
|
| Analog phone line
| |
|--
|ext if 142.x.x.41
|
|Asterisk
|
|int if 192.168.0.1
|--
|
Internal SIP Phones
Hello,
I'm trying to use some VoIP phones behind a Linksys WRT54G router but can't
get them to register. The annoying thing is that I've taken the phones to 3
other locations with non-Linksys NAT routers and the phones work immediately
without any problems.
I've tried STUN, outgoing proxy .
Hello,
I'm having trouble getting BT100 to identify NAT type reliably for Asterisk.
My setup is as follows:
- Asterisk is on the open internet 142.x.x.41
- BT100 phones are behind NATs
- I use STUN for my BT100 : 142.x.x.41 (same server as Asterisk)
- BT100 firmware (tried .16,.18,.23 same
that stun can
bind to. Is that what you are doing?
Alex
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tomas Florian
Sent: Saturday, April 16, 2005 1:17 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] BT100 wrong NAT detection
Hello,
I'm
. Normally Stun requires 2
systems, system with 2 NIC cards, or at minimum 2 IP addresses that stun can
bind to. Is that what you are doing?
Alex
-Original Message-
rom: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tomas Florian
Sent: Saturday, April 16, 2005 1:17 PM
To: asterisk
Hello,
I have a strange problem whenever I have 2 or more BT100s behind NAT. I am
not able to reproduce this error reliably, but it happens every 2-5 minutes.
The general setup is that there is Asterisk server sitting at a central
location. Some peers connect directly (206,205,201) but some
Hello,
So far everything that I'm trying with asterisk is working except for this
weird thing. When I try to call voicemail and it asks me for the password I
enter it in but from the debug message I can see that it thinks I didn't
enter anything in. Also when I'm leaving a message it sais press
:49:46 -0700, Tomas Florian [EMAIL PROTECTED]
wrote:
Hello,
So far everything that I'm trying with asterisk is working except for this
weird thing. When I try to call voicemail and it asks me for the password
I
enter it in but from the debug message I can see that it thinks I didn't
enter
: Tomas Florian [EMAIL PROTECTED]
To: [EMAIL PROTECTED], 'Asterisk Users Mailing List - Non-Commercial
Discussion' asterisk-users@lists.digium.com
Sent: Wed, 19 Jan 2005 15:27:44 -0700
Subject: RE: [Asterisk-Users] Asterisk not recognizing key beeps - DTMF on
BT100
Thanks, this is what I found out so
; Inbound calls from this host go here
mailbox=100 ; Activate the message waiting light if this
canreinvite=no
dtmfmode=rfc2883
allow=ulaw
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tomas Florian
Sent: Wednesday, January 19, 2005 9:33 PM
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