I am trying to configure * to route calls from SIP extension to an
externeal H.323 gatekeeper and vice versa.
The route from * to the gatekeeper is a simple ENUM call and work fine:
[outbound][outbound]
exten => _3XXX,1,Dial,H323/[EMAIL PROTECTED]
One Snom100 phone is defined in sip.conf:
[snom]
going to test it with an network link layer simulator (Adtech
SX-20) because we got registration timeout over a VSAT satcom connection
(~560 ms RTT).
Beside the Adtech COTS youc an use the NIST Net nework emulator
(http://snad.ncsl.nist.gov/itg/nistnet/).
regards
Olaf
--
Dipl. Ing. Olaf Menzel
Hi all,
thank U all for your very fast response. I want to clarify that my
question was not regarding about the fasibility of
Voip over satellite in general but especial the behavior of the Asterisk
PBX on a long delay path. We just successfully tested
H323 Voip with a Innovaphone IP 20
satellite. I want to registser at
the local Asterisks and
only want to send the Voip (RTP) traffic over satellite. I SIP I can
dial any user without
remote registration. Why can't I just reach the registered snom phone by
just dialing his
sip address (sip:[EMAIL PROTECTED]) ? Any suggestion
ten => 2002,102,Voicemail,b2002
exten => 2103,1,Dial,SIP/snom3|30
exten => 2003,2,Voicemail,u2003
exten => 2003,102,Voicemail,b2003
exten => 2104,1,Dial,SIP/snom4|30
exten => 2004,2,Voicemail,u2004
exten => 2004,102,Voicemail,b2004
Hello Tilghman,
--
> IAX is not an IETF standard. However, it is an open protocol (i.e.
> not proprietary), as all source is offered under an open source
> license. You may also license the IAX code with another license by
> contacting Digium, or you may use a clean room appr
On Friday 27 June 2003 16:23, Angelo Sampietro wrote:
> hi,
> can some one tell me a good IP phone (not software, but a "real"
> phone :) that work well with asterisk?
Hi Angelo,
---
I am testing the Snom100 and Snom200 phones. Both working fine under Asterisk.
(http://www.snom.com)
Hello,
-
In several VoIP projects I have very often been asked if IAX is already an
IETF stanadard. I could not anwer this question and I could not find any
IETF draft about this protocol. Can you point me to the right location or do
you know if Digium is going to publish the IAX prot
On Friday 20 June 2003 13:28, Michael Manousos wrote:
> Are there any suggestions for active ISDN CAPI PCMCIA cards
> that are known to work with Asterisk?
>
You can try AVM B1 PCMCIA. This card is fully I4L compliant but AVM has
developed a LINUX capi 2.0 stack.
http://www.avm.de/en/products/ha
Hi all,
I am sorry to ask you this newbie question again but I did not get any answer:
If I have installed a asteriskpbx at the public internet what would be my own
sip address ? I guess it depends on the callerid . ??
sip.conf
[phone2]
type=friend
host=dynamic
dtmfmode=rfc2833
mailb
Hi all,
I have a Asterisk at a public Network (official IP address). In the local
network I have isntalled a Snom 200 IP phone and in my home network (behind
NAT) a Snom 100 device. I can dial the Snom200 device from my home location
without any problems but the Snom200 can not dial m
Hi all,
---
how can I use SIP softphones together with asterisk such as kphone, siset,
... for LINUX wich do not support number dialing. From my Snom 100 it was
easy to dial the softphones, but it did not work in the opposite way.
To Dial phone 2 I have just pressed "2" at the Som100 and I
Hi John,
thank you for pointing me to to some of the additional Asterisk documentation
stuff.
>
> http://www.digium.com/handbook-draft.pdf
>
> In addition, there are a variety of home-built pages.
> http://www.automated.it/guidetoasterisk.htm
> http://asterisk.gnuinter.net/
>
I foun
h of the Digium's devices you
propose for a normal EuroISDN line (2 B-channels) ? I have attached a small
figure for my SIP configuration aaproach.
Thank you for your help
regards
Olaf
--
Dipl. Ing. Olaf Menzel - System Engineer
FOKUS - Fraunhofer Institute for Open Communication Sy
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