Use SIP.
PaulH
- Original Message -
From: <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Friday, November 18, 2005 7:53 AM
Subject: [Asterisk-Users] SIP/H.323 HardPhones
> hi,
>
> Do anyone know a low-cost, simple SI
hi,
Do anyone know a low-cost, simple SIP or H.323 hardphone that works well
with Asterisk?
Jan
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hello,
I using asterisk-oh323
i have a problem for send dtmf
see log error:
reason 24 (Call ended with Q.931 cause [28 - Invalid number format])
thanks for you help
2005/11/15, Martin Vit <[EMAIL PROTECTED]>:
> i would recomend this channel for h323:
> http://www.inaccessnetworks.com/proje
i would recomend this channel for h323:
http://www.inaccessnetworks.com/projects/asterisk-oh323
Abdul Lateef wrote:
Hi all,
I have H.323 Terminator and i want to terminate our
all SIP clients to this terminator, Is it possible to
add H.323 Terminator in Asterisk?
Please give me a little hint o
Hi all,
I have H.323 Terminator and i want to terminate our
all SIP clients to this terminator, Is it possible to
add H.323 Terminator in Asterisk?
Please give me a little hint os i can start to
configure.
Yours,
Abdul Lateef
Computer Programmer
HATIF COM
Mob: +974 - 5405022
Tel: +97
HI all,
Is Asterisk able to work as SIP and H.323 Gatekeeper
same time?
If it has the capability to work which i should open?
Yours suggestion will be high appriciated.
Yours,
Abdul Lateef
Computer Programmer
HATIF COM
Mob: +974 - 5405022
Tel: +974 - 4883068
ICQ: 276994704
YM!: abdul
Try playing with faststart .
Moises Silva wrote:
Hi Carlos. I have never used H323. But im interested in your problem.
Have you tried to use de 'sip debug' 'iax2 debug' commands? and check
the console with a high verbosity level? could you post any warning or
relevant output when the call is ma
Hi Carlos. I have never used H323. But im interested in your problem.
Have you tried to use de 'sip debug' 'iax2 debug' commands? and check
the console with a high verbosity level? could you post any warning or
relevant output when the call is made?
best regards
On 6/11/05, Carlos Alberto Lara de
Greetings to the list:
this is my problen when I make a call from my asterisk towards a nortel
PBX , the call is made but in my telephone sip I do not listen the dial tone
or the busy tone but the call it is completed normally.
sip-phone-g729-asteriskh323-g729---
> Who is going to accept the SIP calls, Asterisk ou SER?
Either one of them can accept/register SIP agents, although what I
have been reading SER is more scalable for large implementations.
> Who is going to redirect calls to PSTN? Asterisk or SER?
Only Asterisk can interface with PSTN using spec
Hello
I want to have a server that supports SIP and H.323
endpoints calling eachother, so I installed GNUGK, SER and
Asterisk.
GnuGK is needed for H.323 endpoints and users
Accounting, Authentication and Authorization.
SER is needed for the same 3 A's but for SIP
users.
Asterisk has some goo
Hi Ryan!
Interesting what experience you have made in this issue.
We have setup the alternative channel for H.323 (the * built in
chan_h323), and we are now in a testing phase.
I was wondering (in case no transcoding is needed), how your setup treats
the RTP streams. Do the RTP streams go end-to
Yes, it can.. I'm doing it at my home. My current setup is
Asterisk-1.0-RC2 using the oh323 driver. I have a SIP connection to
Broadvoice talking to Asterisk. I have a e-tel (now Qtelnet) H.323 VoIP
telephone adapter as my end point talking to Asterisk.
For processing sake, you may want to keep
Hi,
is there a definite answer if asterisk can pass calls between SIP and h.323
protocols?
Thanks,
Yiannis.
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Hi all!
Are there some people who have already implemented a SIP - H.323 Gateway? I
am trying to do so... but I don't know how. Please if anyone can help me...
Thanks a lot for all your answers.
Mireia
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Good day,
anyone had success finding a SIP or H.323 phone with intercom
capabilities? Softphone would be preferred?
TIA
rgds
pos
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