Hi All,
I believe that a lot of the Sound Point Success story here. Can some one
kindly let me know how to set up the Polycom Phones Sound Point or any
reference to refer from...
Your kindly help are appreciated.
Thank you.
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Hi all,
Any polycom phone v1.6 IP301 references? I had purchase
three new phone and I cant connect them into Asterisk 1.2.11.
I do appreciate if some one can point me how and where ?
Thank you
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Hi James,
>James wrote;
>When I hit 9 on the
siemens it does not get a dial tone from asterisk, I assume this is
>because I have not told
asterisk to give it one.
I might be wrong;
My question is, are you sure
your ISDN ( Asterisk span to Siemens ) is up logically?
ISDN is no tone gi
Hi "J.
Oquendo" <[EMAIL PROTECTED]>,
What LAN switch that you are
using, and what type of IP phones that you are using?
>I've set up a "ring
all" context on my gateway on extensions.conf:
>[EMAIL PROTECTED] ~]# grep
"*7" ast/extensions.conf
>exten =>
*7,1,Dial(SIP/1201&SIP/1202&S
If it’s control by
serial port, you need TAPI license, and need some investigation period to
integrate with Nortel system. ( you have to pay for the license, if your system
can do so. )
Cheers.
>Andrew:
>The key here is try to
create a way to integrate Asterisk Voicemail with
Subject: [asterisk-users] Slow dialing from PBX via E1
Hi :)
I have a 'slow dialing' problem. When I dial 200# for the
'echo test' application from my PBX extension 1010, I see this
in the console the instant I press the # key:
-- Starting simple switch on 'Zap/65-1'
-- Accepting overlap c
(Andrew Kohlsmith) wrote:
Re: MWI from Asterisk to Meridian
"So, as I said, you are stuck using a Nortel ATA and an FXS port on Asterisk
and using a hookflash *1 sequence to toggle it. Unfortunately the VM
callback # will be the ATA's DN, so only one person at a time can access
voicemail."
Hi Cesc,
Alcatel does not offer any SIP Terminal, it phone's are H.323 it's handle
for their own features usages. ( like ADSI type )
From: Cesc <[EMAIL PROTECTED]>
Subject: [asterisk-users] alcatel ip touch 4068 ... sip?
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Mes
Hi all,
I got the SPA-2100 and I can dial call from other extension to this SIP
ATA, however I got problem make out the call the Asterisk, I believe something’s
to do with the Dial Plan in side the SPA-2100
configuration file.
My setup simple topologies,
ATA connected to my LAN,
Hi all,
I had some doubt about meetme application, hope that some
one can tell me what to do?
(No GUI and CLI in this
case)
Say 5 user in the conference room 1000. Ofcoz, each user is
holding userID in the conference room.
Say user 1 SIP/200
Say user 2 SIP/201
3 SI
Hi all,
Can some one provide me the infor about polycom phones model that compatible
and stable to work with Asterisk? I intend to purchase IP 300, and IP
501 models.
Tq
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