Displaying 6 results from an estimated 6 matches for "multiplecallers".
2004 Dec 06
0
Kind of off-topic: VoIP services and multiplecallers
> -----Original Message-----
> service is compatible with Asterisk). However, I have a question: can
> more than one person make/receive a call at the same using one VoIP
> line?
>
> If five people in the office all need to use their phones at the same
> time, would I need five VoIP lines, or would I only need one
> VoIP line?
> Am I over-thinking this?
No,
2004 Dec 06
2
Kind of off-topic: VoIP services and multiple callers
Hello Everyone,
I've been running Asterisk as our PBX for several months now, and
recently I've been thinking about using one of the VoIP providers to
lower our phone bill.
I know that VoIP providers can supply their customers with a local
number and/or virtual numbers, and then that number/account can be used
with Asterisk (well, it depends on the provider and whether or not their
2005 Mar 03
0
I have met a message : "No one is available to answer at this time".
Hello, Users.
I loaded module chan_h323.so, chan_vpb.so.
I have met a message : "No one is available to answer at this time".
I don?t know what I do..
My 'h.323 trace 5' result is :
== vpb/1-8: Starting record mode (codec=0)[AST_FORMAT_SLINEAR:VPB_LINEAR]
-- Executing Dial("vpb/1-8", "h323/192.168.1.107") in new stack
1:21:34.936 ThreadID=0x06f2bbb0
2010 Sep 25
0
can call internal branch , but can not call external numbers with avaya , always get return message : Q931IncompatibleDestination
Hi Gurus,
We have configured asterisk to trunk with avaya with ooh323 channel driver. The sip phone registered on asterisk
can dial the extensions registered on avaya via this trunk , and vice versa works too. Even we can make the avaya branch to dial asterisk?s extension and then this extension dial back to another avaya?s extension.
But if we dial the external DID number via this trunk from
2005 Jan 28
3
reason 24 (Call ended with Q.931 cause)
Hi Michael and Everyone
I'm trying to connect Asterisk to a CISCO AS5350 using oh323 and I'm getting
this error
"reason 24 (Call ended with Q.931 cause)"
I've checked the Asterisk wiki and several other resources. Please can
anyone give me a hint on what the problem is I reach my wits end. Thanks
Tola
my config and debug
Configuration of OpenH323 channel driver
2004 Jun 29
5
SIP->Asterisk->GnuGK->Cisco 5300
Hi all,
I would like to call from SIP client to Asterisk then GnuGk, then Cisco 5300
to PSTN phone. Is this possible? I need simple config asterisk and gnugk.Can
somebody help me?
Ganbaa
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