When you set up a new mailbox, the program copies default files. Just
overlay those files with what you want. Look in apps/app_voicemail.c for
guidance.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Justin Killen
Sent: Friday, Nove
Is there a way to specify default files to use for new mailbox creations? For
example, when a mailbox's directory structure is created, there is no greeting,
unavailable, or busy messages, so the incoming calls get the message: "The
person at extension XX is not available". I'd like to be
> my scenario is below
>
> analog phone (10 to 99)--> pbx-->(77)asterisk>
> jitsi(2000)
>
> i have analog telephone interface numbered 77 attached with asterisk
> and
> other sip user is 2000 on jitsi.
>
> I can call from any number from 10 to 99(in intercom) on 77 and ivr
> resp
> I used libpri 1.4.12 version with asterisk 1.8.7, after the pcap
> files, i found that
> in wireshark setup message, the number type always changed from
> subscriber to national number.
> i have set pridialplan= local and prilocaldialplan=local in
> chan_dahdi.conf already. because that, the syst
On Fri, Nov 30, 2012 at 04:54:28PM +0530, Harish Mandowara wrote:
>
> Do not bother about below message. That is auto-generated by my mail
> server.
[snip]
> ---
>
> This
Daniel - Asterisk wrote:
Thank you Carlos,
What does mean 'por-out'?
I'm expecting 1 min/month in & out.
Elder
PORT out = Port or move the number away from a provider
Seems this provider is unaware that one may have moved the number to another
provider, and continues to charge when they
On Fri, Nov 30, 2012 at 7:10 AM, Daniel - Asterisk wrote:
> Thank you Carlos,
>
> What does mean 'por-out'?
> I'm expecting 1 min/month in & out.
>
Port out means a number was ported to another carrier.
10k minutes is not huge but a decent number that should get you a
reasonable rate with th
Thank you Carlos,
What does mean 'por-out'?
I'm expecting 1 min/month in & out.
Elder
On Thu, Nov 29, 2012 at 5:50 PM, Carlos Alvarez wrote:
>
> On Thu, Nov 29, 2012 at 3:22 PM, Daniel - Asterisk
> wrote:
>
>> Hello List,
>>
>> Since I'm looking for a new VoIP provider for US origination/
Does anyone have information or successfully connected Asterisk
to TalkMaster from Digital Accoustics?
Thanks,
Jerry
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Hi,
my scenario is below
analog phone (10 to 99)--> pbx-->(77)asterisk> jitsi(2000)
i have analog telephone interface numbered 77 attached with asterisk and
other sip user is 2000 on jitsi.
I can call from any number from 10 to 99(in intercom) on 77 and ivr
response will come th
On Tuesday 27 November 2012, Adolphus Enaboifo wrote:
> Hi List members,
> Thanks for the support so far as I try to install and test my first
> asterisk system.
> I was able to finally install asterisk-1.8.18.0 with libpri-1.4.13 and
> dahdi-linux-complete-2.6.1+2.6.1 according to the instructions
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