Kevin,
Thanks for the reply.
On 29 June 2012 00:29, Kevin P. Fleming wrote:
> On 06/27/2012 09:30 PM, David Cunningham wrote:
>
> Would anyone else know if Asterisk allows use of SpanDSP's time zone
>> conversion?
>>
>
> No, SendFAX (in res_fax) doesn't currently offer the ability to do what
On 7/2/2012 5:15 PM, Carlos Alvarez wrote:
We are a hosted PBX service provider using Asterisk (primarily 1.6,
moving to 1.8 soon). In the past, when we've been asked to provide call
recording, we deploy a custom server just for that customer. I'd like
to bring call recording to our standard ho
We are a hosted PBX service provider using Asterisk (primarily 1.6, moving
to 1.8 soon). In the past, when we've been asked to provide call
recording, we deploy a custom server just for that customer. I'd like to
bring call recording to our standard hosted system so we can provide it at
a lower c
http://goo.gl/XTjqx--
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Has anyone attempted to use an Allworx 9212 handset with an Asterisk PBX?
M. Hutter
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So take a look here:
http://www.voip-info.org/wiki/view/stream+file
Am 29.06.2012 16:06, schrieb CDR:
This is from the documentation of Perl-AGI
"$AGI->stream_file($filename, $digits, $offset)
Executes AGI Command "STREAM FILE $filename $digits [$offset]"
This command instructs Asterisk to play
actually its a one-way audio issue due to NAT !
alok , please explain your network flow for end to end client-server-client.
You may need to set nat=yes for your sip peer behind NAT. If the server is
behind NAT router/firewall use externip= field.
Also provide sip traces of this call.
Another th
alok srivastava wrote:
dear
i have configured properly asterisk. At the one end i am using x-lite
soft ph and another end twinkle. call is going properly from both end
but after picking the phone not able to listen other one.
when i checked the port 5060 on the asterisk server it is always show
On Fri, 2012-06-29 at 17:27 +0100, Chris Bagnall wrote:
> On 29/6/12 9:59 am, Ishfaq Malik wrote:
> > Does anyone have any experience of connecting SIP phones to an asterisk
> > server through the 2701HGV router that BT supply with their Infinity
> > product?
>
> Good luck with that.
>
> The BT '
1 jul 2012 kl. 09:48 skrev Leandro Dardini:
> Port 5060 when used with the sip protocol is used witj UDP protocol. Telnet
> is using TCP.
That's not correct. SIP supports multiple transports, including TCP. Not all
implementations support TCP though.
/O
--
28 jun 2012 kl. 23:25 skrev Duncan Turnbull:
> Hi James
>
> On 29/06/2012, at 6:19 AM, James Lamanna wrote:
>
>> Hi,
>> I have a bunch of different customers on an Asterisk Box (the PBX).
>> This Asterisk Box is behind another Asterisk box that provides a PSTN
>> connection.
>> Up to this point
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