On Wed, 15 Oct 2014 09:14:41 -0500, Matthew Jordan
wrote:
>On Wed, Oct 15, 2014 at 1:50 AM, A.Santoro wrote:
>> Hi there,
>> I have installed Asterisk version 12.6 (on Debian wheezy) and I note
>> that, only when I make a transfer of call (attended or unattended),
>> the fields 'dst' and 'dconte
begin 644 cdrlog.txt
M6T]C="`Q-R`Q-SHU,3HT-5T@5D520D]315LT,S$T72!C9'(N8SH@(#!X,C8Y
M-V-C."`M($-R96%T960@0T12(&9O#(V.3=C8S@@+2!4"YC.B`@
M("`@+2T@17AE8W5T:6YG(%LQ,#9`96)A;&E?&5C=71I;F<@6W-`
M;6%C"YC.B`@("`@+2T@17AE8W5T:6YG(%MS0&UA8W)O+79O:6-E;6%I;#HR
M72!$:6%L*")325`O=F%L97)I;RTP,#`P,#`P,"(L(")325`O
Hi there,
I have installed Asterisk version 12.6 (on Debian wheezy) and I note
that, only when I make a transfer of call (attended or unattended),
the fields 'dst' and 'dcontex' in the CDR are empty.
This happen both in MySQL record and in CVS.
Someone can confirm this event?
Thanks in advance.
On Fri, 30 Apr 2010 18:52:46 +0200, Philipp von Klitzing
wrote:
>As I said, you could think about creating 4 different SIP gateways on the
>Patton with 4 differing SIP ports. I don't know if the Patton will handle
>4 gateways - but it might.
>
>> We have 4 trunk and 4 company in our office, I
On Fri, 30 Apr 2010 14:16:14 +0200, Philipp von Klitzing
wrote:
>Hi!
>
>> calls from Asterisk: when a call come from SIP/1001 (BRI 1 on Patton)
>> or SIP/1002 (BRI 2) or SIP/1003 (BRI 3) Asterisk record a call coming
>> from SIP/1004.
>
>Read up on how Asterisk does user/peer matching in sip.con
On Fri, 30 Apr 2010 10:39:12 +0200, Carlo Dimaggio
wrote:
>2010/4/30 A.Santoro
>
>> Hi,
>> we have and Asterisk server connected to a Patton Smartnode 4638 with
>> 4 BRI. [...]
>>
>Have you tried setting "insecure=port,invite" in the sip.conf for each sip
>account?
>
Hi Carlo,
thanks for your a
Hi,
we have and Asterisk server connected to a Patton Smartnode 4638 with
4 BRI.
We configured 4 SIP account on Patton (1001, 1002, 1003, 1004).
The system is fully functional, but we have a problem to recognize
incoming calls from Asterisk: when a call come from SIP/1001 (BRI 1 on
Patton) or SIP/1
On Wed, 30 Dec 2009 11:43:59 -0800, vijay.go...@alliance-infotech.com
wrote:
>
>case 2: This skype account (rexesbposolutions) has been assigned with a
>online virtual number (00 44 20 ). If somebody dial this number
>from their landline/cellphone, call is transfered to Asterisk queue bu
I installed festival following this guide (method 1)
http://www.voip-info.org/wiki/view/Asterisk+festival+installation
When I use english voices Festival works, when I change in Italian
voices, Festival return an error (generic).
Someone has faced and solved the same problem?
Thanks in advance f
