Displaying 9 results from an estimated 9 matches for "hovsep".
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house
2010 May 18
3
About option U in Dial Ast version 1.6.2
...ptions in
conjunction with this option. Also, pbx services are not run on the
peer
(called) channel, so you will not be able to set timeouts via the
TIMEOUT()
function in this routine.
Thanks
--
Vardan Harutyunyan,
Senior System Administrator
Enterprise Incubator Foundation
123 Hovsep Emin Street,
Yerevan 0051, Republic of Armenia
Tel: + 374 10 219735
Fax: + 374 10 219777
E-mail: info at eif.am
www.eif-it.com
2010 Dec 20
5
DIALSTATUS on CANCEL
Hello,
We have a strange situation (asterisk 1.6.2.14), where we get a result for
DIALSTATUS for BUSY and No-ANSWER, but nothing for CANCEL.
This is the (relevant) test dialplan:
--------------------------------
[incoming-private]
exten => _X., n, Dial(SIP/1001,30)
exten => _X., n, NoOp(${DIALSTATUS})
exten => _X., n, Gosub(incoming-status,s-${DIALSTATUS},1)
[incoming-status]
exten
2010 Sep 16
5
a2billing
Hey there,
I am trying to setup a2billing on asterisk 1.6 , but ,when I try to access its web page I see the a2billing directories:Index of /a2billingNameLast modifiedSizeDescriptionParent Directory -admin/15-Sep-2010 19:19-agent/15-Sep-2010 19:21-common/15-Sep-2010 19:18-customer/15-Sep-2010 19:20-Apache/2.2.9 (Debian) PHP/5.2.6-1+lenny8 with Suhosin-Patch Server at
Att,
Flavio Roberto
2010 May 21
3
CANCEL Reason
Hello all,
I need that Asterisk Always use Reason in a CANCEL.
How to do?
thank you
*Fran?ois *
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2010 Sep 28
1
1.6 and 1.8 version & A2Billing
Hi All;
Anyone has tried to use A2Billing with Asterisk 1.6 and 1.8 to confirm that is working fine and it is same as 1.4?
Appreciate ur kindly help.
Regards
Bilal
2010 Jul 27
2
CallerID disappear from CDR on transfer
Hi, i've some trouble with an * installation when the following scenario
happen.
1) Inbound call to SIP/xxxxxxxxxxxx ;
2) Call is redirected to ring group 6xx
3) SIP extension 1xx answer.
4) caller want to speak with john doe on his mobile
5) assistant put caller on hold
6) assistant start a call to john doe mobile using a php script (AMI -
Originate with custom context to force outbound
2010 Jul 12
3
need information
Dear All.
I want to become a wholesale VoIP traffic Provider , and i don't have a
experience about the software used this career .
I ask about Freeside billing system , FreeRADIUS AAA server and Asterisk
telephony server gave me all i need to start my business .
thanks
--
Best Regards
Mohamed Daif
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2010 May 19
2
a2billing DID and Queues
Hi all,
I have configured asterisk and a2billing.for inbound i have also configured
did and its forwarded to sip extensions.
But i want to enable queues with inbound numbers(DID).But i could not find a
way to do this in a2billing.
I want enable that if some did comes to asterisk/a2billing it should be
forwarded to queues not sip extensions and
their i want to enable hunting so if one
2010 Jun 12
2
Qwest PRIs
Hi,
I'm trying to bring up two PRIs from qwest with asterisk and dahdi. I'm
using an OpenVox D410E and the drivers are loaded. My system.conf looks
like this:
# Span 1: TE4/0/1 "T4XXP (PCI) Card 0 Span 1" B8ZS/ESF RED
span=1,2,0,esf,b8zs
bchan=1-24
# Span 2: TE4/0/2 "T4XXP (PCI) Card 0 Span 2" (MASTER) B8ZS/ESF RED
span=2,1,0,esf,b8zs
bchan=25-47
dchan=48
These