CDR wrote:
Dear friends
This is my simple dialplan
[demopjsip]
exten = _X.,1,Dial(PJSIP/${EXTEN}@10.10.10.2)
exten = _X.,n,Hangup()
Currently the default outbound endpoint is only used for messaging. This
may change in the future but as of right now it is not used for
sessions. You will
We would like to present a toll free CallerID when making outbound toll
calls. In the past, when our PRIs were directly connected to a Nortel
CS1000 we could do this, without issue. Now that the PRIs are front ended
by a mediagateway facing asterisk, we can no longer do this.
Is it possible to
The following is a basic syntax of what you need to do before the Dial
function is launched:
Set(CALLERID(num)=1234567890)
More details is here:
http://www.voip-info.org/wiki/view/Asterisk+func+callerid
Regards
HASSAN
On Thu, Jun 26, 2014 at 8:10 PM, Positively Optimistic
It depends on your carrier.With some carriers, such as Verizon SIP, you do
this using P-Asserted-Identity. With Verizon SIP, if they can’t figure out how
to bill the call, it will be rejected.
With Level 3 SIP, you can use From: or PAID but if the number you present to
them is not on your
Thanks Hassan,
That is precisely what we are doing... if we
Set(CALLERID(num)=8002211212) the call fails.. however, if we
Set(CALLERID(num)=256963800)
it is successful.
The call will complete if we use any valid non-toll free number in the
LERG. If we use a toll free, it fails.
On Thu,
You need to talk to your carrier.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Positively
Optimistic
Sent: Thursday, June 26, 2014 10:23 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users]
We're using a Earthlink PRI converted to SIP via a MediaGateway. I assume
the mediagateway will convert the headers to something that PRI can
understand.
On Thu, Jun 26, 2014 at 9:22 AM, Eric Wieling ewiel...@nyigc.com wrote:
It depends on your carrier.With some carriers, such as Verizon
Hi All,
In asterisk, default directory to store the call-recording files is
/var/spool/asterisk/monitor.
Can we change this directory? How?
--
Anurag Rana
http://newbie42.blogspot.in/
On the trampoline of life's experiences, Striving towards a saintly life in
the midst of these materialistic
Hi All,
There is an option of starting the recording of call after the call is
bridged. [ b option].
Is there any way of running an AGI script only if call is bridged otherwise
not.
Thanks
--
Anurag Rana
http://newbie42.blogspot.in/
On the trampoline of life's experiences, Striving towards a
In a PJSIP endpoint, how do I set all no-named settings so they get
inherited from another place and I don't need to mention them again
and again for all my endpoints?
In regular sip you could specify those options and they remained valid
if not redefined by a peer. A case would be the codecs
You can use templates.
Templates are defines by putting a “(!)” next to the context name.
[template-name](!)
disallow=all
allow=ulaw
Then define the template next to the endpoint in parenthesis.
[endpoint-name](template-name)
Chad
On Jun 26, 2014, at 12:30 PM, CDR vene...@gmail.com wrote:
In
I did what we use to dim that is add a line to pjsip.conf like
#include /etc/asterisk/pjpeers.conf
but the file is not loaded. Am I doing something wrong this
functionality is disabled?
--
_
-- Bandwidth and Colocation Provided
Hi Anurag.
I didn't undertand much you question. But you have a dial option to a
macro when b answers
example...
exten =
_+2XX!,n,Dial(SIP/sip.XX/+${DESTINO},20,rgM(acceptcall^${SESSIONID})S(${MAXCALLTIME}))
[macro-acceptcall]
; this macro is executed when b answers, requesting b if is
Thanks Rafeal. This is what I needed.
But first line i.e.
exten = _+2XX!,n,Dial(SIP/sip.XX/+${DESTINO},20,rgM(acceptcall^${
SESSIONID})S(${MAXCALLTIME}))
is very complicated.
I have very simple plan which is as below.
[context-demo]
exten=,1,AGI ( pythonscript.py )
Ok.
in this link you will find some easy macro
http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial
Explanation
exten = _+2XX!,n,Dial(SIP/sip.XX/+${DESTINATION},20,rgM(acceptcall^${
SESSIONID})S(${MAXCALLTIME}))
Dial comand
to number DESTINATION
with timout of 20 seconds
r=ring sound
g= when
I am using AMI to Originate a call.
I have been able to get the caller id number to be passed through.
However, I can't get the name to be passed through.
A person I'm working with has a Freeswitch that is able to pass the caller id
name and number through for their call.
Comparing the Asterisk
There seem to be a number of places number presentation could go wrong.
Since the PRI 'used to' show toll-free numbers correctly, you need to
look at the gateway.
Can you debug the ISDN message on the gateway? See how the toll-free is
being sent to the carrier.
Since you are looking to
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