Hi;
It seems the SNOM Phones are requesting to have SRTP but I do not have the
module res_srtp.
I tried to compile it with asterisk 1.8, make menuselect, but I found that it
can not be used (I am not able to select it) with the following details:
Secure RTP SRTP
Depends on: srtp E
Can use:
On Wednesday 19 September 2012, bilal ghayyad wrote:
Hi;
It seems the SNOM Phones are requesting to have SRTP but I do not have the
module res_srtp.
I tried to compile it with asterisk 1.8, make menuselect, but I found that
it can not be used (I am not able to select it) with the
Hi List.
I'm trying to make a call transfer using Action: Bridge, the transfer is
made, but when one end disconnects the link back to the context CONTEXT
that was before it was done to BRIDGE.
example:
A is talking to B.
C is parking.
Brigde run over B and C, 100% B occurs is
I'm trying to make a call transfer using Action: Bridge, the transfer is
made, but when one end disconnects the link back to the context CONTEXT
that was before it was done to BRIDGE.
example:
A is talking to B.
C is parking.
Brigde run over B and C, 100% B occurs is talking to C.
Hi All!
i have a problem with asterisk 1.8.11.
I must have in the SIP cancel message, the line Reason
Example : Reason : SIP;cause=16;text=Normal Call Clearing
I have already enable use_q850_reason=yes, but this not work.
In my dialplan I have already add : exten = _X.,n,Hangup(${HANGUPCAUSE})
It looks like the answer is yes.
http://crazytechthoughts.blogspot.ca/2011/12/call-external-program-from-mysql.html
From the page, here is code to execute a UDF library and call a shell.
Clearly there would be a heavy penalty to launching a shell so you would
want to carefully evaluate the
David
The way we do this is to have a trigger insert into a batch table. This
table can be polled from a secondary process. That process/service is
responsible for monitoring, working and cleanup. This allows for you to
poll a highly optimized table without taking the db performance hit from
Hi,
we have a failover redundancy with two ISPs and an outside SIP Trunk
provider for the telephony. Since network failures of our primary ISP
line happen sometimes I would be interested into a solution where we
could use both IPs to connect to our telephony provider. I was talking
to them and
Many providers ofcer this... if yours do not, find a new sip provider.
On Sep 19, 2012 2:05 PM, Matej Mailing mail...@tam.si wrote:
Hi,
we have a failover redundancy with two ISPs and an outside SIP Trunk
provider for the telephony. Since network failures of our primary ISP
line happen
On Wed, 19 Sep 2012 14:04:33 -0400
David Cook dbc_aster...@advan.ca wrote:
It looks like the answer is yes.
I thought the answer was 42.
If you're going to top-post (which is against the rules on this list),
the least you can do is phrase your answers in a way that illustrates
the question.
Hello David,If you're going to top-post (which is against the rules on this list),I apologize if mistaken, but out of curiosity, can you please refer me to where in the rules it says that we can not top-post in this list?ThanksChristian SavinovichVoIP Telephony Consultant646-982-3572
On Wed, 19 Sep 2012 13:44:47 -0700
C. Savinovich c.savinov...@itntelecom.com wrote:
Hello David,
If you're going to top-post (which is against the rules on this
list),
I apologize if mistaken, but out of curiosity, can you please refer
me to where in the rules it says that we can not
If you're going to top-post (which is against the rules on this list), I agree with you. Just really wanted to locate where it was defined.Christian SavinovichVoIP Telephony Consultant646-982-3572
Original Message
Subject: Re: [asterisk-users] Trigger Asterisk after data
Hello all,
I bought a Cisco model of SPA-301, I saw that in the specifications it supports
SRTP for secure connections, but I'm not finding how to create a certificate
and where and how should I configure my Asterisk 1.8. I already use other ip
phones and these are the other Yealink T32G and
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