Hello everyone.
I am looking for a E1 PRI card which supports network side signaling not
CPE. The main idea is to connect an plain old E1 compliant PBX which
doesn't have an VoIP module to the newly created VoIP infrastructure.
Could we use a Digium TE122P or something other to resolve this
Use pri_net as signalling mode inside chan_dadhi.conf in /etc/asterisk/
folder.
You can set this up using any pri card thats supported on Asterisk.
Mitul
On Mar 31, 2013 12:25 PM, Dimitar Dimitrov ddimit...@consult.bg wrote:
Hello everyone.
I am looking for a E1 PRI card which supports
In article caaogpgr4y2vyndtu3nsnlct_6t1vdojsgouh634ec+zfogp...@mail.gmail.com,
Mitul Limbani mi...@enterux.in wrote:
On Mar 31, 2013 12:25 PM, Dimitar Dimitrov ddimit...@consult.bg wrote:
Hello everyone.
I am looking for a E1 PRI card which supports network side signaling not
CPE. The
Thank you guys for the fast response.
I will try that.
Thanks.
Dimitar
On 03/31/2013 11:15 AM, Tony Mountifield wrote:
In article caaogpgr4y2vyndtu3nsnlct_6t1vdojsgouh634ec+zfogp...@mail.gmail.com,
Mitul Limbani mi...@enterux.in wrote:
On Mar 31, 2013 12:25 PM, Dimitar Dimitrov
Hi, asterisk admin and users.
I need to SIP INVITE uri with domain via peer. And uri domain differ
then peer domain.
dialplan:
exten = s,n,Dial(SIP/peer1/num...@domain2.com,60,r)
[peer1]
type=friend
host=domain1.com
fromdomain=domain1.com
As a result in SIP packet uri:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
I'm running Asterisk 11.3.0 on wheezy.
I'm trying to do TLS +SRTP with blink SIP clients as shown here
https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial.
TLS is fine and I can call between clients. SRTP is a different matter,
On 31 March 2013 18:11, Dmitriy Serov serov@gmail.com wrote:
Hi, asterisk admin and users.
I need to SIP INVITE uri with domain via peer. And uri domain differ then
peer domain.
dialplan:
exten = s,n,Dial(SIP/peer1/number@**domain2.com num...@domain2.com
,60,r)
[peer1]
type=friend
in advance.
Dimitar
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Message: 2
Date: Sun, 31 Mar 2013 12:50:09 +0530
From: Mitul
On 17/12/12 13:34, Joshua Colp wrote:
Barco You wrote:
Dear All,
Hola,
I use sipml5 to register two users from browser and the two clients
are successfully connected. But when I made a call from one of the
users, the other user doen'st have call notification and for a while the
Daniel Pocock wrote:
I'm trying to call from DruCall to Asterisk and I get this error:
WARNING[11021]: chan_sip.c:8687 process_sdp: Error in codec string 'F
103 104 111 0 8 107 106 105 13 126'
== Problem setting up ssl connection:
error::lib(0):func(0):reason(0)
I'm guessing my
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