Tom Moore schrieb:
What are your suggestions to people who have pbx systems that interface with
the world over pri and want to convert them to sip interfaces so that they
can use sip trunking?
I'd go for a Patton SmartNode. See www.patton.com - they have SIP
gateways up to 4 T1/E1.
Christian
: Wednesday, August 27, 2008 9:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Pri to sip interfaces
Are you looking for a hardware suggestion or a software suggestion?
PaulH
Tom Moore wrote:
No, these are mainly Samsung pbx systems.
I know I can use
Hi guys,
What are your suggestions to people who have pbx systems that interface with
the world over pri and want to convert them to sip interfaces so that they
can use sip trunking?
Tom
___
-- Bandwidth and Colocation Provided by
Are you using an Asterisk PBX?
_
Darren Sessions
[EMAIL PROTECTED]
http://www.darrensessions.com
_
On Aug 27, 2008, at 7:06 PM, Tom Moore wrote:
Hi guys,
What are your suggestions to people who have pbx systems that
interface
Asterisk.
PaulH
Tom Moore wrote:
Hi guys,
What are your suggestions to people who have pbx systems that interface with
the world over pri and want to convert them to sip interfaces so that they
can use sip trunking?
Tom
___
-- Bandwidth and
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Pri to sip interfaces
Are you using an Asterisk PBX?
_
Darren Sessions
[EMAIL PROTECTED]
http://www.darrensessions.com
_
On Aug 27, 2008, at 7:06 PM
Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Pri to sip interfaces
Are you using an Asterisk PBX?
_
Darren Sessions
[EMAIL PROTECTED]
http://www.darrensessions.com
_
On Aug 27, 2008, at 7:06 PM, Tom Moore wrote
*From:* [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] *On Behalf Of *Darren
Sessions
*Sent:* Wednesday, August 27, 2008 9:22 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Pri to sip interfaces
Are you
of PBX you have ?
- Allow PRI to SIP trunking failover and vise versa
Good luck...By the way , where is your location ?
--- On Thu, 8/28/08, Tom Moore [EMAIL PROTECTED] wrote:
From: Tom Moore [EMAIL PROTECTED]
Subject: [asterisk-users] Pri to sip interfaces
To: 'Asterisk Users Mailing List
For PRI you have 3 main solutions. This is the order of stability (and
pricing):
1. Digium or Sangoma cards use the computer processor and that could be
bad if you have huge traffic through the PRI
2. Eicon Diva cards have their own processor, which releases the PC
processor and gives more
Or any of a number of gateways that do this. Off the top of my head
you can get one from CarrierAccess, Vega, Audiocodes, Mediatrix,
Adtran, and others.
Just try to be very careful as they all have their strengths and
weaknesses and you need to evaluate how they would fit your needs.
Best
did you use T38 with you patton smart node 2400 ?
why Patton are very good GW and fax must work.
you must also check that the clock source is the Primary and not the
internal clock...
2006/12/14, Jerry Jones [EMAIL PROTECTED]:
Or any of a number of gateways that do this. Off the top of my
Hi
Can someone recommend a PRI to SIP Box that work well with asterisk
We are presently testing with a Patton Smartnode 2400 but we are unable to
fax through it.
We don't want to use digium card in a linux box for the PRI connection.
Which Cisco box would work.
Thanks
Patrick
Virtually any Cisco device from a 2610 up will work. 2610, 2620, 2811,
2821, 3640, 3700, 3800. I have 2610 and 3640 in production for 2+ years
with no issues.
Patrick Fortin wrote:
Hi
Can someone recommend a PRI to SIP Box that work well with asterisk
We are presently testing with a
Hi guys,
this is the scenario:
PRI -Asterisk-SER
If I call from a Sip(SER) user everything is good, I can call
anywhere, but if I try to call from outside(PRI) everything is
wrong!!!
This is the CLI for an incoming call:
--
ast*CLI
-- Executing SetCallerID(Zap/14-1, outside) in
Am Montag 14 November 2005 17:22 schrieb FaberK:
Hi guys,
this is the scenario:
PRI -Asterisk-SER
If I call from a Sip(SER) user everything is good, I can call
anywhere, but if I try to call from outside(PRI) everything is
wrong!!!
This is the CLI for an incoming call:
--
Hi Jens,
this is my sip.conf
---
[general]
context=default
fromdomain=192.168.1.188
port=5060
bindaddr=0.0.0.0
localnet = 192.168.1.0/255.255.255.0
srvlookup=yes
disallow=all
allow=alaw
allow=ulaw
allow=gsm
language=it
register = 1:[EMAIL PROTECTED]:5060/s
[1]
type=peer
username=1
fromuser=1
Hello All,
PRI to SIP calls are being destroyed after a few minutes, and I get the
stream below in a full debugging log. SIP to SIP works okay.
I'm using Asterisk CVS head 10-31-2005 with a TE110XP card set to T1.
Actually two TE100XPs are installed, but I only have one T1 plugged in
Hello all,
I have a problem calling into asterisk on a PRI going out to a SIP phone
(PRI - SIP). The calling party does not hear ringing and after about five
seconds gets an *All circuits are busy* recording. However, the called SIP
phone does ring, and if the called party answers the phone
Yes, many people have had this problem.
Check the mailing list archives... I think the newest code has the fix.
Workaround for older versions is to Answer before Dial, but you may
still need the 'r' option to Dial as ringing may stop for the caller
after about 10 seconds.On 10/27/05, OTR Comm
: [Asterisk-Users] PRI to SIP Problem
Yes, many people have had this problem.
Check the mailing list archives... I think the newest code has the fix.
Workaround for older versions is to Answer before Dial, but you may still
need the 'r' option to Dial as ringing may stop for the caller after about
10
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