Be careful of LI requirements in Australia.
You MAY be able to put the onus for this on your upstream (PRI/IMT)
provider, but if you have many, this could be messy.
Best bet would be to have a solution yourself... when I was looking into
this the good news was that the enforcement agencies
What
do you want to test?
Call
routing under certain failure scenarios or CAMA trunking?
We
tested 911 to a PRI not connected to the PSTN that terminates on
another gateway (back to back PRI) and make a dedicated handset ring using a
dedicated pass through dial-peer. That way you can
Oh well... at least no one here thought E911 was 911 for IM or email
(yes... someone once asked me that)
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Joe Greco
Sent: Wednesday, January 19, 2005 3:43 PM
To: [EMAIL PROTECTED]
Cc:
Just beware of the effects of changing sample size for any codec.
We found that a sample size of 2 for G.711 (ie 2x20ms) allowed for
pretty robust interoperability between vendors. Not specifically with
Asterisk, but we did find that using a mixed CPE/gw environment with a
couple of Call Agent
In a previous company, we had issues with selling Fax-o-IP services
varying from the ability of CPE to support the correct NSE values and
recognize the difference between G3 (standard fax) and Super G3 (more
like a modem fax) through to the gateway's ability to turn echo-can's
on and off based on
I think you need to look at a few other factors.
1. Some IP phones are really flakey (had some serious issues with a
couple of vendors MGCP Business line package).
2. Line power - Cisco uses one standard, other phones use another... but
Cisco is the 900# gorilla in the powered switch market...
Smartbits ?
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Me
Sent: Monday, January 24, 2005 12:33 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Network Test Tool?
We have been having WAY too many issues lately
It could be a number of things... if you are running RFC2833 then some
sounds on the line (high-pitched voice?) can be interpreted as DTMF.
Also beware of some gateways and using SUB/NOTIFY, had a couple of
instances when the DTMF was supposed to be stripped from the in-band
(RTP) but wasn't...
Depending
on the switch they are using, there are a limited number of D-channels (or
D-channel licenses).
CAS signaling
needs RBS (its the winking in this case).
-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt Beebe
Sent: Monday,
I think of *, Broadworks, Vocaldata, Sylantro as line side feature
servers, and SS7 signaling with say IMTs/PRIs more for the class5
network side soft-switch (NexVerse, SONUS etc).
Typically they handle the LERG, complex translations etc and do it quite
well (although typically they take in
Yep, still lineside... you can do it with SIP too. If it was going to do
MGCP, it only makes sense if it does it properly and IS aware of the
channels on the other side of the gateway (multi-chassis trunk failover
etc)
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
Discussion
Subject: RE: [Asterisk-Users] T1 EM vs PRI question
Responses embedded below!
On Mon, 2005-01-24 at 18:49, Keith Burns wrote:
Depending on the switch they are using, there are a limited number
of
D-channels (or D-channel licenses).
CAS signaling needs RBS (it's the winking
Yep, I could buy it in Australia, install it in a * box, and deploy it
in, Fiji for instance (if legal there)
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Dave Green
Sent: Monday, January 24, 2005 6:10 PM
To: Asterisk Users Mailing
Title: AMP with SUSE 9.2
Hi,
I have the newbie guide from AMPs website and (fair enough) it is all about whitebox linux. Has anyone found any gotchas with the newbie guide relating to SUSE 9.2 ?
Any help appreciated.
Cheers
___
Asterisk-Users
Keith Burns wrote:
*Hi,*
*I have the newbie guide from AMP**'**s website and (fair enough) it
is
all about whitebox linux.** Has anyone found any gotchas with the
newbie
guide relating to SUSE 9.2 ?*
Please post to the amportal mailing list:
http://lists.sourceforge.net/lists
] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Keith Burns
Sent: Tuesday, January 25, 2005 9:43 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] AMP with SUSE 9.2
Cool, will do, thanks!
-Original Message-
From: [EMAIL PROTECTED
:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] AMP with SUSE 9.2
One thing for sure it was a rela headache for me... I finaly did get
it
working... Don't for get to follow the installation guide to a t...
Charles
On Tue, 25 Jan 2005, Keith
Title: AMP with SUSE9.2 (Apache2)
Hi all,
After pinging the AMP userlist at SourceForge, I got a great step by step explanation as to how to set up AMP for Apache2 (some maybe obvious stuff that wasnt in the Newbie Guide).
Thanks to Jason Becker of Coalescent Systems.
If anyone needs me
Title: X-lite to Cisco ATA - no RTP
Hi there,
I have X-lite and a Cisco ATA on the same hub (i.e. no NAT, no ACLs) as my Asterisk box.
Ethereal shows normal SIP signaling when I call from X-lite to the ATA.
Ethereal also shows RTP is passed from X-lite to Asterisk, and RTP is passed from
Hi Ryan,
Assuming you have looked at the WIKI at www.voip-info.org, there is some
good info there. Not sure of your background, mine is mainly VoIP,
telephony and networking, and definitely not strong on Linux, so if you
would like to email me directly offline ([EMAIL PROTECTED]) with
some of
Title: X-lite to Cisco ATA - no RTP
Interesting,
SUSE firewall allows SIP but not RTP out of the box.
-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Keith Burns
Sent: Friday, February 04, 2005
8:02 AM
To:
asterisk-users@lists.digium.com
I think most people have spent more time complaining about the
AUTO-RESPONDERS than it takes to hit the delete key.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Wilson Pickett
Sent: Sunday, February 06, 2005 4:35 AM
To: Asterisk
Title: Asterisk performance monitoring
Hello,
Has anyone used any 3rd party web based software to get performance information out of Asterisk?
Looking for CPS, call setup times, voicemail database utilization etc
Cheers
Keith.
___
What is your sample size?
I believe the 7960 supports 40ms (2 samples) per packet by default.
Do you have an ethereal trace? Look at the timestamps between RTP packets if
you can't see/modify this setting.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL
and looking at the delta between timestamps on RTP packets from Sipura to PSTN.
-Original Message-
From: Pedro [mailto:[EMAIL PROTECTED]]
Sent: Wednesday, February 16, 2005 1:37 PM
To: Keith Burns
Cc: Asterisk Users Mailing List - Non-Commercial Discussion; Jeffrey Chan
Subject
Payload type=ITU-T G.729, SSRC=3045937487, Seq=7784,
Time=434932771
RTP Payload type=ITU-T G.729, SSRC=3045937487, Seq=7784,
Time=434933091
On Tue, 15 Feb 2005 18:43:39 -0700, Keith Burns
[EMAIL PROTECTED] wrote:
What is your sample size?
I believe the 7960
Are you both using Digium cards?
Do you know if you are using G3 (standard) or SuperG3 (like a modem) fax
machines?
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Justin Richards
Sent: Wednesday, February 16, 2005 4:25 PM
To:
be the transmit/receive fax machines or *
itself.
I hate fax over VoIP. E-fax anyone? :-)
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Justin Richards
Sent: Thursday, February 17, 2005 9:08 AM
To: Keith Burns
Cc: Asterisk Users Mailing
The only problem is that it is bandwidth inefficient and may cause a CPU
hit on your IAD (since you have effectively doubled the pps for a call).
The packets should be 10ms apart. Perhaps the timestamp is not in ms.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
Hi,
I am looking for SER/Asterisk consultants in Denver, please contact me at
[EMAIL PROTECTED]
attachment: winmail.dat___
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