Displaying 20 results from an estimated 2000 matches similar to: "Re: Echo and other reasons to migrate to BRI"
2006 Feb 28
0
Re: Echo and other reasons to migrate to BRI from POTS? Was (Echo on PRI/BRI?)
Paul,
Ah, I see. Our echo is largly under control now. It took me a while to
figure out the gains and get them tuned, and now the echo only leaves very
small artifacts. Nonetheless, this still provokes the odd complaint here and
there. We use VOIP for outgoing calls when our POTS lines are congested, and
we find zero echo during those calls. Therefore, I assume that our handsets
(Cisco
2006 Feb 27
2
Echo on PRI/BRI?
Howdy:
Does echo only occur on analogue PSTN lines, or can it also occur on PRI and
BRI lines? If so, for the same reasons? This is a part of our consideration
to transition to BRI.
Sincerely,
Brent A. Torrenga
brent.torrenga@torrenga.com
Torrenga Engineering, Inc.
907 Ridge Road
Munster, Indiana 46321-1771
219.836.8918x325 Voice
219.836.1138 Facsimile
www.torrenga.com
2006 Feb 14
4
BRI Newbie - What Hardware, PCI, in the US?
We are looking to lose the TDM400P in favor of an ISDN-BRI solution. This
should get rid of static on the line (at least any static generated by our
half of the circuit), right?
I am a total virgin to ISDN. I understand that we need two BRI circuits to
provide four voice channels, and that the hardware to speak to the BRI
circuits can be passive or active, with the active type being much more
2006 May 26
0
Sip Notify cisco-check-cfg - Does it still workwith 8.2?
It does on my test phone. Is your tftp server available?
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Brent
Torrenga
Sent: Monday, April 17, 2006 11:39 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Sip Notify cisco-check-cfg - Does it still
workwith 8.2?
Has anyone else noticed that
2006 Jan 17
0
RE: Building from scratch would like the benefit of (TOO LONG...)
I use Cisco 7940's and 7960's over here, loaded with SIP, and they do very
nicely. Also, the SCCP channel for * is under heavy development, and may
offer a future option to convert in that direction, too (SCCP, or skinny, is
their native tongue, not SIP). We got our phones from John Putnam at Global
Technology Solutions - competitive price, very prompt service, and delivered
them with the
2006 Jan 30
1
Need to recompile * after changing zap echo method?
Dearest List,
I guess I missed this point: Is it true that if you change the echo canceler
in zconfig.h, and then recompile/install your zap modules, that for this to
be taken into effect by * you must then recompile/install *?
I would have figured that the zap echo cancellation method was independent
of *, and I don't recall seeing any docs mentioning either way.
Sincerely,
Brent A.
2006 May 31
1
Can you dial with different CID's?
Is it possible to dial more than one extension with a different CID to each
extension? I'm thinking macros might be needed, but I don't have a good
handle on macros. Is it possible? Any hints?
BTW - this would be used for showing an internal extension to one phone and
a PSTN accessible number to another phone.
Sincerely,
Brent A. Torrenga
Torrenga Engineering, Inc.
907 Ridge Road
2006 Dec 18
1
Cisco 7940 - NAT Option
I am thinking of turning on the NAT option in our Cisco phones (and the
corresponding sip.conf modification) to allow the phones to be taken outside
the LAN.
Can anyone think of any reason not to just always turn on the NAT enabled
option? I can't think of a reason not to always operate these phones with
this enabled, since it would likely allow them to be taken outside our LAN
and used.
2007 Jul 05
2
REGEX expression for NXXNXXXXXX?
Hola,
What would a valid regexp in Asterisk be to identify a NANP number, i.e.,
NXXNXXXXXX?
Sincerely,
Brent A. Torrenga
Torrenga Engineering, Inc.
907 Ridge Road
Munster, Indiana 46321-1771
tel:+1 219 836 8918 x325
fax:+1 219 836 1138
email:brent.torrenga at torrenga.com
web:www.torrenga.com
2006 Apr 10
0
RE: still no solution for me, if one
>Brent,
>
>you mean, I could just remove the remark signs and number it 103, 104,
>105, .... since it does not matter why it failed (busy, congestions)
>(maybe for statistic purpose to add a log entry for the move to the next
>provider).
>
>bye
>
>Ronald
Yup. Take a look at the macro solution, too. I don't fully understand macros
(I'm no programmer), and
2006 May 15
3
How to tell if RTP stream is has been reinvited?
Howdy,
How can you tell if RTP traffic has been reinvited/is bypassing an * server?
Sincerely,
Brent A. Torrenga
brent.torrenga@torrenga.com
Torrenga Engineering, Inc.
907 Ridge Road
Munster, Indiana 46321-1771
+1 219 836 8918 x325 Voice
+1 219 836 1138 Facsimile
www.torrenga.com
2006 Feb 16
2
Random Hangups/Disconnects
Well, I thought and hoped my issue of random hangups on our TDM400P were
related to busydetect=yes in zapata.conf. The behavior of a call being
hungup has not changed, however, since setting the busydetect option to
'no'. Again, the only affected user is my loud talker...
What are some causes/solutions to seemingly random call disconnects on Zap
channels that people have seen? I have
2006 Jun 06
1
SIP One-way audio: == Forcing Marker bit, because SSRC has changed - trxtel.com
Dear list (and more specifically Bret),
I am getting one-way (inbound only) audio when trying to place a SIP call
via voip.trxtel.com (i.e. 18005558355@voip.trxtel.com). The Cli spits out
"== Forcing Marker bit, because SSRC has changed" 5 times after atempting a
native bridge. I realize this is most certainly a NAT issue, the * server is
behind one. Sip.conf has externip=, and
2002 Nov 27
1
Upgrade failed? 2.2.5-pre1 to 2.2.7
Hello,
I downloaded the source, followed the instaructions to the T (make,
install, etc...). Everything seemed to go fine, I think the compilation and
installation were flawless (after all, I have compiled 2.2.5 sucessfully,
and havent done anything to this box as far as removing software libraries),
there were no reported errors. The install portion even had the messages
saying all went well,
2006 Mar 16
1
MeetMe - Causes * to crash :/
Anyone ever seen MeetMe cause * to crash? Specifically, it happens
consistantly if someone begins to enter a conference and then decides to
hangup while Allison is introducing them - like playing back
"conf-onlyperson". This has been seen with the MeetMe participant connecting
via IAX and SIP (not saying it doesn't happen with Zap, just that I haven't
seen it).
The box is *
2006 Apr 10
1
RE: still no solution for me, if one provider
>Our user places a call, the gateway responds with no sound at all, or
>hangs up, or gives busy tone.
>
>How can we get to the next provider?
>
>I have now:
>exten => _9011Z.,103,Dial(SIP/011${EXTEN:4}@provider-a)
>;exten => _9011Z.,103,Dial(SIP/011${EXTEN:4}@provider-b)
>;exten => _9011Z.,103,Dial(SIP/011${EXTEN:4}@provider-c)
>exten =>
2006 Jan 06
0
How to properly use GROUP
Can someone explain how to use groups? I can't seem to wrap myself around
this, though I know it is something simple.
I have 3 zap lines, and when placing an outgoing call, would like to 1) use
a zap line if and only if 1 or fewer zap lines are being used at the time,
and 2) if more than 1 zap lines are in use then to go ahead and use VOIP to
place the call.
Also, is there a difference
2006 Jan 17
0
RE: Building from scratch would like the benefit of (TOO LONG...)
I use Cisco 7940's and 7960's over here, loaded with SIP, and they do very
nicely. Also, the SCCP channel for * is under heavy development, and may
offer a future option to convert in that direction, too (SCCP, or skinny, is
their native tongue, not SIP). We got our phones from John Putnam at Global
Technology Solutions - competitive price, very prompt service, and delivered
them with the
2006 Apr 17
0
Sip Notify cisco-check-cfg - Does it still work with 8.2?
Has anyone else noticed that notifying a 79[46]0 with cisco-check-cfg
doesn't elicit any response from the phone using fw 8.2?
Sincerely,
Brent A. Torrenga
brent.torrenga@torrenga.com
Torrenga Engineering, Inc.
907 Ridge Road
Munster, Indiana 46321-1771
+1 219 836 8918 x325 Voice
+1 219 836 1138 Facsimile
www.torrenga.com
2006 Apr 18
0
Re: Cisco 7940/7960 SIP 8.2 Freely
It doesn't seem as much broken as just annoying. I am holding off on
upgrading until this resolves, but it doesn't seem to affect performance,
anyways. BTW, some folks say that the server address only gets appended to
the CID when a redirect or something comes about. Our experience here shows
that the IP always gets appended.
>Alexander Burke wrote:
>> Just in case anyone here