Displaying 20 results from an estimated 10000 matches similar to: "Codecs for use with Cisco 7960 and ATA-186"
2003 Jul 03
2
ATA-186 de-register
Is it just me or do others have a problem with the ATA-186
de-registering? Every couple of hours, if I don't make use of the ATA
connected line, I find that I have to unplug and let the ATA reboot.
After that it is good to go for awhile, but eventually I have to repeat
the process. My ATA sits behind a NATd firewall, any ideas what might
cause the de-registration?
Kim C. Callis
2003 Jul 10
2
Transfers on the Cisco 7960
I noticed that there is a soft button for transfer when you initiate a
call. I pressed it, and it actually put the call on hold, although I was
able to call another extension. Is that soft button functional? And if
so, how do you make use of it? And if not, how does one transfer a call?
Kim C. Callis
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2003 Sep 11
1
g729 codex experimentation
Yesterday, I started to experiment with Cisco to Cisco SIP calls using
the g729 codec. According to the documentation, both the ATA-186 and
7960 are able to make use of the g729.
>From an earlier e-mail, I made a change to the configuration of the ATA,
changing the values:
LBRCodec:3
RxCodec: 3
TxCodec: 3
The first thing I noticed was that when I did a sip show channels, the
format had
2003 Jul 17
3
Any dialing tricks...
Alright, I am basically cheap, and I have a cellular plan which allows
for free incoming calls (Nextel). I was wondering if there was any way
to do sort of a dialback trick in the extensions.conf. I call into the
system from my cell phone (maybe via DISA), I dial an internal
extension, and dial a phone number. Then * sends to my cellphone the
number dialed thus giving me a in call on the cell. Or
2003 Jun 27
2
Working: TFTPd for NAT'd Cisco 7960 and ATA-186
For anyone who is interested, I have a working tftpd (modified wvtftpd)
capable of serving configuration, dialplans, and ringtones to Cisco
7960/7940 and ATA-186 devices that are located behind NAT firewalls. As
TFTP is not a very firewall/NAT friendly protocol, I had to break some
rules to get it to work with these cisco devices. It might cause
problems for other TFTP clients, but it works with
2003 Jul 04
1
LD accontability
As I was working on my extensions.conf file, I started to segment
calling privileges. For the everyday workers, I don't free reign to LD
access unless it is business related. So I was wondering if there was a
way to implement some type of accounting code to be entered before
accessing LD, which of course would be noted in the CDR (however it is
implemented, either comma delimited or MySQL).
2003 Aug 12
3
Fair comparison
I was trying to do a little searching to see if there has even been a
comparison between Asterisk and VOCAL or any of the other OSS packages?
"Practical Voice Over IP using VOCAL" published by O'Reilly and
Associates, attempts to make a strong case about how scalable VOCAL. Of
course, considering that the book is written by the makers of VOCAL, it
tends to have a one sided slant.
2003 Jul 28
0
Re: Asterisk-Users digest, Vol 1 #882 - 11 msgs
Hi!
Sure, just look for: Wonder Shaper. It's a HTB based shaper
configuration wich have some very good features, I use a variation of
that here at my College.
http://lartc.org/wondershaper/
It is the page (a simple google search).
Also make sure to uncomment the line tos=lowdelay in every config file
of asterisk that have it.
Hope it is usefull, sincerely,
Ildefonso Camargo
2003 Jul 18
5
cdr_mysql
Considering that I had a failure with compiling the latest version of
asterisk because of cdr_mysql, I am going to assume that I need to have
a copy of the mysql headers on the system in order to compile
cdr_mysql.so. Does that sound correct?
Kim C. Callis
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2003 Jul 06
9
Accurate Billing
<P>hi everyone,</P>
<P>I know this issue has been raised many times before, i think still the problem remains. When a call is made through a Zap channel, whether it is actually made or not (irrespective of whether, engaged, busy, or actually answered), asterisk logs it in CDRs as a call made. This makes it impossible to do an accurate billing. Has anybody found a way to overcome
2004 Feb 03
1
Problems with chan_sip: random calls have no sound withouth any errors
Hi All,
I have been busy with this problem for a while now, but I can't find any
solution. First I thought this was a problem with the phones, but all my
phones have this problem. (2 SNOM 200, 2 GS 102, 2 Cisco 7960). I tried
all firmware versions I could find for the phones.
First, my situation:
- No NAT, No Firewall, same subnet
- Codec configuration:
In general:
disallow=all
2004 Jan 12
4
RFC3389 messages with ATA 186
I'm getting some warnings:
NOTICE[xxx]: File rtp.c, Line 264 (process_rfc3389): RFC3389 support
incomplete. Turn off on client if possible
Asterisk Version: CVS-01/06/04-13:50:26
Cisco ATA 186 version: v3.0.0 atasip (Build 031210A)
Is this something I should be concerned about? Anyone know how to "turn
off" the RFC3389 support on the ata 186?
Thanks!
2003 Jul 16
0
Analog features over the ATA-186
I have been using the ATA-186 with good success (with exception of the
fact that you have to recycle it from time to time). The one thing I
haven't been able to do is to figure out how to make use of parking,
transfers, etc. My cordless doesn't seem to pass DTMF, so I haven't had
any success with parking. As for transfers, I can do a flash and get
dial tone, but then what are my
2003 Sep 10
9
G729
I have come to realize that I don't have to have a g729a license in
order to make use of an ATA-186 or 7460 connecting to another 7460. I
just need to allow the codec in sip.conf.
Now what ramification does that have when I dial out over one of my
analog line (connected to * by a channelbank and a T100P) using my 7460
or ATA-186. The only benefit I am looking for is reduced bandwidth
2003 Nov 19
2
g723 to g723 SIP call - warning message
Hi,
I am calling from a grandstream phone with g723 codec through * to iconnect.
Incoming context as well as outgoing context set to g723.1 codec in *.
Call get connected and I can talk. However I get the following warning,
which scrolls on my screen until I hang-up.
[root@asterisk sath]# cat g723.1
- Executing SetCallerID("SIP/-08122ae0", "1001") in new stack
--
2009 Jul 09
1
Connecting two Asterisk together via SIP + DISA
Hi all,
I need to test the following scenario:
+-----------+ +-----------+
| asterisk 1| | asterisk 2|
+-----------+ +-----------+
| |
| |
_______|__________________|___________
| |
| |
| |
+-------+ +-------+
| ATA 1 |
2005 Mar 27
8
Asterisk on a dialup connection?
How will this fare?
I am planning on putting an asterisk box for my brother in the
Philippines but they only have dialup internet. I want them to be able
to use a telephone set on a phonejack or linejack card and call me and
vice versa via VOIP.
My setup in the US is working already with a broadband cable
connection.
I am thinking that dialup may not work because of the bandwidth required
2004 Jan 18
3
ATA-186 pass-through Flash
Hello all!
I have an FXO port on a cisco router that is directly connected to our PBX.
Our ATA-186 (firmware version 3) registers with asterisk, which connects to our cisco router's fxo port to give me a dialtone on our PBX from the ATA.
How do I pass the flash button to the PBX? It seems the ATA-186 wants to control the flash by putting my first call on hold and prompts me to dial another
2004 Apr 06
1
Asterisk CLI Issues - CVS-03/30/04-14:34:01
Hello all,
I just updated my development Asterisk box today and am noticing
weird behaviour in the CLI. Whenever I hit "tab" the CLI shows me the list
of commands, but the output is garbled. See below. I'm running the
v1.0_stable code, checked out about 2 hours ago.
asterisk*CLI> show version
Asterisk CVS-03/30/04-14:34:01 built by root@asterisk.n2net.net on a i686
running
2004 Jul 18
4
Cisco 7960 SIP V6 and IBM A30P Fedora Asterisk
Hi All
Total noob on the list so all help appreciated....
I've successfully installed Asterisk on an IBM A30P Thinkpad using fedora Core 2 (I'm looking at having a mobile PBX for conferences and shows).
I've plugged in two Cisco 7960 phones....
The phones register with the Asterisk correctly and I can run the demo's and even the AIX demo through to digium works correctly.......