AAH 1.3 with Cisco 7960 phones/ SIP 7.5 software mostly works great,
but there is a problem with one of the phones I use most: It disconnects
calls if I dial on speakerphone and then pick up the handset after the
other side answers.
Thanks in advance for any clues on this. And apologies if thi
On 7/15/05 1:00 PM, "trixter http://www.0xdecafbad.com";
<[EMAIL PROTECTED]> wrote:
> Thought those that use cisco in conjunction with asterisk may want to
> read this. I dont use cisco so I havent read it to see if its actually
> anything new.
>
>
> Vulnerabilities in Cisco's VOIP system
> htt
On 7/14/05 6:33 AM, "Rich Adamson" <[EMAIL PROTECTED]> wrote:
>> AAH 1.3, Digium 4-port FXO card connected to PSTN
>>
>> I am having problems with outbound calls, where the call goes to either an
>> error message from the PSTN, or a fax number, or a wrong number. It works
>> correctly maybe 1
AAH 1.3, Digium 4-port FXO card connected to PSTN
I am having problems with outbound calls, where the call goes to either an
error message from the PSTN, or a fax number, or a wrong number. It works
correctly maybe 1 time in 10.
Also, outbound calls *sometimes* work if they are numbers previous
AAH 1.3, Digium 4-port FXO card connected to PSTN
I am having problems with outbound calls, where the call goes to either an
error message from the PSTN, or a fax number, or a wrong number. It works
correctly maybe 1 time in 10.
Also, outbound calls *sometimes* work if they are numbers previous
On Mon, 7 Mar 2005, Howard Lowndes wrote:
On Mon, 2005-03-07 at 09:02, David Newman wrote:
On Sun, 6 Mar 2005, Marty Mastera wrote:
The full text of the bug you reference above indicates that "pstnVMgain"
was (or is) part of an ongoing feature request/bug report and has not
been imple
On Mon, 7 Mar 2005, Peter Illmayer wrote:
You will need to load the version 3, then 5 and then 7 SIP firmware. I tried
to load the version 7 straight away and of course it wouldnt work.
FWIW, I have also had success doing versions 3, 6, and then 7 in moving
from Skinny to SIP. But it's still thre
On Sun, 6 Mar 2005, Marty Mastera wrote:
The full text of the bug you reference above indicates that "pstnVMgain"
was (or is) part of an ongoing feature request/bug report and has not
been implemented for use at this time (and may never be).
Right. So -- what can I do to boost volume of PSTN -> * v
On Asterisk 1.0 with a 4-port Digium FXO card, voicemails from the PSTN
have volume so low they often can't be heard. Worse, callers sometimes get
cut off in the middle of leaving a message. It is extremely frustrating to
hear "...and my number is...END OF MESSAGE"
A search of the archives show
On Mon, 31 Jan 2005, Mitchel Constantin wrote:
This doesn't solve the clear text issue, but how about an access list
based on the mac addresses? That'll secure tftpd a little more.
MAC addresses go away as soon as you have one or more router hops between
phone and server. ACLs based on IP address
On Mon, 31 Jan 2005, Gregory Junker wrote:
There should not be any, except for the occasional rekeying.
That's right.
If you can, try capturing traffic on either side of the VPN tunnel
endpoints to see what's creating all those packets.
dn
___
Asterisk-
On Mon, 31 Jan 2005, Robert Webb wrote:
Try looking in your extensions.conf file. If you are using ports 0 and 2
then you should see somewhere in there something like zap/1 and zap/4
and those should tied to the dial commands.
Hopefully these have been configure in the globals section of the
extens
Greetings. We are running * on RH9 using a Digium TDM400P four-port FXO
card. We use only two ports on the card (ports 0 and 2 in this case). A
consultant set this up for us, and it mostly works OK.
However, outbound calls use our secondary number rather than our primary
number first. This unde
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