there is some
obvious problem.
Regards,
Andres
http://www.telesip.net
- Original Message -
From: "Terence Parker" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, January 07, 2004 8:38 PM
Subject: Re: [Asterisk-Users] Cisco to Cisco - poor quality
> I
I have managed to find time to have another go at the Cisco phones -
alas, I am still having problems with Cisco to Cisco calls.
Just to re-cap (it's been a few days!) i'm using Cisco 7960's and have
tried setting both phones to different codecs (tried default g729a,
g711alaw, and g711ulaw). Al
I have never used Cisco phones, but I have had problems in the past
relating to * RTP talking to a widget with VAD turned on.
* RTP stack can not run on its own. It relies on receiving RTP packets
for doing its timing.
A simple test is to sniff the line to make sure the phones always send
packets
Thanks for the reply.
The switch is indeed a full duplex 10/100, and we have a relatively small
network with low office traffic so that shouldn't be a major problem in my
case. Also, our cisco phones did work under vocal (except vocal is overall
rather naff) so that shouldn't point to a problem wi
I have set canreinvite=no in the sip.conf for each user (well, there are
only two) using a cisco phone. What does this imply?
As for whether the problem is due to the phones or asterisk however,
indications would suggest both, because:
- Voicemail works fine (and is clear)
- I can initiate a call
Greetings...things got way better for us when we:
0. Opted for voip gateways
1. Eliminated all hubs for switches
2. Eliminated all viruses (I hate PCs)
3. Recabled and seperated our voice from our data network
Three months later we just can't be happier!
Todd
"Terence Parker" <[EMAIL PROTECTED
I seem to recall that you are only sending calls from Asterisk to the
Cisco, not sending calls from the Cisco to Asterisk. Is this correct?
On Sun, 2004-01-04 at 19:10, Jared Smith wrote:
> On Sun, 2004-01-04 at 17:45, Terence Parker wrote:
> > When I make a call between these two phones, the con
see if you can upgrade to firmware 4-3 or 4-4
another point to note, are you using a full duplex 10/100 switch?
if so, you should have 'Port1 Full 100' for full duplex 100Mbit
under the 'Network Statistics'
If you like to email me your config settings, I will check them against our
phones.
telnet
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Terence Parker
> Sent: Sunday, January 04, 2004 8:29 PM
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] Cisco to Cisco - poor quality
>
>
> Thanks for the r
Thanks for the replies.
My cisco firmware is only POS3-04-2-00, though it is SIP. It used to work
fine under vocal though - which was strange. Is this definitely nothing to
do with asterisk? I do note however that my firmware is fairly old... except
cisco aren't exactly generous with firmware upgr
On Sun, 2004-01-04 at 17:45, Terence Parker wrote:
> When I make a call between these two phones, the conversation is of a
> quality so bad that it is barely audible (5% makes sense).
You must be doing something wrong (maybe codec problems), because I've
had absolutely no problems with Cisco to C
what firmware are you using? is it SIP?
to check, push settings then status and firmware
you should have a load ID like this 'POS3-04-4-00'
also check the preferred CODEC
we use g711ulaw as the default
Terence Parker wrote:
> I am just starting to deploy asterisk in our office to use as our prima
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