Displaying 8 results from an estimated 8 matches for "186s".
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186
2004 Jan 23
3
UK BT Interface with asterisk?
Have anyone tried to interface BT's Broadband Voice with asterisk?
Kannaiyan
2003 Jul 30
4
SCO/Linux concerns
...2.dmv.com>
> To: asterisk-users@lists.digium.com
> Subject: Re: [Asterisk-Users] sip -> h323 -> ptsn
> Reply-To: asterisk-users@lists.digium.com
>
>
> I have the same setup, and in the sip.conf file I set the dtmfmode=inband
> for each endpoint defined and my Cisco ATA-186s and 7960 phones all work.
>
>
> On Wed, 30 Jul 2003, Brian West wrote:
>
> > I have this setup:
> >
> > Sip Phones -> Asterisk -> h323 gateway -> ptsn
> >
> > Sip phones are setup for out of band dtmf
> >
> > but the h323 gateway is inb...
2003 Apr 23
6
OT: Multiple SIP phones behind NAT gateway?
Hi,
I know this is slightly off topic but I figured the knowlege here is probably the best on the subject..
I want to setup remote offices with 4 to 6 SIP phones (SNOM 200) using ADSL and the internet to connect to the Asterisk box..
These phone will be behind an ADSL router using NAT...
I don't want to setup another Asterisk system in each office so IAX is not an option..
I could use
2004 Jun 02
2
cisco ata-186 behind NAT
i have been trying to get a newly liberated (from vonage) cisco ata-186
(sip ios v3.1) working properly with asterisk. my client is behind a
linksys wrt-54g, which up to this point hasn't proven to be a problem
(i have several sipura spa-2000's and polycom phones working just fine
behind them). (i'm running cvs-head from yesterday).
after looking at the various suggestions,
2003 Sep 22
1
Switch between calls without initiating a threeway converstaion
I was just wondering if there was a way that you could
have two calls on one line and switch between the two
without initiating a threeway conversation?
I would imagine that Flash is the way to do this, but
when I Flash twice, a 3-way call is initiated. If I
turn threeway off, then I can't transfer.
Also, is it possible to hang up one of the calls, and
then continue talking to the second
2003 Jul 31
1
RTP codec 13 received - Cisco incompatibilit y?
I have a similar setup to you and get the same message regularly. I don't
think it's the cause of your problem. I did some research on it a while ago:
IIRC the cisco uses codec 13 for "silence suppression" whereas asterisk
(correctly) uses codec 19. The router can be configured to use 19 also, but
I didn't bother. I'm sure somebody will correct me if I'm wrong about
2005 Mar 23
4
Vonage Linksys Router - Life after Vonage
I setup a vonage account last year, and cancelled it last night when I
put my asterisk box together and signed up for a Broadvoice account to
use with it.
Now I would like to use my Linksys router as an MTA, but realize it is
still programmed with all of vonage's proprietary information and I do
not know how to clear it. I understand that just pushing the reset
button will not do it.
2005 Oct 10
0
Asterisk behaving wierd!!
hello,
I have been using asterisk now for about 2 years now on a RH8.0 it is our
main call gateway.
I have on the box 3 T1 TDM cards connected to 2 Rhino channel
banks (FXS) and 1 CAC Access bank I (FXO) with so many softphones and ATA
186s.
It has been working good till today some few hours ago. i just
discovered that there were no dialtone on the phones.
Asterisk did not spit out any error, it tried reloading but there was no
response, i did a "stop now" and it gave no such command "use help", i did
a help and i...