Hi,
I have got a Digium TDM04B card (4 FXO modules installed) and i'm having
problems getting it working.
ztcfg reports the following:
asterisk:~# ztcfg -vv
Zaptel Configuration
==
Channel map:
Channel 01: FXS Kewlstart (Default) (Slaves: 01)
Channel 02: FXS Kewlstart (De
Hi Mike,
> The solution I found was to change 1 line in one of the zaptel source files
> and
> recompile.
> The file is zconfig.h and I uncommented:
> #define AGGRESSIVE_SUPPRESSOR
Great - i'll give this a try.
What settings do you have in zapata.conf then?
Thanks
--ian
___
Hi,
> substantially lower (pretty much unhearable) on the NTL line compared
> to the BT Line, so you have a fairly good chance (hopefully) of getting
> a decent solution.
I hope so :)
> 00:0a.0 Communication controller: Tiger Jet Network Inc. Tiger3XX
> Modem/ISDN interface
Yeah, looks l
Hi,
> Try echotraining=800 in zapata.conf
Thanks for the suggestion but I tried that and still get the echo :-(
Any other ideas?
--ian
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-
Hi,
> If you stick with fxsks and add busydetect=no and callprogress=no
> you'll find those random disconnects go away. They did for me.
I just tried that and I still got the disconnection so I changed it back
to fxsls but left those 2 lines in zapata.conf for good measure!
> Yes. The echo dr
Hi Phil,
Thanks for the reply - comments below.
> Change fxsls to fxsks for the UK
I had fxsks until today but when I was playing with the gains to try and
fix this echo issue, I found that if I dial out the Zap to one of my
pstn->sip provider's numbers, it answered the cut off in a few seconds.
Hi All,
I've got an MD3200 modem which is working as a Zaptel FXO interface for
Asterisk (X100P clone I believe). It seems to work, but on incoming or
outgoing calls I can hear the other party ok but when I speak, I hear
my voice echo back at me (quite quietly but it's distracting!) on
everything
Hi,
> For incoming calls, Asterisk matches peer's on IP, meaning that the
> first peer it finds will match. This is the *last* one you have in
> sip.conf. The context given in that peer must have *all* extensions you
> need for incoming calls, which is the extension at the end of the
> registe
Hi,
I have a few accounts with sipgate.co.uk to get some different DiD
numbers. However, when an incoming call comes in, it seems to pick the
wrong peer from sip.conf which sends the call into the wrong context and
it fails because there is no extension in that context to match the
register.
Usin
Hi,
I have 2 g729 licences - what I want to do is use g729 by default but if
I get more than 2 calls at a time, use gsm for the others.
So, I put this on all my sip providers:
disallow=all
allow=g729
allow=gsm
However, this just seems to use gsm for everything. If I comment out the
gsm lin
Hi,
I have an FXO card connected to my phone line which works in Asterisk as
Zap/1.
Is there any way of detecting whether something else is on the line
before picking up on this channel?
For example, I dont want to pick up and dial out on the line if someone
is on it using another phone (which i
Hi,
I have my Sipura SPA-3000 setup with Asterisk as follows:
[spa3k_line1]
type=friend
context=home
secret=PASSWORD
host=dynamic
dtmfmode=rfc2833
dissallow=all
allow=ulaw
When an incoming call comes in, I have a Zap interface in Asterisk which
just does a Wait,15 then answers with
Hi,
I have set emailsubject in voicemail.conf as follows:
emailsubject=New Voicemail from ${VM_CALLERID} in Mailbox ${VM_MAILBOX}
This works fine, but the pager e-mails come through with "New VM". I
would like the pager e-mail to be the same subject as above as I get
this as an SMS to my mobile p
Hi,
> What handsets are you using? Could be the firmware!
It's sip providers i'm having the problem with - not phones.
--ian
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or up
Hi,
I have several SIP registrations on my Asterisk box. Sometimes, I try to
call in the inbound number from 1 and find it doesn't work. When I do
sip show registry, it's showing Unregistered (and sometimes there are
several which are showing Unregistered). If I type reload, it registers
and works
Hi,
I am having a few dial plan problems which I wondered if anyone would be
able to help with.
Firstly, I wanted to send 0800 calls through 1 sip provider and other
08xx calls through another. I have this:
exten => _0800.,1,Dial(SIP/[EMAIL PROTECTED],30)
exten => _0800.,2,Congestion
exte
Hi Chris,
Thanks for the reply.
> Yup, I have Asterisk registering with Gossiptel.
I have now got it to register ok but it's not working properly - I can
call the 160 echo test number but it's leaving the channels open after
the call has ended (in sip show channels) and I can not make any other
Hi,
If I do a "sip show channels" - I seem to be getting channels left open
after calls have ended - any ideas why?
I thought at first it was my Sipura SPA-3000 and that Asterisk was not
detecting that i've hung up.
However, after more testing, it seems to be just on Gossiptel calls - I
tried a
Hi,
I have Asterisk setup and registered with Gossiptel but i'm only getting
1 way audio.
If I call 160 (echo test) or 123 (talking clock), it makes the call but
I just get silence. If I call my Gossiptel number from a pstn line, I
get gossiptel -> pstn audio but not pstn -> gossiptel audio.
I'v
Hi,
Is it possible to create an extension (say *1) that will give access to
the voicemail for the current extension without entering the mailbox or
password?
(or if this is not possible, at least not have to enter the mailbox -
only the password?)
Thanks!
--ian
___
Hi Rick,
> "If" your configuration and firewall actually require you to open a
> group of ports to *, then take a look at limiting the rtp ports that
> are actually used.
How many do I need (or how do I find out?) and why does Asterisk specify
so many by default?
Thanks
--ian
__
Hi Shane,
> http://www.voip-info.org/wiki-DNS+SRV
> http://slacker.com/~nugget/asterisk7.php
The SRV page was useful - i've done that in my domain now.
But, the other page is talking more about dialing sip addresses through
Asterisk rather than incoming sip addresses.
However, after adding the
Hi,
> >I assume ports 5060 and 1-2 need to be opened
> >in the firewall too.
> I don't know much about SIP and firewalls, but opening ten thousand
> ports doesn't sound good, you've just knocked 1/6 of your firewall down
That's what I thought but I was told it was the only way to get i
Hi,
Has anyone got Gossiptel working with Asterisk? - I am having real
problems getting it to register - i'm just getting timeout errors.
Thanks
--ian
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk
Hi,
Is it possible to have an incoming SIP address like
[EMAIL PROTECTED], where sip.mydomain.com points to a box
running Asterisk?
If so, please could someone give an example asterisk config snippet for
this?
If it is possible, I assume ports 5060 and 1-2 need to be opened
in the firewa
25 matches
Mail list logo
