Hi everybody,
I am trying to make up call flow diagrams for for a setup which
include ser as a sip proxy/registrar and asteriks as a voicemail
server.
Is my sequence correct?:
UA 1 send an invite to SER. SER forwards this invite to UA2. UA2
sends back a sends back a 100 trying and 180 ringing
Hi all,
Hope somebody can help-I really am stumped as to why this won't work.
I realise that this isnt an Asterisk problem (Please dont bash me on
the list) and I have emailed the SER list but I havent received a
reply and maybe someone on this list can help...Once this problem is
solved I am
Hi all,
Could someone let me know the most common way that an Internet ISP
would allow customers access to the PSTN?? Do they buy multiple fxo
cards such as the TDM400P and rent multiple lines from a larger
provider??
Would the best way be to connect to a third party voice/pstn
gateway?? Is
Hi,
I currently have Asterisk running behind a linux router running nat.
Clients register with the public address and when the sip requests
reach the router, port forwarding is used to divert the traffic to *
i.e. all sip and rtp go to the asterisk box.
I now want to set up ser (so that i can
From my (fairly limited) understanding, I think the fundamental
difference is that Asterisk is a pbx (offering all the features
associated with a pbx, voicemail, call transfer, call detail
recording etc) whereas SER is just a sip proxy (albeit a good one).
Therefore Asterisk deals in terms of
I wonder does anyone have any thoughts or can give me some direction
on the following:
I have an asterisk testbed environment set up. My task is to make a
personal number service available whereby users would be given one
number (perhaps a voip number) and this number would enable them to
be
read 3K/annum
plus a setup fee (yes - that puts the cost of the card in
perspective).
To conect your * box to PSTN with BRI/PRI interface you'll need one
of
digium's cards or an equivalent CAPI card.
Br /Kev/
- Original Message -
From: Ashling O'Driscoll [EMAIL PROTECTED
Hi,
First of all apologies because this isn't strictly a purely asterisk
question.
I am quite new to asterisk and actually to voip/telephony as a whole.
I currently have sip calls working through asterisk. The asterisk
server is behind a linksys router. I would now like to connect calls
to the
Hi,
I am also interested in what softphones other asterisk people are
using. I am using xlite but that doesnt seem to have any voicemail
capabilities (correct me if im wrong). You have to purchase xpro for
that. Does anyone have any suggestions?.
Apologies to the person who first sent this
will do, and you will have to specify your
local
net in sip.conf as well.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Ashling
O'Driscoll
Sent: 15 November 2004 11:14
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] maximum retries error
Hi all,
I have two
Hi all,
I have two xlite clients which are attempting to make a call through
asterisk. The call seems to connect and the clients are both marked
as connected on either side how ever no audio is transmitted. One
client is behind nat (the asterisk server is also behind nat). I am
getting the
stuff.
Use md5secret rather than secret in sip.conf. You'll have to
MD5 hash your password... there's documentation on this in the
Wiki.
-Chad
On Nov 10, 2004, at 9:25 AM, Ashling O'Driscoll wrote:
Hi,
Hope somebody can help. I have two xlite clients that register
with
asterisk
Hi,
Hope somebody has an idea as to what the following means:
I am making a call from one xlite client (2000) to another xlite
client (2001) via asterisk. The call seems to connect fine and each
client comes up as 'connected'. They both have the same codecs
enabled and have turned the silence
Content-Length: 0
Original Message
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] xlite and asterisk
Date: Wed, 10 Nov 2004 12:39:22 +
404 not found can mean many things, are you using a supporting codec?
On Wednesday 10 November 2004 05:25 pm, Ashling
Hello,
I am having problems getting two xlite clients to communicate through
asterisk. I am getting an error message:
chan_sip.c:2753 process_sdp: No compatible codecs.
I have enabled all possible codecs in xlite (Menu - Advanced system
settings -Codec settings) and have added the appropriate
Hi,
Just wondering if anyone can enlighten me as to what the following
error signifies:
WARNING[2941]: chan_sip.c:665 retrans_pkt: Maximum retries exceeded
on call XX for seqno XXX
This happens when a client is registering. The client still registers
successfully but
Hi,
Hope somebody can help. I have two xlite clients that register with
asterisk. They are called 2000 and 2001.
1)When 2000 rings 2001 a '404 not found' message is returned even
though he is registered with asterisk.
2)When 2001 rings 2000, a 'call not approved' error is returned. I
found a
.
Original Message
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] xlite and asterisk
Date: Wed, 10 Nov 2004 12:39:22 +
404 not found can mean many things, are you using a supporting codec?
On Wednesday 10 November 2004 05:25 pm, Ashling O'Driscoll wrote:
Hi
Hi all,
Hope somebody can help me to figure out the following scenario or
send me on their config files if they have a similiar network
configuration.
I first set up asterisk and two clients on the same network and it
worked fine. I now have asterisk set up which is acting as a sip
registrar.
-Users] asterisk and nat
Date: Mon, 08 Nov 2004 16:40:32 +0100
Ashling O'Driscoll [EMAIL PROTECTED] writes:
I first set up asterisk and two clients on the same network and it
worked fine. I now have asterisk set up which is acting as a sip
registrar. It is behind nat. I also have two clients which
from 172.16.3.13
Date: Thu, 4 Nov 2004 18:55:30 -
MIME-Version: 1.0
Content-type: text/plain; charset=iso-8859-1
Content-Transfer-Encoding: quoted-printable
Hi all,
I hope someone can shed some light on the following: - I came across
a thread with a similiar problem but it didnt fix the
Leg/Transaction Does Not Exist
back
Date: Fri, 05 Nov 2004 03:16:13 +0800
On 11/4/2004, Ashling O'Driscoll [EMAIL PROTECTED] wrote:
[general]
port =3D 5060 ; Port to bind to (SIP is 5060)
bindaddr =3D 0=2E0=2E0=2E0 ; Address to bind to (all addresses on
machine)=
diallow=3Dall=20
allow=3Dulaw
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