Ok got it up and running. In the case for Qwest with NFAS they reserve what
they call "Interface ID 1" for the circuit with the backup d channel. In
our case we only have two circuits with a single d channel. The real key
was realizing the logical span number in the spanmap translated into
"inte
t; 1.2 you can't bring up PRI outside asterisk, since the PRI (I'm
> assuming layer 2+) part loads with Asterisk.
>
> On Sat, Jun 12, 2010 at 10:51 AM, Voip Asterisk
> wrote:
> > Ya i'm not even to the asterisk part yet. I'm still trying to get dahdi
> to
Ya 99% sure that isn't it since they were just pulled working off an AS5300
On Sun, Jun 13, 2010 at 4:27 AM, Doug Lytle wrote:
> Voip Asterisk wrote:
> >
> > Status: Provisioned, Down, Active
> >
> > specifically the "Down" part.
> >
>
>
>
allforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
group=1
channel => 1-23
group=2
channel => 25-48
If anyone could let me know what I should be doing next. I'm sure my issue
is:
Status: Provisioned, Down, Active
specifically the "Down" part.
Thanks
On Sat, Jun 12
settings
on dahdi/asterisk?
The line card used in the cisco was the standard 4 port T1 PRI card.
Thanks
On Sat, Jun 12, 2010 at 7:51 AM, Voip Asterisk wrote:
> Ya i'm not even to the asterisk part yet. I'm still trying to get dahdi to
> bring up the PRIs without alarms.
>
>
>
Ya i'm not even to the asterisk part yet. I'm still trying to get dahdi to
bring up the PRIs without alarms.
On Sat, Jun 12, 2010 at 4:58 AM, Doug Lytle wrote:
> Voip Asterisk wrote:
> > Hi,
> >
> > I'm trying to bring up two PRIs from qwest with asterisk
Hi,
I'm trying to bring up two PRIs from qwest with asterisk and dahdi. I'm
using an OpenVox D410E and the drivers are loaded. My system.conf looks
like this:
# Span 1: TE4/0/1 "T4XXP (PCI) Card 0 Span 1" B8ZS/ESF RED
span=1,2,0,esf,b8zs
bchan=1-24
# Span 2: TE4/0/2 "T4XXP (PCI) Card 0 Span 2"
I'm trying to get asterisk to proxy h263 for a video call, but not having
any luck. I have posted a full call trace here:
http://pastebin.com/d330aecb5
While watching a full packet dump on the asterisk node, I can see the h263
coming in from the clients, but it never leaves (asterisk never origi
Is there anyway to setup a queue with only one agent (device) which is
always logged in. So when a call hits that queue the device will ring (if
not already on a call) or will be put in the queue if the call is already in
place?
Thanks
Miles
___
--Band
Awesome, any chance you can share your resource specs?
Thanks
Miles
Asterisk works great with openvz. Ive run 4 VE's with combined average
around 32 simultaneous calls at any time and you wouldn't know the
difference.
___
--Bandwidth and Colocation
Anyone here running asterisk on openvz, if so what are your experiences?
Right now we are trying to tune out the resources for the difference VEs,
but not with a whole lot of luck. Just wondering if someone watching could
shed some like on what has worked for them, and how many exts/simultaneous
Does anyone have a good suggestion for a automated solution to record calls
on certain interfaces and easily archiving them in a way which is easily
matched against CDRs? Also can someone suggest the appropriate protocol to
archive the recording when the conversations are transpiring in ulaw.
Bas
Is is possible to detect open SIP channels in the dial plan and use that
detection to perform logic?
For instance, say you have a multi line device such as a poly 301, depending
on the path of the incoming call you want to be able to route the call to
the phone if 1 line is already in use or send
What about open sip stack:
http://www.opensipstack.org/
?
Use a far end nat traversal appliance. Acmepacket , kagoor and Jasomi
are some examples.
Leo
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asterisk-users mailing list
To UNS
I know that NAT is something no one really likes to talk about, but does
anyone know how work with it elegantly? There are many providers which deal
with it on a daily basis in fact they cater to it, is this possible to do
with asterisk or does it require other exotic setups? I even know of a
pr
I know when you read that subject everyone thinks NAT, but that isn't the
case here. Incoming calls get 2 way audio, but outbound calls do not have
incoming audio. below is the flow
callee --> asterisk --> firewall/router --> provider
Callee is firewalled, but not NAT. callee is on th
Hi all,
Also got a problem with the RxFax app, I'm using the following packages,
spandsp-20031021
tiff-v3.6.0
Asterisk CVS-12/10/03-20:28:08
Using the Digium TE410P card on a E1/PRI line.
The tiff files under /var/spool/asterisk/incoming/, are a 8-byte file,
and a 314-byte file.
Because it could
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