Hi,
We've recently upgraded to 1.2 and for outgoing PRI calls we are now getting a
SIP 180 Ringing message generated by asterisk along with the RTP audio stream
with the PRI ring tone. This creates a double ring tone on most SIP devices
(Cisco 7960s are an exception and ignore the 180) that people
A Dell 1850 and TE405P didn't work very well for me.
Got a lot of static on ZAP calls. There's a whole thread about it from a
month ago!
Tried everything to fix it and in the end went back to 2 E100P cards which
did the trick.
Aaron
> -Original Message-
> From: [EMAIL PROTECTED]
> [mai
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Olle E. Johansson
> Sent: 06 June 2005 08:03
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Accountcode being ignored?
>
> Nabeel Jafferali wrote:
> >
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Cameron Schaus
> Sent: 17 May 2005 21:18
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Is SKYPE a threat or should we
> dosomething(togeth
Hi,
In regards to the previous thread about static and snapping on incoming
calls to the TE410P card when using a Dell 1850 server I now seem to be
getting significantly better call quality with two E100P cards. So far I
haven't been able to make any calls with detectable static on the line.
Reg
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kris Boutilier
Sent: 04 May 2005 21:20
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] TE410P on Dell 2650
> -Original Message-
> From: Aza [mailto:
I have a problem with a Dell 1850 and a TE410P card as do a few others who
posted over the weekend.
The problem in this case isn't so much echo but static and chop on all calls
using ZAP channels. My zttest results look pretty much the same as yours. We
were thinking it was the RAID controller but
rcial Discussion'
Subject: RE: [Asterisk-Users] Choppy Sound on PSTN End
I doubt it is the RAID controller since my Dell server isn't using one and I
have this problem...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Aza
Sent: Monday, May 02
Hi,
I have the same problem on a Dell 1850 with a TE410P, static/chop on calls
to through the TE410P, and have been attempting to narrow it down for the
last week. Interrupts don't seem to be a problem and I have two PRIs from
two different suppliers and both have the same static/chop on the line
I have the same problem on a Dell 1850 with a TE410P and have been
attempting to narrow it down. Interrupts don't seem to be a problem and I
have two PRIs from two different suppliers and both have the same
static/chop on the line so it's not the PRI.
The leading suspect at the moment is the RAID
On Sun, 17 Apr 2005, Dave Weis wrote:
> On Sun, 17 Apr 2005, Greg Boehnlein wrote:
> > On Thu, 14 Apr 2005, Rod Bacon wrote:
> >
> > > I have been frustrated by a variety of zyxel issues/products and have
found
> > > the best solution for all of them lies in a cylindrical receptacle
that sits
Hi,
Has anyone found a solution for the terrible voice quality
when using the ZyXEL 2002 units?
I have a couple connecting into my asterisk box but the
calls are so chopped up and broken that you can't have a
conversation. The same router connected to FWD gets crystal
clear calls.
FWD say the
Hi,
Does anyone have a solution to that allows an incoming call
to be forwarded to a mobile (or other billable destination)
and provide a CDR for the mobile call.
Such as:
exten -> 1001,Dial(SIP/sipclient&ZAP/g1/023423,20)
The CDR for this call will have 1001 as the destination
regardless of wh
The Message button is broken in newer versions of the GrandStream firmware
(I can vouch that it is broken in 1.0.5.16).
I know it works with version 1.0.5.0 so you might want to try that.
hth.
aza
- Original Message -
From: "David Hajek" <[EMAIL PROTECTED]>
To: "&#
I'm pretty sure if you assign account codes to your SIP and/or IAX clients
in their respective .conf files then cdr files will automatically be
generated for each individual account code in addition to the master.
No idea about how it works with real time.
hth.
Aaron
- Original Message -
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