Hello,
If someone has an example of configuration for Cisco AS5300 / Asterisk,
I am very interested.
Thank you,
Mickael
Le 28/12/12 00:48, Mickael MONSIEUR a écrit :
Hello,
I'm trying to connect a Cisco AS5300 has Asterisk, but I have a problem.
Sound from POTS -> Asterisk does not work. (I
Hello,
I'm trying to connect a Cisco AS5300 has Asterisk, but I have a problem.
Sound from POTS -> Asterisk does not work. (In the sense Asterisk -> POTS
it works!!)
The problem lies in two directions (call initiated from the Asterisk or
POTS)
I have no firewall between Asterisk and Cisco. (it's a
The problem has been fixed.
We are able to hear audio in our calls after adding these lines in the
AS5300 config:
sip-ua
g729-annexb override
There's an issue regarding codec matching in IOS versions 12.3(18) or higher:
https://supportforums.cisco.com/docs/DOC-3186
On Tue, Jan 10, 2012 at 10:3
Here's the cisco AS5300 settings from our provider
codec preference 1 g729r8
codec preference 2 g729br8
codec preference 3 g723r53
codec preference 4 g723r63
codec preference 5 g723ar53
codec preference 6 g723ar63
On Mon, Jan 9, 2012 at 5:18 PM, Roi Stork wrote:
> Hi Alex, here's the config and
Hi Alex, here's the config and the sip debug output.
Guide:
xxx.xxx.xxx.xxx - the provider's cisco as5300 trunk ip add
yyy.yy.yy.yy - our asterisk 1.6.2.14 server
sip config:
type=peer
disallow=all
allow=g729
host=xxx.xxx.xxx.xxx
fromdomain=xxx.xxx.xxx.xxx
dtmfmode=rfc2833
nat=no
canreinvite=yes
You are hereby encouraged to post your AS5300 IOS config, sip.conf peer
declaration, and packet capture. Those three things would aid greatly in
diagnosis, especially the capture.
--
This message was painstakingly thumbed out on my mobile, so apologies for
brevity, errors, and general sloppines
Hi,
We have a problem connecting to a Cisco AS5300 trunk.
We set the sip peer to allow only g729. The call attempt is able to
connect, but when answered, no audio is heard or transmitted.
Our asterisk version is 1.6.2.14 . Codec is licensed, bought from Digium.
We do not have this problem on ou
I know this is a bit off the thread
But I am trying to see if anyone in here know how to config a AS5300 with 2
T1.
Please contact me off list if you can give me a bit of help
Sam Tam
___
-- Bandwidth and Colocation Provided by http://www.api-digital.c
Andrew Pogrebennyk wrote:
Hello Yusuf
yusuf wrote:
Hi all,
I realize this is OT.
I just got a Cisco AS5300, and I need to configure it like such:
Asterisk -(H323/SIP)--> Cisco - (E1/PRI)--->Telco
So calls originate from the Asterisk side (registered users on SIP or
just Z
e
like that.
I hope this helps.
Mark
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Pogrebennyk
Sent: Thursday, January 04, 2007 5:31 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Cisco AS5300
He
Hello Yusuf
yusuf wrote:
Hi all,
I realize this is OT.
I just got a Cisco AS5300, and I need to configure it like such:
Asterisk -(H323/SIP)--> Cisco - (E1/PRI)--->Telco
So calls originate from the Asterisk side (registered users on SIP or
just ZAP phones), and they go out
Hi Yusuf -
I just got a Cisco AS5300, and I need to configure it like such:
Asterisk -(H323/SIP)--> Cisco - (E1/PRI)--->Telco
So calls originate from the Asterisk side (registered users on SIP or just ZAP
phones), and they go
out H323 or SIP to Cisco, where they go out PRI.
Hi all,
I realize this is OT.
I just got a Cisco AS5300, and I need to configure it like such:
Asterisk -(H323/SIP)--> Cisco - (E1/PRI)--->Telco
So calls originate from the Asterisk side (registered users on SIP or just ZAP phones), and they go
out H323 or SIP to Cisco, where
> OK, here goes my next problem.
>
> I have puchased a DID which I can connect to via SIP
>
> I have been given the following details:
>
> Username: uka1xx
> Password: 1000xx
>
> Server: br.net:5160
>
> My equipment is Asterisk CVS HEAD on Red Hat EL 3.0 (NO NAT)
>
> The other en
OK, here goes my next problem.
I have puchased a DID which I can connect to via SIP
I have been given the following details:
Username: uka1xx
Password: 1000xx
Server: br.net:5160
My equipment is Asterisk CVS HEAD on Red Hat EL 3.0 (NO NAT)
The other end is a Cisco AS5300 (NO NAT)
OK, here goes my next problem.
I have puchased a DID which I can connect to via SIP
I have been given the following details:
Username: uka1xx
Password: 1000xx
Server: br.net:5160
My equipment is Asterisk CVS HEAD on Red Hat EL 3.0 (NO NAT)
The other end is a Cisco AS5300 (NO NAT)
; -Original Message-----
> > From: Areski [mailto:[EMAIL PROTECTED]
> > Sent: 29 September 2003 17:08
> > To: [EMAIL PROTECTED]
> > Subject: RE: [Asterisk-Users] cisco AS5300 : problem configuration
> >
> >
> > Hello,
> >
> > Below the IOS config f
; From: Areski [mailto:[EMAIL PROTECTED]
> > Sent: 29 September 2003 17:08
> > To: [EMAIL PROTECTED]
> > Subject: RE: [Asterisk-Users] cisco AS5300 : problem configuration
> >
> >
> > Hello,
> >
> > Below the IOS config file.
> > Shoul
7;s on that box I'd suggest you comment out the bindaddr line altogether.
> -Original Message-
> From: Areski [mailto:[EMAIL PROTECTED]
> Sent: 29 September 2003 17:08
> To: [EMAIL PROTECTED]
> Subject: RE: [Asterisk-Users] cisco AS5300 : problem configuration
>
>
---
> > From: Areski [mailto:[EMAIL PROTECTED]
> > Sent: 29 September 2003 14:02
> > To: [EMAIL PROTECTED]
> > Subject: [Asterisk-Users] cisco AS5300 : problem configuration
> >
> >
> > Hi all !!!
> >
> >
> >
> > I m trying to
config file
to select the preffered codec and when I change this to G.729/A-law/U-law all works
perfectly for me.
> -Original Message-
> From: Areski [mailto:[EMAIL PROTECTED]
> Sent: 29 September 2003 14:02
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] cisco
Hi all !!!
I m trying to setup a cisco AS5300 and I ve got some problem !!!
During a call test I m getting this error message all the time.
NOTICE[15371]: File rtp.c, Line 263 (process_rfc3389): RFC3389 support
incomplete. Turn off on client if possible
[general]
port = 5060
Hi to all!
I have this config,
PSTN <--> AS5300 <--> ASTERISK
I am using the Cisco as5300 to receive incoming calls
and routing them to Asterisk for IVR.
When I ran asterisk this is what I get when calling
the voicemail demo.
*CLI> -- Executing Playback("SIP/-081058
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