] no sound between extensions
Do you agree something is blocking the audio in one direction? Can you do a
'rtp debug' and then initiate a SIP call and see if there is two way audio
traffic. Also make sure these extensions have 'canreinvite=no'.
Zeeshan A Zakaria
--
Sent from my Android phone
I have remote access to the server so I checked the canreinvite .. they
are all set to no. I can't try the call from here, I will get back to you.
Gary Baribault
On 06/01/2010 07:24 PM, Zeeshan Zakaria wrote:
Do you agree something is blocking the audio in one direction? Can you
do a 'rtp
directory (
/usr/lib/asterisk/modules).
*-THQ- !!!ONE*
Date: Tue, 1 Jun 2010 19:24:41 -0400
From: zisha...@gmail.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] no sound between extensions
Do
-0400
From: zisha...@gmail.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] no sound between extensions
Do you agree something is blocking the audio in one direction? Can you
do a 'rtp debug' and then initiate a SIP call and see if there is two
way audio traffic. Also make
Hello all,
I have Asterisk 1.4.26 installed on an OpenSuSE 11.2 server with a
Digium 8 port FXO card. The local network is 100Mbps Ethernet and my
phones are Linksys SPA-921 or Linksys Analog adaptors.
The phones are setup with DHCP, and are on the same flat non-routed
network. There is no
Incoming and outgoing calls are on SIP or on ZAP?
Zeeshan A Zakaria
--
Sent from my Android phone with K-9 Mail.
On 2010-06-01 3:28 PM, Gary Baribault g...@baribault.net wrote:
Hello all,
I have Asterisk 1.4.26 installed on an OpenSuSE 11.2 server with a
Digium 8 port FXO card. The local
Baribault
Sent: Tuesday, June 01, 2010 2:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] no sound between extensions
Hello all,
I have Asterisk 1.4.26 installed on an OpenSuSE 11.2 server with a
Digium 8 port FXO card. The local network is 100Mbps
Output of 'iptables -L -n' would also be helpful. I am sure its a NAT issue
if incoming and ougoing calls are on ZAP channels.
Zeeshan A Zakaria
--
Sent from my Android phone with K-9 Mail.
On 2010-06-01 3:53 PM, Danny Nicholas da...@debsinc.com wrote:
My assumption is that inbound/outbound
Incomming calls are on TDM lines connected to the Digium card. Calls
between extentions are on the LAN for SIP registered users/ip phones.
Gary Baribault
On 06/01/2010 03:32 PM, Zeeshan Zakaria wrote:
Incoming and outgoing calls are on SIP or on ZAP?
Zeeshan A Zakaria
--
Sent from my
Subject: [asterisk-users] no sound between extensions
Hello all,
I have Asterisk 1.4.26 installed on an OpenSuSE 11.2 server with a
Digium 8 port FXO card. The local network is 100Mbps Ethernet and my
phones are Linksys SPA-921 or Linksys Analog adaptors.
The phones are setup with DHCP
As I stated, the incoming calls are on TDM DS0s connected to the Digium
card, and the extensions are on the same local network as the Asterisk
server. There is currently no NAT anywhere.
Gary Baribault
On 06/01/2010 05:22 PM, Zeeshan Zakaria wrote:
Output of 'iptables -L -n' would also be
Do you agree something is blocking the audio in one direction? Can you do a
'rtp debug' and then initiate a SIP call and see if there is two way audio
traffic. Also make sure these extensions have 'canreinvite=no'.
Zeeshan A Zakaria
--
Sent from my Android phone with K-9 Mail.
On 2010-06-01
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