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 NAT involved. If I call from extension 6000 to extension 6001, or vice-versa both are SPA-921s. The 6001 rings, but when the phone is picked up, I have no sound. I have the same problem between any phones in the system, but this is the simplest example. Incoming calls and outgoing calls work fine, sound is correct. Voice mail works fine as well, the IVR works great. Any ideas? Gary Baribault
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" <gary at 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 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 NAT involved.
If I call from extension 6000 to extension 6001, or vice-versa both
are SPA-921s. The 6001 rings, but when the phone is picked up, I have
no sound. I have the same problem between any phones in the system,
but this is the simplest example.
Incoming calls and outgoing calls work fine, sound is correct.
Voice mail works fine as well, the IVR works great.
Any ideas?
Gary Baribault
--
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My assumption is that inbound/outbound calls are DAHDI and that internal
calls are SIP. Can OP post "core show channels" from working and
non-working calls?
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Gary 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 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 NAT involved.
If I call from extension 6000 to extension 6001, or vice-versa both
are SPA-921s. The 6001 rings, but when the phone is picked up, I have
no sound. I have the same problem between any phones in the system,
but this is the simplest example.
Incoming calls and outgoing calls work fine, sound is correct.
Voice mail works fine as well, the IVR works great.
Any ideas?
Gary Baribault
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
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" <danny at debsinc.com> wrote: My assumption is that inbound/outbound calls are DAHDI and that internal calls are SIP. Can OP post "core show channels" from working and non-working calls? -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bou... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100601/aef3a528/attachment.htm
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 Android phone with K-9 Mail. > >> On 2010-06-01 3:28 PM, "Gary Baribault" <gary at baribault.net >> <mailto:gary at 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 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 NAT involved. >> >> If I call from extension 6000 to extension 6001, or vice-versa both >> are SPA-921s. The 6001 rings, but when the phone is picked up, I have >> no sound. I have the same problem between any phones in the system, >> but this is the simplest example. >> >> Incoming calls and outgoing calls work fine, sound is correct. >> Voice mail works fine as well, the IVR works great. >> >> Any ideas? >> >> Gary Baribault >> >> >> >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100601/ac7368e8/attachment.htm
This is done while the calls are active? I just issued the command and got nothing, but there where no active calls. Gary Baribault On 06/01/2010 03:45 PM, Danny Nicholas wrote:> My assumption is that inbound/outbound calls are DAHDI and that internal > calls are SIP. Can OP post "core show channels" from working and > non-working calls? > > -----Original Message----- > From: asterisk-users-bounces at lists.digium.com > [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Gary 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 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 NAT involved. > > If I call from extension 6000 to extension 6001, or vice-versa both > are SPA-921s. The 6001 rings, but when the phone is picked up, I have > no sound. I have the same problem between any phones in the system, > but this is the simplest example. > > Incoming calls and outgoing calls work fine, sound is correct. > Voice mail works fine as well, the IVR works great. > > Any ideas? > > Gary Baribault > > > >
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