Replication completes now. However default operator got no actions
requiring parameters and it still doesn't transfer.
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I don't really understand what to do with the patches. Copy them to
certain directories? Or edit certain files? Please provide CentOS full
path of these.
Thanks
Ingomar
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Currently the default operator receives the call. Next it says to hold
while transferring the call when extension is pressed. The call never
really gets transferred though. It just remains silent. Normal transfer
works fine. What's the path of autoattendants.xml in CentOS? Maybe another
one can be
Searching "GSM" on SIPfoundry Wiki yields 0 results, so does sipX support
GSM gateways?
http://sipx-wiki.calivia.com/index.php?title=Special%3ASearch&search=GSM&ns0=1&fulltext=Search
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I got a GSM gateway setup in similar way as PSTN gateway. Unlike the
AudioCodes MP-108 (PSTN), the Sun Comm SC-375 (GSM) doesn't complete calls
when in 2-stage mode and a phone extension is chosen. It only completes
when the phone's IP is. It doesn't complete 1-stage either. Anybody else
got undesi
I tried upgrading via command line interface and it didn't work. Even
manually editing /etc/yum.repos.d/sipxecs-stable-centos.repo with the
right URL, sipX says 'HTTP Error 404: Not Found'. The CD doesn't seem to
include a way to upgrade without clean installation. Primary and redundant
servers can
Each server is the DNS & DHCP server for its LAN.
> Are you supplying multiple DNS servers to the phones through DHCP?
>
> Mike
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I tested redundant registration with physically separate servers in HA
system with DNS SRV load balancing. Only a router allows communication
between the LAN segments. The phones are connected on the Primary LAN
segment. First I make sure all phones are registered, then I unplug the
Primary server
This was the link where we could find this guide prior to decommission of
Pingtel.com...
http://www.pingtel.com/onlinedocs/AudioCodes/wwhelp/wwhimpl/common/html/wwhelp.htm?context=Working_with_Your_Gateway&file=AudioCodes_v46a_Interop_Guide.1.6.html
Where can we find it now?
Since installation our Soundpoint phones had the same time as sipXconfig
but recently without any change concerning time parameters, the phones are
55 minutes behind sipx. Last change was implementing true load
balancing and the phones kept the same time after that - independent of to
which server
What steps should one take to have BLF on Soundpoint phones? Preferred
result is that a user can see his/her contact status at all times without
even attempting to call him/her.
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When Polycom Soundpoint phones registration expire while Primary is
offline (network cable unplugged), they auto-register on Redundant server.
But when I reboot them manually before they expire with Primary offline,
they won't register. How come?
These phones get registration via TFTP. Is there ma
Today we got partial load balancing - Redundant servers that only provide
critical processes in case Primary fails to do so.
The next step can be integral load balancing - Mirror servers that evenly
distribute all features/services with Primary.
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Which other services are not supported by Redundant server? Will they be
in future releases?
> [please do not reply to an existing message when starting a new topic-
it messes up threading]
>
> On Mon, 2009-07-13 at 14:24 -0400, ingo...@netvision.an wrote:
>> MOH not heard when Primary server is
MOH not heard when Primary server is offline and calls are handled by
Redundant. Currently only Primary provides MOH.
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I'm running a test with Polycom Soundpoint 320, 501 and 650.
SipX gives this error as well...
Starting DHCP server test.
Socket Bind Exception: Address already in use
DHCPACK responce received.
Message Type: DHCPACK
Server Identifier: 192.168.10.133
Subnet Mask: 255.255.255.0
Router: 192.
I know sipXsupervisor is one of the services, but how does it relate to
Registrar (and other services)? How can it be configured?
> Make sure sipXsupervisor is working correctly between the two.
>
>
> On Fri, 2009-07-03 at 14:28 -0400, ingo...@ wrote:
>> I got phones expiring after 2000 t
I got phones expiring after 2000 to 3000 seconds. And they won't register
again by themselves. How can this be solved? I didn't have this issue when
I tested them in a non-HA configuration. Both Registrars on Primary and
Redundant servers are running OK.
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For PSTN and GSM I use respectively Audiocodes MP-108 and Sun Comm SC-375.
Audiocodes call routing works. Check out Automatic Dialing on page 18 of
this guide...
http://www.pingtel.com/onlinedocs/AudioCodes/wwhelp/wwhimpl/common/html/wwhelp.htm?context=Working_with_Your_Gateway&file=AudioCodes_v46a
Current result:
1)Someone calls in via PSTN.
2)Auto-attendant responds and the person can choose to transfer to user
phone.
Preferred result:
1)Someone calls in via PSTN.
2)User phone rings immediately without auto-attendant intervention.
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With auto-attendant disabled, incoming PSTN or GSM calls are not routed to
any extension whatsoever. How does sipX (by default) route these calls?
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