Hello,
I'm sorry to warm up this thread again, but I think I just went into the
same pit and need some advice.
We're facing voice stuttering or disruption since ever (it just happens one
or two times a week) but it is getting worse since 3-4 weeks I think. After
digging the mailling list I
Hello,
I am trying to route calls from one gateway to another one, both connected to
sipX.
I have tried a lot of combination, 2 sip trunks, 2 unmanaged gateways, 1T and
1unmanagedGW an so.
I have studied the DNS point of view and i am sure that sipX can resolve
pertinent DNS records
Have you tried capturing the traffic between sites and measuring it rather
than just assuming what's going on?
On Tue, Apr 24, 2012 at 6:00 AM, Claas Hilbrecht
claas.hilbrecht+maillinglists.sipx...@linum.com wrote:
Hello,
I'm sorry to warm up this thread again, but I think I just went into
Are you on the most current release of 4.4?
This was a problem with older releases of 4.4. Once you install from ISO
you must 'yum update'.
Thanks,
Mike
On Tue, Apr 24, 2012 at 6:07 AM, Marand Remi rmar...@prosodie.com wrote:
**
Hello,
** **
I am trying to route calls from one
Have you tried capturing the traffic between sites and measuring it
rather than just assuming what's going on?
Only with iftop so far and that shows the VPN link is saturated to 100%. I
can use tcpdump next to create a capture and load this into wireshark.
--
Claas Hilbrecht
yea, I think the key is to inspect all of the traffic going through your
tunnel to see what it is. Once you know what it is, then you can go about
looking for resolutions.
also, whatever you're using for VPN should be able to prioritize voice
traffic above all else. otherwise you'll continue to
yea, I think the key is to inspect all of the traffic going through your
tunnel to see what it is. Once you know what it is, then you can go
about looking for resolutions.
I'm trying to figure out how the BLF related traffic looks like.
also, whatever you're using for VPN should be able to
Well subscriptions will look like SIP traffic.
And, even though you only have 'voice' traffic going over that line,
there's really much more than that. There's voice traffic (RTP),
signalling traffic (SIP) and server replication traffic of registrations
and CDR. RTP is much more important than
All,
Having a newly found problem where we have assigned 12XX to a location,
server IP is 172.16.20.8. From our main office at 192.168.1.8, calls
destined to 12XX seem to route to that server only. Dialplan says 12 and
2 digits send to 172.16.20.8 gateway. Doing a network grep our 4.4
server
does the other system also have a gateway from your system? if not, it
might not pass a check which shows it is an allowed call in that manner.
On Tue, Apr 24, 2012 at 9:45 AM, Aaron Pursell aar...@esgw.org wrote:
All,
Having a newly found problem where we have assigned 12XX to a location,
The 12xx gateway has a gateway setup for 37xx and the appropriate
dial-plan.
Aaron Pursell
Network Systems Administration
Easter Seals-Goodwill Northern Rocky Mountain, Inc.
4400 Central Ave
Great Falls, Montana 59405
(406) 771-3721
aar...@esgw.org
Tony Graziano tgrazi...@myitdepartment.net
Is the call an intra or inter domain call?
On Tue, Apr 24, 2012 at 10:36 AM, Aaron Pursell aar...@esgw.org wrote:
The 12xx gateway has a gateway setup for 37xx and the appropriate
dial-plan.
Aaron Pursell
Network Systems Administration
Easter Seals-Goodwill Northern Rocky Mountain,
Yes
Aaron Pursell
Network Systems Administration
Easter Seals-Goodwill Northern Rocky Mountain, Inc.
4400 Central Ave
Great Falls, Montana 59405
(406) 771-3721
aar...@esgw.org
Tony Graziano tgrazi...@myitdepartment.net 4/24/2012 8:41 AM
Is the call an intra or inter domain call?
On Tue,
Thats not a valid answer. Your answer should be in the form of either
intra or inter. (Inside the same sipdomain--intra, or between two
different sip domains--inter)
Can you call from a UA to the same destination in the same format? ie.e.
1234@ip? Is the dialplan entry CUSTOM or SITE TO SITE
Inter, its a custom but I've tried it every way I can think of. So back
to the drawing board.
Aaron Pursell
Network Systems Administration
Easter Seals-Goodwill Northern Rocky Mountain, Inc.
4400 Central Ave
Great Falls, Montana 59405
(406) 771-3721
aar...@esgw.org
Tony Graziano
I am in the need to query the sipXecs system of the fax Service DID/Alias
numbers that I have entered for my users. If I use search I am unable to locate
the user account with the DID/Alias assignment. The information is not
available in the export report as well.
If I grep the
did you try calling from a UA at one end in the same form?
1234@ipaddress) to reach the foreign system?
On Tue, Apr 24, 2012 at 11:06 AM, Aaron Pursell aar...@esgw.org wrote:
Inter, its a custom but I've tried it every way I can think of. So back
to the drawing board.
Aaron Pursell
its in the alias file, see:
/var/sipxdata/sipdb/alias.xml
see also your last query about this on March 21 for the same information.
See also http://track.sipfoundry.org/browse/XX-10127
On Tue, Apr 24, 2012 at 11:06 AM, Robert Schroeder
robert.schroe...@memberfirstmortgage.com wrote:
I am
is it an internal (vpn or locally connected) call?
if it is, do the subnets on each side reflect a local subnet (to not invoke
media relay)?
i am thinking if each system can see the other system and look it up via
dns and resolve the SRV to the local address or use a simple gateway method
(not
WoW, Ouch!!
Thank you for the update. I do not remember asking however it would appear that
I did.
Thanks again for the help and assistance,
Rob
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano
Sent: Tuesday, April
VPN.
We use standard internal subnets 172.16 and 192.168 for all networks.
All the subnets are there, but it is what it is, the call on the
192.168.1.x network doesn't even leave our sip server even though the
dial plan says shoot it over to 172.16.20.8.
I'm sure its an mistake on our end
Can you duplex care if from a non-dialplan issue? Phone dials 123@IP
If not then maybe the con is not actually allowing the traffic. I have
ipsec Vpn connections but then I have to allow the traffic (filter) to
actually use it.
Can sipx1 trace route to sipx2, vice versa.
On Apr 24, 2012 12:02
I would use the model 4524 instead of 4114.
On Apr 24, 2012 2:48 PM, Bryan Anderson branderso...@msn.com wrote:
Sorry I sent this yesterday from the wrong email account. Since I tried
sending that though, It looks like we might be going with a patton. I was
looking at the SN4114/JO/EUI? Is
On 4/17/2012 5:49 PM, Cyril Constantin wrote:
Hi Guys,
I just would like to know if there is any plan to integrate Jitsi into
provisioning phone?
They now have a stable release since beginning of April.
http://jitsi.org/
Thanks a lot for your feedback.
We have been using this for a few
(4500 series have two ethernet portsand are typically more flexible in the
event a basic sip connection has to be used from it (i.e. some people use
them to connect one ethernet port to a sip provider as it has a nat
function and bring in trunks for non-sipx related stuff too). Unless you
have a
Uncheck your monitored speed dials and see if that settles things down - it
will. I had to setup a separate sipXecs site at our remote so they could
monitor speed dials.
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Michael Picher
there is indeed a loop going on there after a hold attempt. It seems
you can now reproduce this at will. sipx bridge debug level log would
help pinpoint where the loop is happening.
On 04/24/2012 05:58 PM, Sven Evensen wrote:
We have this call scenario (sipX 4.4)
Ext calls to int A (860)
27 matches
Mail list logo
