The two boxes are labeled as per the town they are in: Leongatha and Korumburra.
The receptionist is in Korumburra.
When a call comes in off the PSTN in Leongatha, the first number in the call
queue is the receptionist. If she answers it, then the media flow looks like
this:
PSTN -> Leongatha -> IAX Trunk -> Korumburra -> Receptionist Phone
If she then transfers the call back to a Leongatha extension, the media path
looks like this:
PSTN -> Leongatha -> IAX Trunk -> Korumburra -> IAX Trunk -> Leongatha -> IP
Phone
I believe that it is possible to stop this 'bouncing' of the call from
happening by using the re-invite feature. However, taking the Trixbox's out of
the media path is undesirable, as they client needs to be able to record calls.
Also, doing this does not 'fix' the underlying problem.
They are also having some issues with outbound calls from Leongatha (over
VoIP), and they are having no real issues at Korumburra.
Many Thanks,
Daniel Cole (CCNA)
P Please consider the environment before you print this e-mail or any
attachments.
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alexander Lopez
Sent: Wednesday, 12 December 2007 3:11 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Call Quality Issues With 2 Trixbox's -
RouterIssue?
How are the calls being transferred from Box A to Box B?
On what box is the receptionist registered too?
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Daniel Cole
Sent: Tuesday, December 11, 2007 9:00 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Call Quality Issues With 2 Trixbox's - RouterIssue?
Hello Everyone,
We have recently installed a pair of Trixbox servers in for a client of our.
They have two locations, with one server each. The servers terminate 3 standard
POTS lines into a Sangoma A200D card. The servers are IBM x3250 1RU servers
(1GB Ram, Raid 1 160GB HDD, Dual Core Xenon Processors). We are using Trixbox
2.2, and G729 all around.
Each site has two (2) 512k/512k ADSL connections terminating into a Cisco 877W
router (using an additional 'dumb' modem in a separate VLAN for the extra dsl
connection). Using policy based routing, all Voice Data goes over one DSL
connection (the one that terminates directly into the router), and all other
traffic (e.g. Web and VPN) goes out the second connection (the bridged dumb dsl
modem).
We are also the ISP for this client, and as thus we have full monitoring of our
Layer 2 and Layer 3 networks. From our analysis, it doesn't appear that there
is any issue in these networks. We have other customers using the VoIP service,
who have not complained of these issues.
Now for the Fun part!
The client is complaining of issues with inter-site calls. They are reporting
issues with crackly and broken speech, and horrible jitter (or packet loss).
This presents a huge issues, because they have one receptionist answering all
calls for both sites. So if a call comes in from the other site, it
automatically an inter-site call, and the quality falls out of it. If the call
is then transfered back to the originating site, the audio 'bounces' between
the two sites, which add to the call quality degradation.
We have been monitoring the router while these incidents have been reported,
and it does not appear to be a bandwidth issue. The DSL tail used for Voice
gets to no more then 120k in each direction (we have tested the links, and can
pull data at 53k/s between sites). CPU usage floats at around 20-25% under
load. The router has only shows major packet loss (that we can tell) when
REALLY pushing it in testing (e.g. 10+ calls between sites).
We have enabled the SIP jitter buffer, as well as the IAX jitter buffer, which
appeared to make a huge difference, but the issue is still ongoing.
These issues have also been reported with some outbound VoIP calls. Internal
calls, and calls directly in or out of the Sangoma card are clear, with no
issues reported.
Does anyone have any thoughts on what could be causing these issues? We have
been racking our brains here, and have tried everything that we can think of.
These system is a million times better then what is what when it was first
installed, but it is still not where it should be in terms of quality.
Any thoughts/ideas are most welcome.
Thank you
Daniel Cole (CCNA)
P Please consider the environment before you print this e-mail or any
attachments.
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
