[asterisk-users] Preferred Clock
Chris Knipe
savage at savage.za.org
Mon Feb 2 04:12:35 CST 2009
Hi,
We're running on a * 1.4.21 system. We run about 80 SIP Extensions, mainly
ATCOM phones (and a few Snoms - 300 and 360), and have an additional 80 IAX2
extensions to iaxmodem devices for fax2email. We are rapidly growing and
will be adding an additional PRI trunk and grow to about 150 SIP & IAX2
extensions towards the end of the year.
We have two Digium Wildcard TDM800P cards (8 x FXO and 8 x FXS) and are busy
migrating the remaining 8 analogue lines over our ISDN Pri line. The TDM
cards are without hardware echo cancellation boards. The 8 Analogue
extensions are mainly for traditional paper Fax Machines.
Our main trunk is a Digium Quad Port PRI Card (PCIe) - with a hardware echo
cancellation board:
Span 1: TE4/0/1 "T4XXP (PCI) Card 0 Span 1" (MASTER) HDB3/CCS/CRC4 ClockSource
Timing slips: 1
Only 1 of the 4 spans are being used, the other three has been disabled
completely.
We are getting very strange results on some very intermittant calls. It is
not isolated to a single extension or calls to/from a single destination.
It seems that on random occasions, there are very bad echo (calls via the PRI
trunk), it sounds like Silence Supression is being used, and there seems to be
isolated cases that seems to sound more like a sidetone than echo on actual
conversations / handsets. All these are only occuring on external calls,
internal calls are working flawlessly.
We've pretty much had this issue intermittantly since day one, had certified
Digium consultants look at it, independent audits was done, various settings
has changed and debugged from the Telco side, but those three issues persist,
and just won't go away. Everyone that looked at the system so far, gave it a
pretty clean state and congratulated us on a very well implemented
installation.
I am currently using my PRI as my clock source, as indicated above. Would it
at all be beneficial to use ztdummy as a clock source instead of the PRI?
What would normally be preferred?
Certain pre-recorded messages, notably those of meetme rooms, sound absolutely
TERRIBLE on telephones, yet, MOH, and voicemail prompts for example, sounds
absolutely perfect. I confirmed and checked, all files being used are the
same format, and same encoding. I fail to see why one recording would play
absolutely flawlessly, whilst another would sound so terrible?
I would appreciate it if more educated members can possibly engage in a
discussion surrounding these issues to hopefully resolve them. I've worked
with Asterisk before, but this is my first 'major' implementation in terms
of a PRI trunk, as well as the number of extensions involved.
Any configuration files required would be happily provided. I thank you
for your time, and effort, and I look forward to hearing from you.
Regards,
Chris.
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