really solid recommendations, or point
me towards a more appropriate forum to request the same?
Thanks
Ed W
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suggestions on how to configure a Sangoma card for use with
a normal BT single line?
Thanks
Ed W
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the antennas into a location where 90% of
the signal isn't already killed going through walls before it has to
travel some distance is the trick. Probably also consider a repeater of
some sort rather than just one high power device
Good luck though!
Ed W
Mr Shunz wrote:
Hi,
We have an issue where Polycom's lose BLF functionality after a reboot. The
only way to fix it is to reboot the Polycoms.
Anyone else have this issue? We are using 1.4.18.
If I run 'sip show subscriptions' all the subscriptions come back after the
restart but the
?
Also, anyone understand why DCT is different between home and business
lines? Can the Zap code be changed to avoid needing something tweaking
on the exchange?
Thanks
Ed W
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if
that's required). Backups can be done very easily (make the /vserver
dir an LVM disk)
Good luck
Ed W
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, but thought it might give you something to consider...
Ed W
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only need a single channel of GSM right now (and a single SIM)
Any thoughts? Remember this needs to be production quality and priced
sensible for a commodity market
Thanks for pointers to hardware
Ed W
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-dials are setup with a + at the start of
them... Trying to fix the phone rather than the addressbook...
Thanks
Ed W
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to be rock solid, so no bluetooth to a cell phone
kind of solutions need apply.
Can any of the 3G usb devices out there offer outbound analogue calls
(ie other than via voip)?
Cheers
Ed W
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Anthony Rodgers wrote:
We tried with MS Exchange but couldn't get it to work (MS Exchange
doesn't support a master account).
It used to? Not out the box though...
Ed W
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asterisk
names should be fine though (and
you can google for details on their specs)
Ed W
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Check first using something like testmyvoip.com to get an idea of your
situation (stress the internet by opening up lots of simultaneous
downloads during the test)
Repeat: Try the above before you do anything else...
Ed W
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Hi
usecallerid=yes
cidsignalling=v23
cidstart=polarity
Although this is what the wiki recommends, I just couldn't get the
cidstart=polarity to play well with immediate=yes, I kept loosing the
callerid?
This is what I ended up with and now it avoids the annoying 2 rings
before the
the test)
Cheap fix is to get a separate DSL line and run the voice over that...
Ed W
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? It would be convenient not to have extra analogue lines in the
building if we go down the PRI route...
Grateful for any thoughts
Ed W
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recommendations on a server box (2U with space for a couple
of PCI cards would be sensible), the PRI card and also any ATA adaptors
which are known to work well with fax units
Cheers
Ed W
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Paul Hales wrote:
I know Brett and Jurgen have been pretty happy with the Snom's - Brett
even wrote an auto-provision utility for the Snom's at one time.
Yes, look at the latest Trixbox for the basic SNOM templates and then
off you go.
You setup a tftp server (easy), the phone looks for
. The link above should help you figure this out
Good luck
Ed W
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packets might be
accidently in the bulk box
Basically VOIP goes from perfect to horrible when the jitter rises and
packet loss goes up. Probably this is happening in your case
Good luck
Ed W
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each time...
Any suggestions on how to debug this further?
To my ear it sounds like what happens when you get an overflow in some
decoder code and the levels have wrapped around?
Any thoughts?
Ed W
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in internal and external numbers (not very hard).
Grateful for any thoughts
Ed W
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from Thunderbird (not Outlook). For example is HUDLite ever going to
support Thunderbird...?
Cheers
Ed W
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completely in the scenario
you describe?
I think the solution here is really that the CID type applications
become aware of prefix digits and strip them. Anyone know of good
solutions to this?
Any backend solutions to get Asterisk to hook into Exchange server etc?
Cheers
Ed W
diagnose your setup)
Ed W
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your findings - however, I'm still not clear
exactly what the problem is in your case. There are numerous kinds of
disconnect problems - which one are you having (so we know which one the
CPC fixes...)
Cheers
Ed W
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no random disconnects during calls
either.
Can you confirm that this is what you mean, or whether it's something else?
Ed W
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that the calls get hungup -
we had a 9 hour call earlier before someone noticed It's rare, but
the consequences are potentially quite dire.
Cheers
Ed W
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Hi
i'm not very happy with TDM404B voice quality, low volume
Check the gain set in the zap config file. You can increase the in/out
gain quite a bit over standard.
Echo is frequently a symptom of wrong country settings, hence wrong
impedence settings. Also endpoints matter
Ed W
Hi
Echo cancel almost works, but the users
hear
what they describe as a 'crackle' coming back when they talk.
Just a thought, but check that your gain levels are not too high?
Ed
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taken (because asterisk thinks that the handset is still in
a call) and other problems
I added an L() entry on the dialplan to limit calls to something
sensible in the meantime, but would like to get a proper workaround?
Any thoughts
Ed W
any settings yet)
Thanks for any thoughts
Ed W
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Philipp Kempgen wrote:
Ed W wrote:
I have a problem with calls not hanging up if for some reason the
physical phone dies or gets unplugged
Have you tried the RTP timeout settings in sip.conf?
Sounds exactly like what I need! Thanks
Is there no default set then??
Cheers
Ed W
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