terisk to route the call via a middle extension which
is set to only allow the new codec. What are my options to achieve
this? (meetme just occurred to me? could that work?) Any suggestions?
Thanks
Ed W
--
_
-- Bandwidth
can offer some really solid recommendations, or point
me towards a more appropriate forum to request the same?
Thanks
Ed W
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Reg
all...
Please - any suggestions on how to configure a Sangoma card for use with
a normal BT single line?
Thanks
Ed W
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update op
hotspot which is suboptimally placed. If you
do go that route then getting 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 repea
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
>>
dering what changes you made?
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
___
-- Bandwidth and Colocation Provided by htt
s also (preserving IP addresses also if
that's required). Backups can be done very easily (make the /vserver
dir an LVM disk)
Good luck
Ed W
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22
d of the 360.
When I pushed some vendors for prices there was only a tiny gap between
the 300 and 360. Would suggest looking hard at the 360 always...
Ed W
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mai
up on the remote end, so I get a phantom 2 rings
at my end and then it stops...
No solution, but thought it might give you something to consider...
Ed W
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing li
ter and all my speed-dials are setup with a + at the start of
them... Trying to fix the phone rather than the addressbook...
Thanks
Ed W
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRI
ater for a wifi card...)
I 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
___
-- Bandwidth and Co
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
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com
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
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
ce they may not have the
correct wear levelling. Decent brand names should be fine though (and
you can google for details on their specs)
Ed W
___
Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/
--Bandwidth and Colocation Pr
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 internal
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
___
--Bandwidth and
time
Ed W
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
ownloads during the test)
Cheap fix is to get a separate DSL line and run the voice over that...
Ed W
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.dig
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
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing
ality? 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
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or u
voip 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
___
--Bandwidth and Colocation provided by Easynew
the
line. The link above should help you figure this out
Good luck
Ed W
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
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 t
the prompt 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
___
--Bandwidth and Colocation p
ht as well dump the whole "dial 9" thing 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 Asteri
suggestions on other software than Snap which does callerId lookup
from Thunderbird (not Outlook). For example is HUDLite ever going to
support Thunderbird...?
Cheers
Ed W
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users
the prefix digit and working around
any clashes in internal and external numbers (not very hard).
Grateful for any thoughts
Ed W
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options
your setup? (can be a clue to help diagnose your setup)
Ed W
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
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
___
--Bandwidth
d I see no random disconnects during calls
either.
Can you confirm that this is what you mean, or whether it's something else?
Ed W
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or updat
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
them to change any settings yet)
Thanks for any thoughts
Ed W
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
x27;t 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
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
___
--Bandwidth and Colocation provided by Easynews.com --
aster
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
meone please suggest a way to ensure 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
___
--Bandwidth and Colocation provided by Easynews.
36 matches
Mail list logo