I have some questions about the T.38 faxing capability. I have been able to
successfully setup the inbound receive fax. However, I am having problems
tracking down the format of the outbound extensions.conf SendFAX command. I
have looked at the code and it looks like it only takes a single
On Fri, 13 Mar 2009 10:43:13 +, Julian Lyndon-Smith
aster...@dotr.com wrote:
David Quinton wrote:
On Thu, 12 Mar 2009 10:21:06 +, Julian Lyndon-Smith
aster...@dotr.com wrote:
Has anyone in the UK got ANI to work on an inbound call ?
Using asterisk 1.4 trunk and zaptel 1.4 trunk,
Hi,
I'd like to implement the following:
Extension 101 calls 102 but 102 is busy and has no voicemail so 101 is sent to
a custom IVR that says something like extension $EXTEN is $DIALSTATUS. Please
try again later or dial $CODE now to notify you as soon as $EXTN is available..
So the
I've seen that the CDR manager and i think that it can be enough for
my needs, with the timestamp=yes action.
I think that it wouldn't be too much difficult to set in the
manager_event function (main/manager.c) a condition that if is set
events_on_db=yes in the manager.conf it store the
hi every body,
can anyone give me the right configuration of BRI cards; zapata.conf ,
zaptel.conf ans extensions.conf;
please help
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2009/3/14 Vieri rentor...@yahoo.com
Hi,
I'd like to implement the following:
Extension 101 calls 102 but 102 is busy and has no voicemail so 101 is sent
to a custom IVR that says something like extension $EXTEN is $DIALSTATUS.
Please try again later or dial $CODE now to notify you as soon
--- On Sat, 3/14/09, Olivier oza-4...@myamail.com wrote:
you also can make use of SIPPEER(curcalls)
Thanks. Will come in handy.
I don't know if DND is widely implemented in softphones
as users might be
tempted to simply turn softphone off
If my softphone users turned their software off
2009/3/14 Rayed Bs rayed.i...@gmail.com
hi every body,
can anyone give me the right configuration of BRI cards; zapata.conf ,
zaptel.conf ans extensions.conf;
please help
which config do you target ? b410p or junghanns ?
which asterisk version ?
2009/3/14 Vieri rentor...@yahoo.com
--- On Sat, 3/14/09, Olivier oza-4...@myamail.com wrote:
you also can make use of SIPPEER(curcalls)
Thanks. Will come in handy.
I don't know if DND is widely implemented in softphones
as users might be
tempted to simply turn softphone off
If my
On Sat, Mar 14, 2009 at 12:00 AM, Steve Underwood ste...@coppice.org wrote:
Fully open-to-the-public FAX servers tend to get just get a lot of bad
calls, many of them wrong numbers, or voice users. FAX servers for
I've definitely seen that, and have been able to either identify the
validity of
--- On Sat, 3/14/09, Olivier oza-4...@myamail.com wrote:
If I understand correctly, you're suggesting to
implement the h priority
instructions (or a hangup macro) to:
1) run a deadagi or a system() script to see if
someone has left a request
(eg. in astdb) to call-back-when-avail
Hi everybody:
I'm having a problem with asterisk 1.4.22-3 on trixbox, This server
have 8 lines connected at SIP VOIP provider vono in Brasil, all calls
going to a same queue and are answered with 4 attendent on other
network and location connected via PAP2 over nat. When the network
down, in
I have meetme working with BLF on polycom phones however when
meetme is not actually being used by anyone the 'status' of meetme
becomes idle.
Which the Polycom phone sees and produces a clock symbol and FLASHING red
LED.
Are there any 'tricks' or work-arounds to change this status to something
hello
please ho to get a free did number for my asterisk ?
also, is it pocible to assign it to a group of extentions ?
thanks!
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On Sat, Mar 14, 2009 at 12:46 PM, Meftah Tayeb tayeb.mef...@gmail.comwrote:
hello
please ho to get a free did number for my asterisk ?
also, is it pocible to assign it to a group of extentions ?
thanks!
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I'm trying to implement EM type V over a T1 and have not had much luck.
Type V sends 1's when on-hook and 0's when off-hook (for a little background)
which is the reverse of type II, which is what the em keyword in zaptel.conf
gives.
The closest I have gotten is to set the channels to cas or
On Sat, 14 Mar 2009, Meftah Tayeb wrote:
please ho to get a free did number for my asterisk ?
Yes. You did not specify which part of the world you want a DID, but
searching voip-info.org for free did should give you some clues.
also, is it pocible to assign it to a group of extentions ?
I can't force them to use star codes to set DND in astdb).
Once again, someone who underestimates the power of physical violence.
PaulH
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To
ipcomms for an incoming number, probably in RI
Outbound is 10 bucks per month, an 1.5 cents per minute
John Novack
Meftah Tayeb wrote:
hello
please ho to get a free did number for my asterisk ?
also, is it pocible to assign it to a group of extentions ?
thanks!
Meftah Tayeb wrote:
please ho to get a free did number for my asterisk ? also, is it
pocible to assign it to a group of extentions ?
On Sat, 14 Mar 2009, John Novack wrote:
ipcomms for an incoming number, probably in RI. Outbound is 10 bucks per
month, an 1.5 cents per minute
outbound
Yup, I should have thought of echo can and jitterbuffer. That completely
explains what I'm seeing. Thank you all.
Mike.
On Friday 13 March 2009 07:19:21 M Hulber wrote:
I believe it's echo and/or jitter being measured when the call is
connected as I recall it being explained. This issue
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