On Friday 16 January 2009 20:27:57 sean darcy wrote:
Tilghman Lesher wrote:
On Friday 16 January 2009 17:43:21 sean darcy wrote:
Danny Nicholas wrote:
Why not do a zap restart instead of restarting asterisk? You could
write an AGI to do the ZR when the condition occurred and lines where
Yes.
That's the correct way to do it. Placing # as a rule in callnum forces
the Portech to use the number defined in the SIP INVITE packet.
Bye.
Marco.
Marco Signorini
INGEGNI Tech S.r.l.
http://www.ingegnitech.com http://www.ingegnitechcom/
Pascal Bruno wrote:
Sorry
On Friday 16 January 2009 22:43:51 Jose P. Espinal wrote:
About UpdateConfig syntax, how did you find out the correct way of sending
various sets of parameters? I was looking in google, the ATFOT v2 Book, and
nothing showed up.
I wrote a patch for a problem with that function last month, and
Are you sure that the TRANSFER is supported by the other side at all? see
http://thread.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/15267
Thanks
l.
2009/1/16 Paul bulkm...@monafamily.com
Yes, this is the first method I tried. The transfer only works if it is
done before a media path
Grey Man greymanv...@gmail.com writes:
The trick with transfers is to forget about the src field for billing
purposes and make sure the accountcode for the call is set in
accordance with the business rules. For example if two customers A and
B are talking to each other and A blind transfers B
Hi,
Has anyone compared SPA8000 vs. GXW4008 especially in terms of firmware and
hardware stability (the feature sets are apparently similar)?
Vieri
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asterisk-users
On Sat, Jan 17, 2009 at 10:39 AM, Benny Amorsen benny+use...@amorsen.dk wrote:
Only if the dial plan actually gets enough information to set the
accountcode, which at least historically wasn't the case for Asterisk.
In 1.2.x, you couldn't in the dialplan tell if a call went A-B or
A-C(SIP
after doing that (erasing Asterisk 1.4 completely and installing Asterisk 1.2)
will this impact all of the trunks configurations that are existed in FreePBX
that i made before
i mean, will i need to make something to operate all these trunks
configurations as before?.
From:
and canreinvite=yes ?
2009/1/17 Lenz Emilitri lenz.lo...@gmail.com
Are you sure that the TRANSFER is supported by the other side at all? see
http://thread.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/15267
Thanks
l.
2009/1/16 Paul bulkm...@monafamily.com
Yes, this is the first
Tilghman Lesher wrote:
On Friday 16 January 2009 20:27:57 sean darcy wrote:
Tilghman Lesher wrote:
On Friday 16 January 2009 17:43:21 sean darcy wrote:
Danny Nicholas wrote:
Why not do a zap restart instead of restarting asterisk? You could
write an AGI to do the ZR when the condition
On Saturday 17 January 2009 11:04:33 sean darcy wrote:
Tilghman Lesher wrote:
On Friday 16 January 2009 20:27:57 sean darcy wrote:
Tilghman Lesher wrote:
On Friday 16 January 2009 17:43:21 sean darcy wrote:
Danny Nicholas wrote:
Why not do a zap restart instead of restarting asterisk?
The scenario we have is fax send/recieve software that ONLY talks T38
and an asterisk box.
We have ITSP providers that do NOT talk T38 but G711 only.
Does asterisk have the capability to take the T38 call from an ATA
or T38 software then bridge/transcode it and do G711 out to the PSTN
providers?
Hello,
I read a thread on the asterisk dev list (call file handling suggestion)
May i have some comment/opinion on these two ways below to place a call file in
the future ? (from the wiki and the asterisk book but added typos and stupidity
come from me)
The best is ? (and should work ?)
- sean darcy seandar...@gmail.com wrote:
Doug Bailey wrote:
- sean darcy seandar...@gmail.com wrote:
pstn incoming on a TDM400P, sometimes i* won't answer, going into
a loop like this:
-- Starting simple switch on 'DAHDI/4-1'
[Jan 16 10:38:55] NOTICE[5808]:
On Sat, Jan 17, 2009 at 7:52 PM, didier.cuffaut didier.cuff...@neuf.fr wrote:
May i have some comment/opinion on these two ways below to place a call file
in the future ? (from the wiki and the asterisk book but added typos and
stupidity come from me)
The best is ? (and should work ?)
This
Is it possible for asterisk to support multiple USB audio devices
independently as Console/Dsp???
At the moment a call comes in and routes to Console/Dsp which is the
sound card on the motherboard.
What if I needed another audio device so I added a second or third USB
audio device. How do I
On Sat, 17 Jan 2009, randulo wrote:
On Sat, Jan 17, 2009 at 7:52 PM, didier.cuffaut didier.cuff...@neuf.fr
wrote:
May i have some comment/opinion on these two ways below to place a call file
in the future ? (from the wiki and the asterisk book but added typos and
stupidity come from me)
Hi
Can anybody help me on this ?
I am using Asterisknow 1.5.0-Beta(Freepbx)
I am having a problem getting the sip trunks to register.
It makes no different which provider one is using.
Trunk name: callcentric
Peer Details:
context=from-pstn
fromdomain=callcentric.com
fromuser=1777xxx
Can I activate/deactive the canreinvite SIP flag on the dial plan?
The idea is to allow reinvite only for exten - exten calls, and not for
outbound calls
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asterisk-users mailing
core set vervose 100
reload
and see what error throw the terminal.
2009/1/17 Zamtron Spain zamt...@terra.es
Hi
Can anybody help me on this ?
I am using Asterisknow 1.5.0-Beta(Freepbx)
I am having a problem getting the sip trunks to register.
It makes no different which provider one
Hi All,
I got this error:
[Jan 18 09:56:58] WARNING[9617]: ast_expr2.fl:407 ast_yyerror:
ast_yyerror(): syntax error: syntax error, unexpected $end, expecting
'-' or '!' or '(' or 'token'; Input:
1
^
exten = s,3,GotoIf($[${GROUP_COUNT(${ARG1})} ${LINELIMIT}]?101)
exten =
Have canreinvite set for your internal extens.
You can also have canreinvite enabled by default for all and use one or more of
the 't','T','h','H','w','W' or 'L' options set in your dial commands which will
override the canreinvite option and not send re-invites.
cheers
- Ben
--- On Sat,
Doug Bailey wrote:
- sean darcy seandar...@gmail.com wrote:
Doug Bailey wrote:
- sean darcy seandar...@gmail.com wrote:
pstn incoming on a TDM400P, sometimes i* won't answer, going into
a loop like this:
-- Starting simple switch on 'DAHDI/4-1'
[Jan 16 10:38:55] NOTICE[5808]:
sean darcy wrote:
Doug Bailey wrote:
- sean darcy seandar...@gmail.com wrote:
Doug Bailey wrote:
- sean darcy seandar...@gmail.com wrote:
pstn incoming on a TDM400P, sometimes i* won't answer, going into
a loop like this:
-- Starting simple switch on 'DAHDI/4-1'
[Jan 16
On Saturday 17 January 2009 13:49:16 Gordon Henderson wrote:
On Sat, 17 Jan 2009, randulo wrote:
On Sat, Jan 17, 2009 at 7:52 PM, didier.cuffaut didier.cuff...@neuf.fr
wrote:
May i have some comment/opinion on these two ways below to place a call
file in the future ? (from the wiki and the
We have a caller ID from our phone provider Shaw Cable (digital phone) and it
was working OK until recently.
I get an error:
WARNING[6769]: chan_sip.c:8553 check_auth: username mismatch, have 4, digest
has pstn-
NOTICE[6769]: chan_sip.c:14316 handle_request_invite: Failed to authenticate
Who sends the 500 failure code? Asterisk or the VOIP supplier through which
you got the call? Note that Asterisk has the basic mechanism for call
trasnsfer, just as you transfer a call, so the problem is either in using
Transfer() inside IVR context, or the provider.
As David noted - use
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