in your pattern as a plain character you
should put it into a character set: [.] or [!].
Mc GRATH Ricardo
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk community forum
How about if you set;
exten => _se,1,Dial(IAX2/cloud/1000,30,r)
Mc GRATH Ricardo
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk community forum at: ht
it will be very difficult to let you the causes of no voices.
Mc GRATH Ricardo
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk community forum at: https://community.asterisk.org/
New
Hi Motty Cruz
Probably it could become from missed configuration, check contexts issue.
Check SIP context=sip-phone and extension dialplan context, probably you forget
to set include or mistyping or other related to context issue.
Mc GRATH Ricardo
GRATH Ricardo
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
http://www.asterisk.org/community/astricon-user-conference
New
extension to KX-TDA600, PBX answer incoming call from analogue line through
Voice mail configured as Automated attendant, or DISA card.
The other point is line attenuation.
Mc GRATH Ricardo
--
_
-- Bandwidth and Colocation Provided
Panasonic PBX KX-TDA600 it doesn't support SIP protocol for VoIP technology it
only support H323 Trunk through 4 or 16 channels gateway card and TDM
technology with ISDN BRI and PRI card.
Mc GRATH Ricardo
--
_
-- Bandwidth
PBX or line it feels echo.
In case of acoustic analogue device it used analogue circuit, SIP used DSP
(Digital signal processing circuit) these last one is better than analogue
system.
Mc GRATH Ricardo
--
_
-- Bandwidth
Hi sean
Sorry echo canceller it's only works for FXO ports (PSTN Line).
Mc GRATH Ricardo
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every
Hi Sean Darcy
Question about "the remote party always hears an echo on it's side", strange
because eco suppression circuit is for local side.
Mc GRATH Ricardo
--
_
-- Bandwidth and Colocation Provided by http
digit form 0 to 9
_x. any pattern digit from 0 to 9 and dot it mean remnant digits could be 2 or
3, 4 ... etc., so what ever you dial on sip it will be valid.
Mc GRATH Ricardo
--
_
-- Bandwidth and Colocation Provided by http
, service dahdi restart and check result, (could a mistake on parameter
value on system.conf).
Mc GRATH Ricardo
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live
Roberto
Could you provide more details about Panasonic PBX test? model unit and
configuration details?
Best regards
Mc GRATH Ricardo
E-Mail mcgra...@mail2web.com
--
_
-- Bandwidth and Colocation Provided by http://www.api
-bloc, by the way you can close dialling number by using # key.
Best regards
Mc GRATH Ricardo
E-Mail mcgra...@mail2web.com
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us
to signalling = bri_cpe_ptmp
Good luck
Mc GRATH Ricardo
E-Mail mcgra...@mail2web.com
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs
Why you configure zaptel.conf? should configure on dahdi files
Mc GRATH Ricardo
E-Mail mcgra...@mail2web.com
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live
, chan_dahdi.conf),
PSTN side if network configuration or in service mode, should both side work
and debug in the same time.
Mc GRATH Ricardo
E-Mail mcgra...@mail2web.com
--
_
-- Bandwidth and Colocation Provided by http://www.api
to configures directory and others My Allworx manager it should
made through system ip address or domain name.
I think better idea to choose another brand, save your time! good luck.
Mc GRATH Ricardo
E-Mail mcgra...@mail2web.com
to PSTN start call handler process during
asterisk starting process or stop.
Another point it seems on your log information a call have been answered, so
it seems is working.
Good luck
PD: 0x00 it doesn't mean idle state (it mean force release)
Mc GRATH Ricardo
E-Mail mcgra...@mail2web.com
Hi could let more details of your test scenery?
Just in case of Panasonic, if you use SIP trunk resource, it need to configured
on CO Lines settings (DIL Tables) distribution method and DDI tables, for
incoming calls.
Best regards
Mc GRATH Ricardo
E-Mail mcgra...@mail2web.com
: Configuring port 0 span 1 in
NT mode with termination resistance ENABLED.
It could help by checking parameters on dahdi-channels if signalling =
bri_cpe_ptmp # The signalling for TE mode
Best regards
Mc GRATH Ricardo
E-Mail mcgra...@mail2web.com
Hi John
How about Digium G100 G200 voip gateways?
You can try with a test drive at Digium web site
http://www1.digium.com/en/products/voip-gateways
Best regards.
Mc GRATH Ricardo
E-Mail mcgra...@mail2web.com
--
_
-- Bandwidth
Niccol
How about to change tone list on indications.conf file?
Please comment call waiting line ; according to country zone or default
settings.
Good luck
Mc GRATH Ricardo
E-Mail mcgra...@mail2web.com
--
_
-- Bandwidth
ms_|Second ring burst |
So basically any kind of device should be work without any problem,
unfortunately during these process if some noises (as miss ground connection or
others) happens during the process can make failed to process caller-id
information, by the modem.
Mc GRATH Ricardo
E
exchange DR2 signalling between
Nortel and Asterisk is about 5 sec.
Best regards
Mc GRATH Ricardo
E-Mail mcgra...@mail2web.com
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us
(CCBS), Call Hold (HOLD)—by ISDN etc.
Best regards
Mc GRATH Ricardo
E-Mail mcgra...@mail2web.com
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar
Hi
I'm not agree problem could be cause from IRQ setting, I think in that way
problem should be more unstable, moreover no voice communication problem with
DAHDI service start up.
Best regards
Mc GRATH Ricardo
E-Mail mcgra...@mail2web.com
call.
Best regards
Mc GRATH Ricardo
E-Mail mcgra...@mail2web.com
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of
asterisk-users-requ...@lists.digium.com
[asterisk-users-requ...@lists.digium.com]
Sent
.
Just in case when it use enbloc PBX send whole number, TDM phones it press #
or other key setting as send digits.
By the other way check dialplan rules to resolve receiving number length, a
good practice is use and Asterisk extension to simulated call from PBX system.
Best regards
Mc GRATH
and others.
Best regards
Mc GRATH Ricardo
E-Mail mcgra...@mail2web.com
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs
Can be done calls from each system? How about to capture data with Wireshark?
I have experiences Asterisk with Panasonic with H323 without any problem.
Mc GRATH Ricardo
E-Mail mcgra...@mail2web.com
--
_
-- Bandwidth
31 matches
Mail list logo
