Displaying 20 results from an estimated 29 matches for "stepaniuk".
2009 Oct 18
4
Customising Firmware
Hi,
Does anyone have any advice on customising firmware of an SPA921 so that
it can be locked to a sip provider and display logos on the config
pages.
Many thanks
Dan Journo
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2009 Sep 07
2
Echo and Playtones not working on SIP after upgrade
..., moh, and everything else we are using works just fine.
We are using Asterisk 1.4.21.2~dfsg-3 (on debian stable), SIP channels
with both grandstream and soft phones. Everything on the same network
segment.
Codec does not seem to affect this behavior (tried them all)
Any clues? Thanks!
--
Iv?n Stepaniuk
Alba Fot?nica S.L.
2009 Oct 05
5
Networking Concept
Hello,
I would like to know how Asterisk deal in this case:
Assume I have a Main Asterisk Server located in UK, and another box that
have PSTN interfaces located in China, now the purpose is to FW calls
through PSTN.
Assuming I have a client who is calling from Japan to my main switch in UK
and he is calling China, (japan have latency around 500ms to UK and 100ms to
China), how asterisk
2009 Dec 09
1
Problem with Asterisk and SPA-3000
...ension (and transfers the PSTN leg to the new extension as
normally).
At the CLI there is nothing but a new incoming call from the SPA,
exactly as the original call.
It seems to happen with both asterisk 1.2 and 1.4, I am quite lost, Does
anyone know what could be causing this problem?
--
Iv?n Stepaniuk
Alba Fot?nica S.L.
www.albafotonica.com
2009 Sep 10
2
ASR & ACD
Is there any program Asterisk users use to calculate ASR and ACD ??
Thanks for any comments.
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2009 Oct 08
4
No sound on voicemail from analog line
Hello.
I have a server installed with asterisk 1.6. I have a PSTN line that comes in to one of those clone cards. Everything seem to be working fine. The only problem I have is that I can't get voicemails coming from the PSTN line. All other: SIP, IAX work fine. I can hear those ok but, when it comes to a call that comes in from PSTN I get no sound.
What can cause that problem?
Thanks in
2009 Oct 11
5
Call Recording and Posting
Hello,
I'm working on a call recording solution. I would like recordings to
either be automatically uploaded via FTP, or posted to a URL for
processing by our main server.
Is Asterisk capable of doing this or will I have to create a separate
application that monitors a temp directory for new recordings?
I ask because I don't have any experience in Linux programming, so I
2009 Sep 09
2
Call getting stucked !!
I am using asterisk.
I also have an access to VOIPSwitch ver 2 where I can see live calls.
Many times I have seen that my calls are getting strucked and then it gets
disconneected after 59 mins ( as settings are done accordingly in
VOIPSwitch)
What could be the reason ?
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2009 Sep 15
3
Which is best provider for G.729
hello
I dont want to disgrace any company but i want to know from
your(user)experience which one is good in case of g.729 (performace etc)
is it Howler(http://www.howlertech.com/products/howlets)
OR its Digium (
http://store.digium.com/productview.php?category_id=5&product_code=8G729CODEC&main_category_id=5
)
plz note i dont want to degrade any company... But to know what experience
you
2009 Sep 29
2
kill sip user
I have a user but I need to give that user only kill and disable all
connection cut calls what is the command in the CLIC
--
Bayardo S?nchez Garc?a
Web Developer - Internet Portals - Asterisk Support - Windows Server Support
- Proxy Support - Linux Server
E-mail: bayardo.sanchez at gmail.com
Linux User: #418392
America Central - Managua, NI (505) 2249-2853 - 84886876
IM msn messenger:
2009 Sep 30
1
How to finish a Meetme
Hi people, I want to make a meetme between 2 numbers.
First I enter the number1 into the meetme. It is waiting for the other number, but the other number never entered, so, how can I finish the meetme from the dialplan?. Is it posible by using MeetmeAdmin and kick all the users?
Thanks,
Anahi Ludue?a
_________________________________________________________________
Descubre
2009 Oct 01
1
DTMF problems during a message play
I'm using the latest asterisk-1.4.26.2 and no zaptel trunks used, all SIP.
I have one user that is having problems once he connects to asterisk.
He's dialing from his home phone (pstn) to a Vitelity DID (SIP Trunk)
which goes to my asterisk IVR.
If he presses a dtmf during any message, the press is ignored unless the
press was a #, 0 or *. Otherwise, he needs to wait for the
2009 Oct 02
1
Problem with inbound calls - asterisk 1.6.1.6
Hi all,
I have a new installation with asterisk 1.6.1.6 but I'm unable to
receive calls from a SIP trunk:
[Oct 2 14:30:09] NOTICE[21554]: chan_sip.c:18523
handle_request_invite: Call from 'user001' to extension 'user001'
rejected because extension not found.
Are there any changes from 1.6.0 to 1.6.1 (or there is a bug)?
Below my simple configuration:
sip.conf
2009 Oct 10
1
Asterisk to Asterisk access voicemail - not working
Asterisk to Asterisk voicemail not working (accessing voicemail from another asterisk).
PSTN to Asterisk is working, but not between two asterisk :-(
I've tried setting my asterisk dtmf to rfc2833, inband it is not working.
The other Asterisk Linksys is set dtmf = auto
--
Joseph
2009 Oct 14
2
ACD & ASR
Is there a ready add-on to asterisk that will display the ACD/ASR per
channel, source & destination?
Thanks.
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2009 Oct 16
1
The City of Amsterdam has been deploying asterisk throughout the city!
Hi,
As you may know by now, yesterday on the Astricon the City of
Amsterdam presented their large scale asterisk deployment of
20000 phones. Because they do not allow brand names to be used
within the city, they call it 'IP Business Manager', but the
software they use is in fact the Astium PBX, by NeoNova.
Since we are very proud of this project, we have made the Astium
available for
2009 Oct 16
2
SIP to IAX to SIP
Hi all,
I have a machine running Ubuntu that I run Asterisk 1.4.x on and it runs
very well. On that machine I have a SIP phone. I have configured a
netgear wgt634u with asterisk and a SIP phone and linked the two systems
together via IAX. Audio from Ubuntu to netgear is not bad, audio from
netgear to ubuntu is unintelligible. Any clues as to whether this will
work? Configuration suggestions?
2009 Nov 10
2
Gradstream Budge Tone-201
Hi All;
I just need to know the openion about Grandstream phone, actually I tried Budge Tone 201 and I chocked that there is a noise in the handset (zzzzzzzzzzzzzzzzzzzzzzzzzzz) always, but in the speaker the sound is good and no noise.
Anyone has idea about Grandstream, and if they have a lot of problems and such noise in handset? Or my luck was bad that this phone is defected?
Regards
Bilal
2009 Nov 10
1
Silent Dialing
Is there a way to disable ringing while dialing?
Example, external users come into our IVR, and if they dial certain IVR
options, these are sent off to a remote server for call handling (
Dial(SIP/extension at remoteserver) for example).
It rings once, then the remote system picks up. I would like it to be
more transparent to the users.
2009 Sep 14
3
G.729 for Asterisk
Hello
I have a confusion relating to G.729 codec.
I know how to install where to get license but i really don't know why we
need it?
Why people use G.729 codec with asterisk?
look all functionality can be done with out it ie calling from sip to iax
protocol and sip/ iax to E1, then why we need this??
regards
Adam
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