limit etc.
--
Regards,
Sammy
On Wed, Oct 19, 2011 at 12:27 AM, Aksel Celasun
mailto:ak...@abacus-it.no>> wrote:
Thank you for the reply.
The Asterisk is behind a firewall, but not in a dmz, been thinking of placing
it in a dmz soon, maybe that will solve the problem.
Or else, I will try
erver in DMZ mode.
There can be many other guesses, but the above is a good start.
--
Zeeshan A Zakaria
PBX - visionvoip.com
Blog - ilovetovoip.com
On 18/10/2011 10:02, Aksel Celasun wrote:
Thank you for replying
My sip.conf is
limitonpeers=yes
limitonpeer=yes
allowsubscribe=yes
Maybe there is something with the sip client, qualify=yes?
;Sentralbord
[501]
type=friend
secret=501
host=dynamic
context=phones
mailbox=501@defualt
callerid=Sentralbord Abacus-IT
qualify=yes
Thank you in advance.
Regards
Aksel Celasun
Fra
Hello dear list.
We run a Asterisk 1.6.2.6 on testbasis (SIP), and experience every day, when
making calls, that the calls become silent.
Not every calls, but 1 out of 3-4 calls, becomes silent suddenly during the
conversation.
When we then hangup, and redial immediately, the calls do not go thr
Hello there
You should have a look at features.conf
Regards Aksel
Fra: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] På vegne av Mike
Sendt: 28. juni 2010 21:39
Til: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Emne: [asterisk-users] Ast
gards
Abacus IT AS
- din Visma Software Partner
- your Visma Software Partner
Tor Aksel Celasun
Mobilnummer/cell phone: (+47) 900 15 103
Sentralbord/Support 4000 1850
ak...@abacus-it.no<mailto:ak...@abacus-it.no>
--
_
-- Ban
Hello, and thank you for your response.
When I push transfer, the buttons with the function "transfer" disappears, and
then I enter the sip number,
Wait 10 seconds and then it transfers with the MOH in the background, when the
connection/channel is made,
Then transfer button is revealed again su
>>And I can't see any button on the Cisco phone which will function like a
>>"direct transfer now", do I have to wait...?
Thank you for your reply.
In my Dialplan menu on the SPA525g, I have a field where the input are, and I
must say, I don't know if this is the right one, but the field conta
nsfer/send to another sip phone?
Ex. Push *200 and the SIP phone will directly call SIP/200. Or push *401 and
the Sip phone will directly call SIP401?
Default features.conf context.
Thank you.
Med vennlig hilsen / Best regards
Abacus IT AS
- din Visma Software Partner
- your Visma Software P
: Asterisk Users Mailing List - Non-Commercial Discussion
Emne: Re: [asterisk-users] Error trying to add context: Context 'internal'
tries to include nonexistent context 'nighttime|12:30-8:00|mon-fri|*|*'
On Friday 18 June 2010 09:49:39 Warren Selby wrote:
> On Fri, Jun 18,
Discussion
Emne: Re: [asterisk-users] Error trying to add context: Context 'internal'
tries to include nonexistent context 'nighttime|12:30-8:00|mon-fri|*|*'
On Fri, Jun 18, 2010 at 7:44 AM, Aksel Celasun
mailto:ak...@abacus-it.no>> wrote:
Minor edit on the include =>
Minor edit on the include => nighttime|12:30-8:00|mon-fri|*|*
Correct now.
Fra: Aksel Celasun
Sendt: 18. juni 2010 14:30
Til: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Emne: Error trying to add context: Context 'internal' tries to include
nonexistent contex
TEN});
[incoming]
exten => s,1,Noop();
exten => s,n,Verbose(Call ${EXTEN});
exten => s,n,Dial(SIP/501);
exten => s,n,Hangup();
[macro-dial-trunk-sip]
exten => s,1,Noop(${ARG1},${CALLERID(num)})
exten => s,n,Set(CALLERID(num)=67209600)
exten => s,n,Dial(SIP/phonect_01/${ARG1})
exte
iling List - Non-Commercial Discussion
Emne: Re: [asterisk-users] Automatic attendant - Error in CLI.
Aksel Celasun wrote:
>
> This should be:
> exten => 501,n(LoopEnd),EndWhile
>
> I don't understand, i do have the same thing you wrote above.
>
The difference betwe
> Extensions.conf
> [mainmenu]
> exten => 501,1,Answer
> exten => 501,n,Wait(2)
> exten => 501,n,Playback(velkommen_abacus)
> exten => 501,n,Set(Loop=0)
> exten => 501,n,While($[${Loop} < 3])
> exten => 501,n,Background(tast123vent_)
> exten => 501,n,WaitExten(5)
> exten => 501,n,Set(Loop=$[${Loo
[Jun 18 10:38:16] WARNING[1692]: pbx.c:3680 pbx_extension_helper: No
application '' for extension (phones, 501, 9)
== Spawn extension (phones, 501, 9) exited non-zero on 'SIP/301-0248'
asterisk*CLI>
sip.conf regarding sip 501
[501]
type=friend
secret=XX
host=dyn
stack
extensions.conf snipped.
exten => 501,1,Answer
exten => 501,n,Set(Timeout(5)=timeout)
exten => 501,n,Set(Timeout(30)=response)
exten => 501,n,Background(velkommen_abacus&tast123vent_)
Med vennlig hilsen
Abacus IT AS
- din Visma Software Partner
Tor Aksel Celas
-
extensions.conf snipped
;exten 301
exten => 4767209611,1,NoOp();
exten => 4767209611,n,Verbose(Callerid num ${CALLERID(num)});
exten => 4767209611,n,Dial(SIP/301,5);
exten => 4767209600,n,Queue(teknisk
t; 4767209611,1,NoOp();
exten => 4767209611,n,Verbose(Callerid num ${CALLERID(num)});
exten => 4767209611,n,Dial(SIP/301,5);
exten => 4767209600,n,Queue(teknisk);
exten => 4767209611,n,Voicemail(301); ;Added 06.Mai.10-Aksel
Could someone please help m
mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Aksel Celasun
Sent: Monday, May 31, 2010 15:27
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Reloading queue members (realtime DB)
Hello there.
Have you tried "reload"
Hello there.
Have you tried "reload" in CLI?
Greeting
Aksel
Fra: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] På vegne av Mike
Sendt: 31. mai 2010 21:00
Til: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Emne: [asterisk-users] Reloading
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