Hello,
can someone please tell me if this is possible and how ?
Kind regards,
Jonas.
On 04/24/2012 12:59 PM, Jonas Kellens wrote:
Hello,
is there a way to put a certain SIP peer on state busy ?
I know you can do this by pressing DND on your IP-phone, but can
this state also be set in
Check the command Busy() of the dialplan, it return the busy state at the
calling party.
Leandro
2012/4/26 Jonas Kellens jonas.kell...@telenet.be
**
Hello,
can someone please tell me if this is possible and how ?
Kind regards,
Jonas.
On 04/24/2012 12:59 PM, Jonas Kellens wrote:
Thank you for your answer but this is not what I'm looking for.
I want to place the state of a SIP peer 'busy'.
So that every time the peer is dialed with the Dial()-command or with
Call Queues, there is no INVITE send to this peer.
With other PBX-systems you can send some code ( for example
Perfect that's work ;=)
very thanks
Le 25 avril 2012 10:19, Olivier CALVANO o.calv...@gmail.com a écrit :
Ok thanks i test.
I put match_auth_username=yes on the two server ?
And for insecure, into the realtime database ? or into sip.conf of the
second server ?
best regards
olivier
I've found this bit of information, maybe it'll help:
http://www.voip-info.org/wiki/view/PBX+Do+Not+Disturb
On Tue, Apr 24, 2012 at 12:59 PM, Jonas Kellens jonas.kell...@telenet.bewrote:
**
Hello,
is there a way to put a certain SIP peer on state busy ?
I know you can do this by pressing
Anyknow know this problems ?
I read on the net that it's a possible network problems, but i don't think
because it's a VMWare server and in the same server i have other
asterisk without this problems.
best regards
Olivier
Le 25 avril 2012 09:35, Olivier CALVANO o.calv...@gmail.com a écrit
Usually its a firewall issue, or at least it has been for me
Its saying it can't form sip packets, and that will be because something isn't
letting it,
Cheers Duncan
On 26/04/2012, at 8:15 PM, Olivier CALVANO wrote:
Anyknow know this problems ?
I read on the net that it's a possible
Thanks for the link.
As I understand, if the SIP peer is member of a queue (agent) then the
example in your link is not usable ?
I'm looking for a global way of putting a certain SIP peer busy or
away or DoNotDisturb.
Jonas.
On 04/26/2012 10:14 AM, Michal Mrus(kovic( wrote:
I've found
As far as I know, there should be a way to use local channels in queues and
within the local channel would be the right place to check for the status
and either dial or don't.
You could also pause the queue member for all his queues, that would not be
scalable though.
On Thu, Apr 26, 2012 at
On 25 April 2012 18:05, Kevin P. Fleming kpflem...@digium.com wrote:
On 04/25/2012 11:54 AM, Steve Davies wrote:
A further question... It appears that for SIP endpoints, this facility
only updates RPID and PAI headers? I have found that there appear to
be 4 different SIP CID-update mechanisms
Hi
I'm looking for a SIP client for Mac OS X (I'm running Lion) that has
video support. I've tried Linphone but for the life of me I can't
get it to add a sip account (the apply button is always grayed
out) :-( Can anyone recommend other SIP clients that have video
Support for Mac OS X?
I'm using Bria,but X-Lite from counter path
I have good result with these programs under Lion
On 26 Apr 2012, at 12:05 PM, Alex Balashov wrote:
Have you looked into Blink?
On 04/26/2012 05:41 AM, Paolo Supino wrote:
Hi
I'm looking for a SIP client for Mac OS X (I'm running Lion) that
Am 26.04.12 13:23, schrieb Arjan Kroon | Mobillion:
I'm using Bria,but X-Lite from counter path
I have good result with these programs under Lion
I had very good results using jitsi for video calls. maybe its also
worth a look
best regards
--
Hi
No firewall on the server
Other idea ?? Hihi
Olivier
Le jeudi 26 avril 2012, Duncan Turnbull a écrit :
Usually its a firewall issue, or at least it has been for me
Its saying it can't form sip packets, and that will be because something
isn't letting it,
Cheers Duncan
On
The only way you can do this is by enabling DND on the phone.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Thursday, April 26, 2012 4:27 AM
To: Asterisk Users Mailing List -
Jonas Kellens jonas.kell...@telenet.be writes:
I know you can do this by pressing DND on your IP-phone, but can this
state also be set in the dialplan ?
You cannot actually achieve this by pressing DND on your IP-phone. All
that will accomplish is that the phone answers all calls with busy,
Two hammer options you could use - #1 use the sip notify to periodically
reset the phone #2 set up a dialplan snippet to change the credentials of
the peer and do a sip reload so it is disabled. (of course you would need a
corresponding snippet to re-enable the phone).
-Original Message-
On 04/16/2012 05:21 AM, Niccolò Belli wrote:
I suspected it, but it didn't work at first. I fear I didn't
understand what the context refers to in Pickup(extension[@context]).
I will make an example: phone-100 wants to pick up a ringing phone-200
(call comes from my-sip-provider).
This is
Hello, what about: This feature means you can restart Asterisk after
a failure (or asterisk restart itself with safe_asterisk), and keep
existing calls up with only a few seconds of audio dropped. That
would be a feature! there is a other pbx that has this feature...
Anyone else would like to see
Il 26/04/2012 16:04, Mark Michelson ha scritto:
What is the strategy of the queue?
Ringall.
How are the queue members listed (i.e. are they SIP channels or
local channels)?
There is only one member listed: SIP/phone-200
My suspicion is that the queue is simultaneously
dialing local
this will be a wonderful feature, if be possible, with asterisk...
as i looked back to this, i think you will find this on Asterisk-SCF. but
if this be possible with Core-Asterisk! then this what i am looking back
for a long..
On Thu, Apr 26, 2012 at 6:59 PM, Kristijan Vrban
Hello,
We are toying with setting up a redundant data center for our hosted PBX
product, and plan to use the OpenVPN feature of our Yealink phones for
connectivity to each data center. The feature has been fantastic with
the first data center, allowing us to bypass all SIP NAT issues entirely
Hi Stefan
A short while after I replied about blink I found jitsi... Does what
I need :-)
TIA
Paolo
On Thu, Apr 26, 2012 at 1:30 PM, Stefan Schmidt s...@sil.at wrote:
Am 26.04.12 13:23, schrieb Arjan Kroon | Mobillion:
I'm using Bria,but X-Lite from counter path
I have good result with
Hello,
I have a doubt (basic i guess, but not for me). I have an escenario where
customer site has Asterisk PBX behind Nat/firewall with private IP address
and sone phones also; BUT there are some other phones on different sites
and of course behind its nat/firewalls; with IAX i have no problem,
On Thu, Apr 26, 2012 at 9:54 AM, Danny Dias ing.diasda...@gmail.com wrote:
I have a doubt (basic i guess, but not for me). I have an escenario where
customer site has Asterisk PBX behind Nat/firewall with private IP address
and sone phones also; BUT there are some other phones on different
I cant put public ip adress to the asterisk server.
The main problem i see is with the sip headers (contact, sdp ip and ports,
etc)...with the result of one way audio or no registrstion at all. Does an
STUN server should works?
AGAIN: phones in one site behind nat and PBX in another site also
On 12-04-25 03:35 AM, Olivier CALVANO wrote:
Hi
i have a lot of error in the CLI of one of my Asterisk:
[Apr 25 09:30:46] WARNING[10542]: chan_sip.c:2901 __sip_xmit: sip_xmit
of 0x8ef2130 (len 867) to 172.16.250.34:5060 returned -1: Operation
not permitted
[Apr 25 09:30:46] WARNING[10542]:
On Thu, Apr 26, 2012 at 10:47 AM, Danny Dias ing.diasda...@gmail.comwrote:
I cant put public ip adress to the asterisk server.
The main problem i see is with the sip headers (contact, sdp ip and ports,
etc)...with the result of one way audio or no registrstion at all. Does an
STUN server
I'm using Asterisk 1.8.6.0 on my router talking to my ISP's Taqua 7000 (?)
switch.
I'm using a config that looks like:
[sip_proxy-out]
type=peer
authuser=208
remotesecret=xyzzy
qualify=100
host=n.n.n.n
call-limit=5
nat=no
; sendrpid=yes
insecure=no
But the Taqua responds to outbound
Does not work for me!
El 26/04/2012 20:14, Carlos Alvarez car...@televolve.com escribió:
On Thu, Apr 26, 2012 at 10:47 AM, Danny Dias ing.diasda...@gmail.comwrote:
I cant put public ip adress to the asterisk server.
The main problem i see is with the sip headers (contact, sdp ip and
On Thu, Apr 26, 2012 at 2:07 PM, Danny Dias ing.diasda...@gmail.com wrote:
Does not work for me!
What router(s) are in use?
Did you disable any SIP/VoIP helpers or ALGs they may have?
--
Carlos Alvarez
TelEvolve
602-889-3003
--
Well you have to tell asterisk what's the external ip of the nat else its never
gone work
Look at externip and localnet
-Original Message-
From: Carlos Alvarez car...@televolve.com
Sender: asterisk-users-boun...@lists.digium.com
Date: Thu, 26 Apr 2012 14:15:39
To: Asterisk Users Mailing
Hello,
I have an OpenVox A400E02 (2FXO) in a box running Debian 6.0.2 running
Asterisk 1.8.6.0. I have to POTS line on it from Verizon in Virginia,
USA. Whenever I place a call to one of the two lines I get a red alam
and then it clears and repeats this till I hang up. There is no caller
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