Displaying 20 results from an estimated 500 matches similar to: "Registration timed out - for created users"
2007 May 06
2
Were i make mistake
I've found some manuals and tried this to do :
Sip.conf
[test]
type=friend
username=test1
secret=test1
host=192.168.1.238
context=tutorial
fromuser=SIP Phone
callerid=101
nat=no
canreinvite=yes
dtfmode=info
disallow=all
allow=ulaw
[test]
type=friend
username=test
secret=test
host=192.168.1.240
context=tutorial
callerid=100
nat=no
canreinvite=yes
dtfmode=info
2009 Jan 11
2
sip peer permit/deny - Need some explanation
Hi all,
I tested with few Asterisk versions from 1.4.18 to 1.4.21, same result.
Here is the problem: I have a peer -which is peer AND user- setted up
like this
[MyPeer]
;
type=peer
host=xxx.xxx.xxx.139
deny=0.0.0.0/0.0.0.0
permit=xxx.xxx.xxx.136/255.255.255.248 ;IP address from range 138 to 142
permit=yyy.yyy.yyy.yyy/255.255.255.255
context=from-MyPeer
dtfmode=auto
disallow=all
allow=ulaw,alaw
2007 Jun 25
0
four ringing and hangup with error
Dear All
I have this setup
[asterisk]----[mediant2000]-------E1 Trunk----------[Avaya]
When i call from avaya to asterisk i got long ringing tone then hangup but when i call from asterisk to avaya i got 4 ringback and then hangup with this error
*CLI> Jun 26 01:26:08 NOTICE[5555]: chan_local.c:523 local_alloc: No such extension/context 1022 at mysip
2013 May 12
2
Integrate Astreisk with SIP interface
Hi
Once I installed astrisk , how do we connect with SIP interface ?
Can somebody guide me how to integrate SIP interface with asterisk ? I want to use Astrisk just for IVR purpose.
Thank you
Luke
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2013 Jun 01
1
Most suitable version for Production ENV
Hi?
As I seen on the Asterisk web site , there is packages called ,?
AsteriskLatest Version - 11.4.0
asterisk-11-current.tar.gz?and?
asterisk-1.8-current.tar.gz
May I now which one is the most suitable for a production environment ?
Thanks in advance
Luke
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2006 Jan 06
2
Budge Tone-100 as a Ext in the LAN
HI ,
I installed asterisk in fedora core 3 machine perfectly. and i have 10 units of GrandStream IP phone ( Budge Tone-100 ) . I wanted to know how can i use it as extentions in my LAN ? Asterisk PBX alredy there. I didn't try to do any configurations of any files .
What are the configurations has to be made with asterisk ?
Thanx in advance,
Luke.
Send instant messages
2010 Apr 26
0
DTMF from SIP phone to FXS/FXO
Hello,
I am having trouble passing DTMF digits from a Polycom 330 SIP phone to my FXS/FXO lines. I am running Asterisk 1.4.21.1
In sip.conf I configured dtmfmode=inband. RTP traffic (voice) goes perfectly from SIP to FXS, but in the SIP phone I only hear a continuous noise. However, when I press any digit in the pone (FXS) I hear the DTMF tone fine in the SIP phone (the noise goes away for as
2003 Nov 28
0
Can't seem to connect/call fwd network Help!
I have tried everything and still can't place / receive calls from the fwd network. At one point today I was able to call my test machine on the fwd network, I'd answer the call on the test machine (which stated Call Connected), but then the computer I was calling from, through the Asterisk server would give me a 403 Error. I am using sjphone software. I am able to call various
2005 Aug 25
0
Internal FXS to SIP problem
I've just setup a new asterisk box (cvs HEAD) with a digium tdm411 and
a couple computers with eyebeam. I have one small. I cannot call the
eyebeam clients from the phone connected the fxs port. I can call the
phone from the eyebeem clients. And, I get both the fxs phone and
eyebeam clients to ring when a call comes in through the fxo port.
I have been trying to get this straightened out
2006 Apr 19
1
Callerid matching in extensions.conf
I'm using Asterisk 1.2.7.1. Has the callerid number matching functionality in extensions.conf changed recently?
exten => 5555,1,NoOp(${CALLERID})
hestia*CLI>
-- Executing NoOp("SIP/2944093-d24d", ""Cletus the Slaw Jawed Yokel" <2944093>") in new stack
== Auto fallthrough, channel 'SIP/2944093-d24d' status is 'UNKNOWN'
This
2007 Apr 03
1
Hints not working using SVN-branch-1.4-r59289
Group
I'm having trouble getting hints to work correctly using
SVN-branch-1.4-r59289
I have hints working on several other systems but I must be missing
something this time around.
VoIPGW*CLI> show hints
-= Registered Asterisk Dial Plan Hints =-
30@default :
State:Unavailable Watchers 3
29@default :
2020 Jun 05
2
pjsip subscribecontext support
Hello,
I would like to ask about current state of subscribecontext in pjsip.
I found out some 6 years old discussion on that without any plans to
implement it in the future.
I have phones in different contexts. I suspect, when I use its context
to subscribe, they will not see phones from the different contexts. Am
I right?
Marek
2020 Jun 05
0
pjsip subscribecontext support
On Fri, Jun 5, 2020 at 6:02 AM Marek Greško <mgresko8 at gmail.com> wrote:
> Hello,
>
> I would like to ask about current state of subscribecontext in pjsip.
> I found out some 6 years old discussion on that without any plans to
> implement it in the future.
>
> I have phones in different contexts. I suspect, when I use its context
> to subscribe, they will not see
2006 Feb 15
2
Hint priority
Hi All
Has anyone managed to get the hint priority with Swissvoice IP10S phones
working?
I have 2 phones: a Snom 360, setup as the reception phone on extension
11, and a Swissvoice IP10S on extension 12.
When calling each other (tested both ways) I can only ever see the Snom
360 in the Active State from 'show hints'. The Swissvoice stubbornly
remains in the Idle State when on a call!
2005 Jan 04
1
Re: Polycom Buddy Feature
I'm still trying to work this out.
I've got this in my sip.conf
[1003polycom]
type=peer
secret=abc123
host=dynamic
defaultip=192.168.1.215
context=default
mailbox=1003
subscribecontext=phonestatus
[1004polycom]
type=peer
secret=abc123
host=dynamic
defaultip=192.168.1.214
context=default
mailbox=1004
subscribecontext=phonestatus
And this in my extensions.conf
[phonestatus]
exten =>
2008 Nov 04
1
users.conf and hints
Is there a way to override sip peers defined in users.conf with respect to their context and hints?
Every extension I have defined in users.conf always gets an @default for the hint priority. Below are asterisk outputs and users.conf entries. In peer 1203 I've set a subscribecontext, which is completely ignored.
Thanks for any help.
nurscarepbx*CLI> core show version
Asterisk 1.4.22
2006 Dec 17
5
BLF on GXP2000
I am trying to set up the BLF on a GXP2000.
Currently what I have is
extensions.conf:
[globals]
polycom430=SIP/101
[internal]
exten => 101,1,Macro(voicemail,${polycom430})
[macro-voicemail]
exten => s,1,Dial(${ARG1},10,tT)
exten => s,2,VoiceMail(u${MACRO_EXTEN}@default )
exten => s,102,VoiceMail(b${MACRO_EXTEN}@default)
[ext-local-custom]
exten => 101,hint,${polycom430}
2015 Dec 30
2
Signaling ringing on other extension
Ishfaq Malik <ish at pack-net.co.uk> schrieb:
Hi Ishfaq
> Look into Busy Lamp Field/Presence
>
> Here's a starting point:
>
> http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-DeviceStates-SECT-1.html
Thanks a lot, but it does not seems to work...
Here my configuration:
sip.conf:
[general]
allowsubscribe=yes
subscribecontext = default
2015 Dec 30
2
Signaling ringing on other extension
Ishfaq Malik <ish at pack-net.co.uk> schrieb:
> The hints have to be in the same contexts in extensions.conf as defines in
> the sip.conf subscribecontext which can be set per peer.
Well, [anika_incoming] will be included in [default], of course...
But I tried to define anika_incoming in subscribecontext, too. No changes...
> Also, have you configured the phones as well?
What do
2006 Jan 06
2
controlling SIP subscriptions from SNOM phones
We recently deployed 10 SNOMs as part of a PBX hosted solution. We have one
phone setup as the receptionist phone, using hints to show busy office
lines. This all works as expected.
This is a new installation, and people are just starting to setup their
phones. For those of you not familiar with SNOM phones, there is a row of
keys on the right side of the phone which SNOM calls function keys. In