Displaying 20 results from an estimated 500 matches similar to: "no ring tone"
2010 Feb 25
1
Asterisk 1.6.0.17 PBX with two interfaces does not routes RTP packets - SIP Conf Problem likely
Hi,
I am try to configure Asterisk as PBX system with two interfaces as
shown below. One interface pointing to the local subnet with a SIP phone
and another interface pointing to the external ISP SIP Sever.
SJPhone(X.X.141.32)<--------->(Y.Y.47.149)local-intf-|Asterisk|external-
intf(Z.Z.247.106)<-------->(w.w.158.26)ISP-SIP-Server----OutsideWorld
I am able to setup a call from the
2009 Aug 05
0
Asterisk with gizmo5 and google voice only takes one call at a time.
my problem is this. I have google forward the call to gizmo5. I have this line in my sip file :
register => user:password at proxy01.sipphone.com
I believe this lines connects asterisk with gizmo5 so when it gets a call from Google, asterisk will answer it?
At the end of my sip file i have this
[Calls-From-Gizmo-Network]
type=user
context=demo
disallow=all
allow=ulaw
allow=ilbc
allow=gsm
2007 Nov 27
10
Asterisk behind a PIX firewall?
Is there anything special that anyone here has had to do to get an Aastra
phone (on the Internet) to talk to Asterisk behind a PIX firewall?
Ports 10000-20000 UDP are open on the PIX and forwarding to the Asterisk
server. The Asterisk server's RTP.CONF is set to use 10000-20000. The
phone registers, and will place AND receive calls, however, no audio is
passed. The phone is an Aastra
2020 Sep 21
2
Asterisk Drop call
Hello
I have an asterisk 16.2.1 on an ubuntu on AWS, which is experiencing a
drop in call. It does not have a certain time, it is random. The audio
is flowing normally and the call is dropped.
Has anyone ever experienced this?
My settings changed below:
allowoverlap = no
udpbindaddr = 0.0.0.0
tcpenable = no
tcpbindaddr = 0.0.0.0
transport = udp, ws, wss
srvlookup = yes
directmedia = no
2020 Sep 22
3
Asterisk Drop call
Hello.
Thanks for the reply.
Yes. In the traffic analyzed, the BYE is sent by the originator of the
call, but there is no "human" hangup, but the asterisk one.
BYE is sent, received and confirmed.
I don't know how I could investigate the reason for this BYE.
Em 21/09/2020 17:12, Dovid Bender escreveu:
> Is there anything in the Asterisk logs? Which side sends the BYE? Were
2011 Mar 19
1
Getting No Antenna bar when behind a NAT
My Asterisk server is behind a NAT and I have set:
----------------------------------------------------------------------------
externhost="my.server.address"
externrefresh=180
localnet=192.168.0.0/255.255.0.0
localnet=10.0.0.0/255.0.0.0
localnet=172.16.0.0/12
nat=yes
---------------------------------------------------------------------------
in [general] section of sip.conf.
I can
2014 Apr 16
1
Connecting 2 asterisks, one with PJSIP and other SIP returning 401
It's my first post here, so I'll cut to the chase
I have 2 Asterisk servers and want to connect them using sip on one and
pjsip on the other one. One is running at home and another at a VPS. The
first one will be the client (with dynamic ip) and the 2nd the server.
The client uses sip and the server pjsip.
This is the client's sip.conf
[general]
context = default
allowguest = no
2009 Aug 07
5
Asterisk in VMWare, how does it perform and what is the limit?
Hi,
I'm coming up with ideas about building a cluster of asterisk servers,
and am exploring the virtualization option.
I'm curious to know some real-world data about how many extensions a
VMWare install on good hardware could support.
I've seen stories about how the hypervisor timeslicing can wreak havoc
on call quality at some point.
Is this really the case? If so, what's a
2006 May 22
1
Asterisk on Proxy
Good Day All
I recently implemnetd asterisk outside our LAN (external network).It works well in a NAT settings.
But on external network with PROXY setting ASTERISK DID NOT WORK.
My question are
1 Can ASTERISK work in a PROXY setting .
2 If it can work how can i implement it .
Expecting your reply
Thanks
Paul
---------------------------------
Yahoo! Messenger
2009 Aug 09
6
CyberPower OR2200
i own a CyberPower OR2200LCDRM2U and was wondering if any nut drivers
support this ups. none i have tried have worked although usbhid-ups did
tell me that the productid=0601 was not supported yet. i see in the
compatability list on the nut website that the PR2200 is supported with
the powerpanel driver but the OR2200 is not listed. i am interested in
what i am doing wrong or if this is a support
2012 Feb 01
2
Getting one way audio even NAT is configured
Hi all,
I'm getting one way audio when calling over the SIP trunk i.e. end device B
(remote end of SIP trunk) can hear device A (softphone registered with
Asterisk) but device A can't hear device B. Even though I configured same
NAT configurations on other servers and they are working good. The NAT
configuration is listed below;
localnet=130.0.0.0/130.0.0.0
externhost=12.131.12.13
2005 Aug 16
1
problems with eyebeam - video phone
I am trying to connect two Xten eyeBeam Video Phone
No problems in voice connecting.
I tryed to modify my sip.conf
[general]
language=it
videosupport=yes
; enable Asterisk video support
port = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
disallow=all
allow=h263
allow=gsm
allow=ulaw
allow=alaw
; H.263 is our video codec
;
2009 Aug 10
6
"context" does not work
Hello,
i have a problem with the context parameter in the sip.conf. i'm using
a german sip provider (sipgate.de) and everything worked fine in
asterisk 1.4, but on 1.6.1 i got the following error message:
NOTICE[3071]: chan_sip.c:18160 handle_request_invite: Call from '' to
extension '8001187e0' rejected because extension not found.
sip.conf:
register =>
2015 Jun 07
3
Curious problem with NAT
Hi list!
Since the internal calls work as expected and I can register my Asterisk on
an external provider, I'd like to add a new feature and allow my mobile phone
to connect to my Asterisk and manage calls.
Well, first of all, my Asterisk is NOT direct on Internet available, but
behind a NAT.
So I configured my sip.conf:
localnet=192.168.200.0/24
externhost=myhost.noip.com
externrefresh=180
2006 Nov 30
2
Force re-read of sip.conf
I have an asterisk server with a dynamic public IP address. Once the IP
changes, remote clients suddenly have one-way audio again.
I can resolve the problem with a restart, but am thinking have adding a cron
command which does this every night. Will a "reload" cause asterisk to
respect the new IP address specified in sip.conf? Or do I have to restart?
Thanks,
MD
--------------
2007 Oct 01
1
SIP trought Firewall
Hi to everyone!
I have succerfully instaled my new Asterisk 1.4 on my debian etch.
I have my users in sip.conf like this:
[200]
type=peer
host=dynamic
context=home
secret=200
callerid= 200
dtmfmode=rfc2833
nat=yes
mailbox=200 at home
disallow=all
allow=ulaw
I can make calls in my LAN but i can?t ear comunications with another client
trought Internet.
My Asterisk is in my LAN and i not have a
2009 Apr 13
10
Asterisk-beginner : cannot make phonecalls using Asterisk
Hi there,
this is the first time that I'm building an Asterisk-server.
I have compiled Asterisk together with Zaptel on an CentOS 5.3-system.
Zaptel is for later, when configuring the POTS-line. Now first internal
communication with SIP.
Thought it would go easier...
I have 2 Grandstream IP-phones : BT-201 and GXP-1200.
These are my settings :
sip.conf :
[root at asterisk asterisk]# cat
2011 Apr 18
2
Registrations stops after 403 FORBIDDEN
Hello list,
I have in sip.conf :
/maxexpiry=60 ; Maximum allowed time of incoming
registrations
; and subscriptions (seconds)
minexpiry=60 ; Minimum length of
registrations/subscriptions (default 60)
defaultexpiry=120 ; Default length of incoming/outgoing
registration
;-----------------------------------------
2007 May 21
3
Aastra MWI
I need to setup MWI on a few Aastra 9112's. I've tried doing so in the
web interface by setting "Explicit MWI Subscription" to true, but no
lights, no stutter tone.
Firmware: 1.4.0.1048
Thanks!
--
Warm Regards,
Lee
2009 Jul 09
0
q: port forwarding or NAT
hi,
making may way through all this...internal sip registration works,(cant call
yet but anyhow)...
the asterisk box is obvisoulsy behind a router. im not 100% sure if i should
go with port forwarding or NAT and if a or b, what additional setup is
actually correct?
sip_nat.conf # this is when i got the NAT -route, right?
#gets all the dyndns-stuff
#externip = home.mydomain.com (Enter your