Displaying 20 results from an estimated 10000 matches similar to: "1-way audio"
2010 Jan 12
2
SIP Security
Hey guys,
I've been running asterisk on my server for some time now (currently
running Asterisk 1.6.2.0). I am having security issues with my SIP
accounts. Unauthorized people have been able to access the server (bots)
and they have been able to make calls (in today's case to Cuba).
Here's a copy (slightly modified) of my sip.conf:
[general]
context=default ; Default
2006 Apr 12
1
Where is the difference sip.conf - Real-time ?
I have two phones (111 and 112) on a LAN, and I have on a users site a
phone 333.
phone 111 uses sip.conf, while 112 uses real-time set-up.
111 can call 333 AND the audio is working
112 can call 333 but audio is just white noise.
333 can call 111 or 112 and audio is working.
The phones are identically set-up (just user name = phone number and
password are different)
sip.conf (for 111 - all
2005 Jul 29
0
ReInvite X Broadvoice
I've been wondering for a long time why my reinvite option is not working with
Broadvoice anymore. It happend during the time Broadvoice was having a lot of
issues, so I decided to wait.
Recently I decided to test the same sip.conf with another VSP (SIPphone) and it
worked fine! No issues on the reinvite.
Note: clients and server using ULAW (only), no NAT or firewalls, public ip address
and
2004 Jul 26
1
Nat...again....
This has probably been answered somewhere, but I'm stumped.
I have two Zap channels (FXS and FXO), both working fine. I
can call from Zap/1 to Zap/2 and reverse.
I've also configured SIP channels, both inside and outside of my
firewall. Inside can call outside, and outside can call inside.
Also, both inside and outside can make and receive calls to/from
Zap/1 & Zap/2.
What
2007 Nov 30
2
My AsteriskNo unable to registration
Dear The Expert,
I am very new with this, I have installed AsteriskNow, X-Lite as my
SoftPhone, I am using SPA-3102.
I have 3 extensions,
me at 250, 998 is my Linksys SPA-3102 and 999 for PSTN Line (see below)
My problem is, I am unable to call 998, I thought this is registration
problem, (because the Linksys screen info said Registration Failed)
Could any body please help?
Many thanks in
2008 Dec 18
1
[Fwd: Asterisk client for ekiga.net NAT problem]
I am experiencing a "606 not Acceptable" error trying to set up an
Asterisk server as an ekiga.net client. My server is behind a firewall
with NAT routing. I have googled this problem and read about Asterisk
feeding its local ip address to ekiga.net. That seems to be my
problem.
I tried putting stunaddr=stun.ekiga.net into the sip.conf file under
[ekiga]. I also tried
2003 Nov 04
0
Need Help with SIP/H323.
Hi list,
why I cannot hear voice when I call from a SIP telephone (Budgetone and others) to a H323 telephone (several models)?
could anybody please give any idea to solve this issue?
Please, let me know.
Thanks in Advance.
N.B.
The configuration for "extensions.conf", "sip.conf" and "h323.conf" files are:
***************************************
2011 Apr 11
1
Asterisk codec negotiation and canreinvite=no
Hi all,
I realise that asterisk's codec negotiation has been discussed in
the past multiple times. What I haven't been able to understand is
how asterisk decides which video codecs to advertise to the other
end when canreinvite=no in sip.conf and the initial caller
doesn't support video.
My tests are quite simple, I use an asterisk with 4 peers all on the
same LAN. My sip.conf
2004 Jan 19
4
CVS Changes (NAT-SIP)
I had been running an older patched CVS to get VOIP working with NAT and
everything had been running fine. I just built * on a new box with
CVS-01/18/04-12:19:25. And now I can get remote SIP users to register.
Has anything major changed...
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind to
externip = 69.132.68.17 ; Address
2011 Oct 24
0
device state of SIP device is stucked into NOT_INUSE, and cannto be reverted to unavailable
Hello
Have a setup of asterisk with realtime SIP devices.
Trying to organise monitoring of my SIP devices. Once device
registered, its state becomes NOT_INUSE (result of
DEVICE_STATE(SIP/device) function).
Simulating of device breakage - powerdown it.
Waiting for a while (minute or two), retrieving
DEVICE_STATE(SIP/device) again - no changes! Awaited UNAVAILABLE.
doing from CLI:
sip qualify peer
2009 Jan 30
2
Asterisk with Avaya
Hi !
I am trying to connect Asterisk with Avaya Definity.
I use this tutorial to do this http://cyril-constantin.blogspot.com/2008/04/howto-connect-avaya-to-asterisk.html
The comunication between avaya and asterisk is fine but without sound. I can call from Asterisk to Avaya and extension ring or Avaya to Asterisk and extension ring too but I cant hear anything
Example
Asterisk ---> Avaya
--
2004 Apr 04
3
SIP Registration Errors
Hi...I've got two Grandstream phones attached to my Asterisk on the same
subnet. The phones have fixed IP addresses. Asterisk is generated an error
for one of them only, even though both appear to be registered correctly.
The current state of the sip.conf is included below. Anyone know what is
going on here? Both appear to be working fine between each other and between
themselves in and
2007 Jan 28
0
Trouble outgoing VOIP Provider Calls
I have a weird problem....
Asterisk 1.4
E100P connected to a Panasonic TDA phone system
Here is what I get
SIP Ext -> Panasonic Extensions No Problems
Panasonic Ext -> SIP Extensions No Problems
SIP Ext -> VOIP Provider No Problems
Panasonic Ext -> VOIP Provider Errors
---------- Working SIP -> VOIP
-- Executing [903........@from-sip:1]
2006 Feb 25
2
sipgate.de question
Hi,
Anyone here using sipgate.de ?
It worked for months, but for a couple of days now I'm
unable to register with them.
My account is ok, because I can login to the website.
Asterisk keeps showing me:
Feb 25 23:50:18 NOTICE[5144]: chan_sip.c:5269 sip_reg_timeout: -- Registration for 'XXXXX@sipgate.de' timed out, trying again (Attempt #n)
I looked at the sip debug stuff, and all I
2005 May 12
0
Asterisk, SIP and NAT: Help needed!
I've been googling and talking with Libretel about my
setup and the fact that incoming calls to my asterisk
box through the Libretel number reach my box (I hear
the greeting being played) but then don't accept DTMF.
Here is a rough diagram of my setup:
Asterisk |
server | NAT <------------ Libretel
| router
|
Note that there are NO SIP
2006 Apr 20
0
Re: Asterisk-Users Digest, Vol 21, Issue 113
Hi List!!
Thanks for the colaboration, especially to Richard Cavanna who gave me the
necessary support.
I followed your indications and the comunication was better for the test
users. The warning indication is no jumping anymore and the voice is not
delayed. This is my sip.conf:
[general]
context=default
;allowguest=no
;realm=mydomain.tld
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
2007 Feb 04
1
Help - Received response: "Forbidden" from '"Unknown"
I have a weird problem....
Asterisk 1.4
E100P connected to a Panasonic TDA phone system
Here is what I get
SIP Ext -> Panasonic Ext No Problems
Panasonic Ext -> SIP Ext No Problems
SIP Ext -> VOIP Provider No Problems
Panasonic Ext -> VOIP Provider Errors
---------- Working SIP -> VOIP
-- Executing [903........@from-sip:1] Dial("SIP/610-097aee60",
2010 Sep 16
4
one way audio for xlite clients behind NAT
I am having a one way audio issue with xlite clients behind NAT. They can
connect to the server and make calls but no audio is heard on the other
end.
my sip conf
[general]
context=default
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
canreinvite=no[tomfmason]
type=friend
secret=secret
callerid="Thomas Johnson" <XXXX>
host=dynamic
nat=yes
canreinvite=no
disallow=all
allow=gsm
2007 Jul 30
0
asterisk 1.4.8 and google talk - no audio
Hi all,
Iam using asterik 1.4.8 and connected to google talk. When iam calling from
my google talk account to sip phone i can hear the voice (2 way). (this
happens only within the LAN).
when my friend tries to call my asterisk server (connects to the public ip)
using his googletalk client it comes to my sip phone but either party cant
hear a voice.
I have fully allowd both tcp,udp on my
2009 Jun 26
0
Problem loss 2 seconds audio when Packet2Packet bridging
I'm sorry, i send mail in asterisk-bug, but asterisk-users is better for
my problem
Hello,
During a call with canreinvite = no, at the beginning of the call I lose
2 seconds of audio.
is obvious when I call autoattendant.
schema:
SipPhone --> Centrex (asterisk 1.4.24.1) --> Voip1 (Asterisk 1.4.24.1)
--> Operator SIP
capture of voip1:
- Executing [0825387205 at