Hello,
I have set the direct media to be off, but still doesn't work. I am not sure
about NAT configuration!
SIP.conf, [general]
Hello,
If Asterisk version is 1.6 use nat=force_rport,comedia
On Fri, Sep 20, 2013 at 4:01 PM, Asmaa Ahmed asabatg...@hotmail.com wrote:
Hello,
I have set the direct media to be off, but still doesn't work. I am not
sure about NAT configuration!
SIP.conf, [general] section
Hello,
I have Asterisk 1.8.10.1Moving to nat=force_rport,comedia hasn't solved the
problem. Still having the same error!
I am not sure if this is related to the problem here, but I was trying to test
my voicemail and got this error No audio available).[Sep 20 14:05:41]
WARNING[11424]:
Hello,
i think your logic is wrong please explain me what are you trying to do?
[internal]
exten = 7002,1,Answer()
exten = 7002,n,Playback(vm-nobodyavail)
exten = 7002,n,Hangup()
exten = 7001,1,Dial(SIP/7001,60)
exten = 7001,n,Hangup()
try this dial 7002 and you should listen vm-nobodyavail or
Asmaa,
You're getting ahead of yourself. How do you expect audio to work if
your firewall/NAT settings aren't even configured correctly to
establish SIP sessions?
Go back and read the message that I sent yesterday. Fix the SIP
three-way handshake problem. That is step 1 and you'll know you
Hello,
Here is my extension context,
[internal]exten = 7001,1,Answer()exten = 7001,2,Dial(SIP/7001,60)exten =
7001,3,Playback(vm-nobodyavail)exten = 7001,4,VoiceMail(7001@main) ;forward to
voicemail mailboxexten = 7001,5,Hangup()
exten = 7002,1,Answer()exten = 7002,2,Dial(SIP/7002,60)exten =
Hello,
paste you extension context.
On Fri, Sep 20, 2013 at 4:21 PM, Asmaa Ahmed asabatg...@hotmail.com wrote:
Hello,
I have Asterisk 1.8.10.1
Moving to nat=force_rport,comedia hasn't solved the problem. Still having
the same error!
I am not sure if this is related to the problem here,
Hi Matthew,
Indeed I missed your previous message!After changing the externip, it worked
successfully... The sip session is established with the complete three-way
handshake, and the voice packet is exchanged with no problem!
Many thanks.
Date: Fri, 20 Sep 2013 10:01:52 -0500
From:
Asmaa Ahmed wrote:
Indeed I missed your previous message!
After changing the externip, it worked successfully... The sip
session is established with the complete three-way handshake, and
the voice packet is exchanged with no problem!
Many thanks.
Asmaa,
That's great news!! I guess
Hello,
I am trying to make my first call on Asterisk to succeed. I have Asterisk
1.8.10.1 running on Ubuntu machine.The configuration is quite simple just for
my first test, Trying to have a call between two X-lite sipphone. The
subscribers succeeded to register and the call is established, but
Choose suitable NAT settings from sip.conf
turn direct media in sip.conf or per peer off
On Thu, Sep 19, 2013 at 12:54 PM, Asmaa Ahmed asabatg...@hotmail.comwrote:
Hello,
I am trying to make my first call on Asterisk to succeed. I have Asterisk
1.8.10.1 running on Ubuntu machine.
The
Asmaa Ahmed wrote:
I am trying to make my first call on Asterisk to succeed. I have
Asterisk 1.8.10.1 running on Ubuntu machine.
The configuration is quite simple just for my first test, Trying to
have a call between two X-lite sipphone. The subscribers succeeded
to register and the
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