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
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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