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
om: mr...@imminc.com
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] The call is established but without exchanged
> voice packets
>
> Asmaa,
>
> You're getting ahead of yourself. How do you expect audio to work if
> your firewall/NAT settings
/wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
>
>
> Thanks.
>
> --
> Date: Fri, 20 Sep 2013 16:05:35 +0200
> From: asghar...@gmail.com
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] The call is established but without
> exch
splay/AST/SIP+Retransmissions
Thanks.
Date: Fri, 20 Sep 2013 16:05:35 +0200
From: asghar...@gmail.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] The call is established but without exchanged
voice packets
Hello,If Asterisk version is > 1.6 use nat=force_rport,comedia
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 h
exten => 8001,2,Hangup()
>
> exten => 8002,1,VoicemailMain(7002@main)
> exten => 8002,2,Hangup()
>
> --------------
> Date: Fri, 20 Sep 2013 16:25:42 +0200
> From: asghar...@gmail.com
> To: asterisk-users@lists.digium.com
> Subject: Re: [as
mail.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] The call is established but without exchanged
voice packets
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 wrote:
Hello,
I a
the
> bleeding choir invisible!! THIS is an EX-CANARY. (Reducing priority)
>
>
> Thanks.
>
> ------------------
> Date: Thu, 19 Sep 2013 13:14:59 +0500
> From: msalman...@gmail.com
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] The c
S is an
EX-CANARY. (Reducing priority)
Thanks.
Date: Thu, 19 Sep 2013 13:14:59 +0500
From: msalman...@gmail.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] The call is established but without exchanged
voice packets
Choose suitable NAT settings from sip.conf
turn direct med
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
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 wrote:
> 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
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
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