[asterisk-users] Soft phone not registering

[Rakesh Sabharwal]

Hi Darrin

Thanks for your kind reply.

Your description is right, PC(Soft Phone) > ADSL Router > Internet > Asterisk box

Thanks for your suggestion on the security.

Please advise , I am specifically concerned about the port to which server reply after initial communication (random above 32000) Retransmitting #3 (NAT) to x.x.x.x:38155:

Thanks in advance

Rakesh



----- Original Message ----
From: Darrin Henshaw <darrin.asterisk at gmail.com>
To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com>
Sent: Friday, 16 October, 2009 21:31:47
Subject: Re: [asterisk-users] Soft phone not registering

First suggestion is if this Asterisk server is accessible from the
internet put a secret in the peer definition. What you have now is
wide open. Second thing is if I understand it you are going:

PC(Soft Phone) > ADSL Router > Internet > Asterisk box. Is that
correct? If not, can you descibe it better.

On Fri, Oct 16, 2009 at 7:56 AM, Rakesh Sabharwal
<sabharwal_rakesh at yahoo.co.uk> wrote:
>
> HI All,
>
> I have installed Asterisk 1.4.26.2 on a centOS box on a public IP and trying to connect from softphone behind ADSL router.
>
> The softphone is not able to register, we get some SIP messages on the server, which look like below.
>
> Please advise where could be the issue.
>
> Thnx
> Rakesh
>
> ---
> Retransmitting #3 (NAT) to x.x.x.x:38155:
> OPTIONS sip:test2 at 192.168.1.7:5060;rinstance=5b19b87f10954011;transport=UDP SIP/2.0
> Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK39432aec;rport
> From: "asterisk" <sip:asterisk at x.x.x.x>;tag=as7d8aae9d
> To: <sip:test2 at 192.168.1.7:5060;rinstance=5b19b87f10954011;transport=UDP>
> Contact: <sip:asterisk at 203.211.60.167>
> Call-ID: 3c92389c5e72d3e92fd8d20b70055d46 at x.x.x.x
> CSeq: 102 OPTIONS
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Fri, 16 Oct 2009 10:47:56 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
> Supported: replaces
> Content-Length: 0
>
>
> ---
> Retransmitting #4 (NAT) to x.x.x.x:38155:
> OPTIONS sip:test2 at 192.168.1.7:5060;rinstance=5b19b87f10954011;transport=UDP SIP/2.0
> Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK39432aec;rport
> From: "asterisk" <sip:asterisk at 203.211.60.167>;tag=as7d8aae9d
> To: <sip:test2 at 192.168.1.7:5060;rinstance=5b19b87f10954011;transport=UDP>
> Contact: <sip:asterisk at x.x.x.x>
> Call-ID: 3c92389c5e72d3e92fd8d20b70055d46 at x.x.x.x
> CSeq: 102 OPTIONS
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Fri, 16 Oct 2009 10:47:56 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
> Supported: replaces
> Content-Length: 0
>
> --------------------
>
> sip.conf ----
>
> [general]
> context = tutorial
> bindport = 5060
> bindaddr =0.0.0.0
> domain = x.x.x.x
> nat=yes
> disallow = all
> allow = alaw
> keeprtpalive = yes
> notifyringing = yes
> canreinvite = no
> type = peer
> realm = asterisk
> qualify = yes
>
> [test2]
> type = peer
> host = dynamic
> username = test2
> context = tutorial
> port = 5060
> notifyringing = yes
> nat = yes
> type = friend
> canreinvite = no
> realm = asterisk
> qualify = yes
> mailbox=888 at mb_tutorial
>
> ---------------
>
>
>
>
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