[VoIP] Help...............

[Andrew Green]

Try forwarding port 5060 on the Draytek router to its own internal ip, example:

External port: 5060 Internal port: 5060 IP: 10.0.0.1 (Or what ever is
the internal ip of the router)

-Andrew

On 10/15/06, Ian Jolly <ian at uax.org.uk> wrote:
> Hi All
>
> Help ! - we are still trying to set up an ATA working off my * tandem (as a
> 'remote' extension) to an FXS port on a Draytek 2600V router. We are using
> SIP. The other SIP peer 'SPA3102' is working OK on my internal network.
>
> We've got as far as me being able to ring the phone at the ATA  identified
> as 'jgriffiths'.
> My fixed IP is 81.174.170.48
> 'jgriffiths' fixed IP is 80.176.242
>
> However I get the following - I'm not shouting - it is just easier to see my
> comments and the cut and paste from the CLI> prompt.
>
> AT THE CLI> prompt - I GET A MESSAGE LIKE THIS - ANY CALL TO THE ATA IS CUT
> OFF -
>
> Oct 15 23:35:18 NOTICE[2490]: chan_sip.c:11328 sip_poke_noanswer: Peer
> 'jgriffiths' is now UNREACHABLE! Last qualify: 134
> Destroying call '791c82153ca0c38868892e5438d88da2 at 81.174.170.48'
>
> A MESSAGE LIKE THIS IS SENT EVERY FEW SECONDS WITH A DIFFERENT DIGIT BEFORE
> THE # SIGN-
>
> Retransmitting #4 (NAT) to 80.176.242.9:5060:
> OPTIONS sip:jgriffiths at 80.176.242.9 SIP/2.0
> Via: SIP/2.0/UDP 81.174.170.48:5060;branch=z9hG4bK3d3f9f80;rport
> From: "asterisk" <sip:asterisk at 81.174.170.48>;tag=as162e4695
> To: <sip:jgriffiths at 80.176.242.9>
> Contact: <sip:asterisk at 81.174.170.48>
> Call-ID: 2e7c5657365eb28d67a552756a95ba33 at 81.174.170.48
> CSeq: 102 OPTIONS
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Sun, 15 Oct 2006 22:45:23 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Content-Length: 0
>
> ---
> Destroying call '2e7c5657365eb28d67a552756a95ba33 at 81.174.170.48'
>
> EVENTUALLY A MESSAGE LIKE THIS IS RECEIVED
>
> Oct 15 23:35:56 NOTICE[2490]: chan_sip.c:9689 handle_response_peerpoke: Peer
> 'jgriffiths' is now REACHABLE! (83ms / 2000ms)
> Destroying call '071faaca10c99ee53e73b39c261e2472 at 81.174.170.48'
>
> THEN MORE -
>
> Retransmitting 14 (NAT) to 80.176.242.9:5060:
> OPTIONS sip:jgriffiths at 80.176.242.9 SIP/2.0
> Via: SIP/2.0/UDP 81.174.170.48:5060;branch=z9hG4bK3d3f9f80;rport
> From: "asterisk" <sip:asterisk at 81.174.170.48>;tag=as162e4695
> To: <sip:jgriffiths at 80.176.242.9>
> Contact: <sip:asterisk at 81.174.170.48>
> Call-ID: 2e7c5657365eb28d67a552756a95ba33 at 81.174.170.48
> CSeq: 102 OPTIONS
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Sun, 15 Oct 2006 22:45:23 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Content-Length: 0
>
> ---
> Destroying call '2e7c5657365eb28d67a552756a95ba33 at 81.174.170.48'
>
> UNTIL ANOTHER 'UNREACHABLE' MESSAGE IS RECEIVED AND THE PROCEURE REPEATS
> ITSELF
> EVENTUALLY I GET -
>
> Oct 15 23:49:52 WARNING[2490]: chan_sip.c:1208 retrans_pkt: Maximum retries
> exceeded on transmission 178ce1c47486afd264bdf26f56eb1c9e at 81.174.170.48 for
> seqno 103 (Non-critical Request)
> Destroying call '178ce1c47486afd264bdf26f56eb1c9e at 81.174.170.48'
>
>
> IF I DO A SIP SHOW PEERS WHILST  'JGRIFFITHS'  IS REACHABLE - I GET
>
> asterisk*CLI> sip show peers
> Name/username             Host                 Dyn     Nat     ACL     Port
> Status
> jgriffiths/jgriffiths             80.176.242.9     D         N
> 5060     OK (83 ms)
> SPA3102/SPA3102      92.168.1.12       D                             5060
> Unmonitored
> 2 sip peers [2 online , 0 offline]
> asterisk*CLI>
>
> IF I DO A SIP SHOW PEERS WHILST  'JGRIFFITHS' IS UNREACHABLE - I GET
>
> Oct 15 23:54:43 NOTICE[2490]: chan_sip.c:11328 sip_poke_noanswer: Peer
> 'jgriffiths' is now UNREACHABLE! Last qualify: 115
> Destroying call '6d81be4c1eeb3bcc72978a5720057279 at 81.174.170.48'
> asterisk*CLI> sip show peers
> Name/username         Host                     Dyn     Nat     ACL     Port
> Status
> jgriffiths/jgriffiths         80.176.242.9         D        N
> 5060     UNREACHABLE
> SPA3102/SPA3102 192.168.1.12          D                              5060
> Unmonitored
> 2 sip peers [1 online , 1 offline]
> asterisk*CLI>
>
> Anyone any ideas?
>
> Is it something to do with timing of the 'sip poke' ?  One ends timer out of
> sync with the other end?.
>
> We are only beginners at this game but learning fast!
>
>
> Ian Jolly
>
> +44 (0)352 85 26 (via 1929 GPO UAX5 exchange)
> CNET - the Heritage Telephone Network
>
>
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