[Serusers] Missing something !!!!
g.billoudet at arwen-tech.fr
g.billoudet at arwen-tech.fr
Mon Nov 29 17:16:33 CET 2004
Hi,
In route[1], I read 10.0.0.73 instead of 10.0.0.13 (see rewritehostport...)
Does it solve your problem ?
Gwen
> Hi,
>
> I set up my ser+asterisk in order to make it scalable suggested.
>
> The amazing thing is that when lookup(location) failed, call is forward to
> asterisk as I asked (see diagnostic below) but when I got a client is not
> responding, calls are not forwarding to asterisk.
>
> Where do I miss something ?
>
> Thanks in advance
> ---------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------
> if (uri==myself) {
> if (method=="REGISTER") {
> save("location");
> break;
> };
>
> if (!lookup("location")) {
> sl_send_reply("404", "Not Found");
> rewritehostport("10.0.0.13:5070");
> t_relay_to_udp("10.0.0.13","5070");
> break;
> }; # THIS IS WORKING !!! IT'S FOR TEST PURPOSE
> };
>
> if (!t_relay()) {
> sl_reply_error();
> };
>
> if (method=="INVITE"){
> t_on_failure("1");
> t_relay();
> break;
> }
> }
>
> # THIS IS NOT WORKING AT ALL !!!
> route[1]{
> if(uri=~"^sip:72[0-9]{2}@*"){
> revert_uri();
> rewritehostport("10.0.0.73:5070");
> append_branch();
> t_relay_to_udp("10.0.0.13","5070");
> }
> }
>
> --------------------------------------------------- LOGS SHOW WHEN TRYING
> TO CALL USER 7200 WHO IS TURN OFF
> ----------------------------------------------------
>
> Retransmitting #3 (no NAT):
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 10.0.0.242;branch=z9hG4bK5abb.ca4466d3.0
> Via: SIP/2.0/UDP
> 10.0.0.155:5060;branch=z9hG4bK1A7CD2A548DE4C5A9ADC42D9117B3B5B
> Record-Route: <sip:7200 at 10.0.0.242;ftag=1412313924;lr=on>
> From: Jean <sip:7201 at 10.0.0.242>;tag=1412313924
> To: <sip:7200 at 10.0.0.242>;tag=as25d200ae
> Call-ID: 240C974F-FFF5-4EB7-B9D9-C8B69484018B at 10.0.0.155
> CSeq: 47883 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Contact: <sip:7200 at 10.0.0.13:5070>
> Content-Type: application/sdp
> Content-Length: 255
>
> v=0
> o=root 5586 5587 IN IP4 10.0.0.13
> s=session
> c=IN IP4 10.0.0.13
> t=0 0
> m=audio 10602 RTP/AVP 3 0 8 101
> a=rtpmap:3 GSM/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
>
> to 10.0.0.242:5060
> Retransmitting #4 (no NAT):
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 10.0.0.242;branch=z9hG4bK5abb.ca4466d3.0
> Via: SIP/2.0/UDP
> 10.0.0.155:5060;branch=z9hG4bK1A7CD2A548DE4C5A9ADC42D9117B3B5B
> Record-Route: <sip:7200 at 10.0.0.242;ftag=1412313924;lr=on>
> From: Jean <sip:7201 at 10.0.0.242>;tag=1412313924
> To: <sip:7200 at 10.0.0.242>;tag=as25d200ae
> Call-ID: 240C974F-FFF5-4EB7-B9D9-C8B69484018B at 10.0.0.155
> CSeq: 47883 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Contact: <sip:7200 at 10.0.0.13:5070>
> Content-Type: application/sdp
> Content-Length: 255
>
> v=0
> o=root 5586 5587 IN IP4 10.0.0.13
> s=session
> c=IN IP4 10.0.0.13
> t=0 0
> m=audio 10602 RTP/AVP 3 0 8 101
> a=rtpmap:3 GSM/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
>
> to 10.0.0.242:5060
> Retransmitting #5 (no NAT):
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 10.0.0.242;branch=z9hG4bK5abb.ca4466d3.0
> Via: SIP/2.0/UDP
> 10.0.0.155:5060;branch=z9hG4bK1A7CD2A548DE4C5A9ADC42D9117B3B5B
> Record-Route: <sip:7200 at 10.0.0.242;ftag=1412313924;lr=on>
> From: Jean <sip:7201 at 10.0.0.242>;tag=1412313924
> To: <sip:7200 at 10.0.0.242>;tag=as25d200ae
> Call-ID: 240C974F-FFF5-4EB7-B9D9-C8B69484018B at 10.0.0.155
> CSeq: 47883 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Contact: <sip:7200 at 10.0.0.13:5070>
> Content-Type: application/sdp
> Content-Length: 255
>
> v=0
> o=root 5586 5587 IN IP4 10.0.0.13
> s=session
> c=IN IP4 10.0.0.13
> t=0 0
> m=audio 10602 RTP/AVP 3 0 8 101
> a=rtpmap:3 GSM/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
>
> to 10.0.0.242:5060
> Nov 29 15:22:39 WARNING[31452080]: chan_sip.c:683 retrans_pkt: Maximum
> retries exceeded on call 240C974F-FFF5-4EB7-B9D9-C8B69484018B at 10.0.0.155
> for seqno 47883 (Non-critical Response)
> Nov 29 15:22:43 NOTICE[125635504]: res_musiconhold.c:306 monmp3thread:
> Request to schedule in the past?!?!
> -- Playing 'vm-password' (language 'en')
> -- Incorrect password '' for user '7201' (context = <any>)
> -- Playing 'vm-incorrect-mailbox' (language 'en')
> -- Playing 'vm-password' (language 'en')
> -- Incorrect password '' for user '7201' (context = <any>)
> -- Playing 'vm-incorrect-mailbox' (language 'en')
> -- Playing 'vm-password' (language 'en')
> -- Incorrect password '' for user '7201' (context = <any>)
> -- Playing 'vm-incorrect' (language 'en')
> -- Playing 'vm-goodbye' (language 'en')
> -- Executing Hangup("SIP/10.0.0.242-085da938", "") in new stack
> == Spawn extension (default, 7200, 4) exited non-zero on
> 'SIP/10.0.0.242-085da938'
> Destroying call '240C974F-FFF5-4EB7-B9D9-C8B69484018B at 10.0.0
>
>
>
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