sip show peer ovh * Name : ovh Secret : <Set> MD5Secret : <Not set> Context : entrant-ovh Subscr.Cont. : <Not set> Language : fr AMA flags : Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup : Pickupgroup : Mailbox : VM Extension : asterisk LastMsgsSent : 32767/65535 Call limit : 0 Dynamic : No Callerid : "" <> MaxCallBR : 384 kbps Expire : -1 Insecure : port,invite Nat : RFC3581 ACL : No T38 pt UDPTL : Yes CanReinvite : No PromiscRedir : No User=Phone : No Video Support: No Text Support : No Trust RPID : No Send RPID : No Subscriptions: Yes Overlap dial : Yes DTMFmode : auto Timer T1 : 500 Timer B : 32000 ToHost : sip.ovh.net Addr->IP : 91.121.129.17 Port 5060 Defaddr->IP : 0.0.0.0 Port 0 Transport : UDP Def. Username: 0033972112355 SIP Options : (none) Codecs : 0x100 (g729) Codec Order : (g729:20) Auto-Framing : No 100 on REG : No Status : UNREACHABLE Useragent : Reg. Contact : Qualify Freq : 60000 ms Sess-Timers : Accept Sess-Refresh : uas Sess-Expires : 1800 secs Min-Sess : 90 secs
--- Retransmitting #1 (no NAT) to 91.121.129.17:5060: OPTIONS sip:sip.ovh.net SIP/2.0 Via: SIP/2.0/UDP 172.20.1.1:5060;branch=z9hG4bK599f649d;rport Max-Forwards: 70 From: "asterisk" <sip:aster...@172.20.1.1>;tag=as1545fb99 To: <sip:sip.ovh.net> Contact: <sip:aster...@172.20.1.1> Call-ID: 578ac87b06eaa6526aa313e130be3...@172.20.1.1 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.0.8 Date: Wed, 08 Apr 2009 05:57:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 --- Retransmitting #6 (no NAT) to 91.121.129.17:5060: REGISTER sip:91.121.129.17 SIP/2.0 Via: SIP/2.0/UDP 172.20.1.1:5060;branch=z9hG4bK312a379b;rport Max-Forwards: 70 From: <sip:0033972112...@91.121.129.17>;tag=as16505dec To: <sip:0033972112...@91.121.129.17> Call-ID: 165ff552001c7f1e202e67200ae67...@172.25.3.51 CSeq: 1465 REGISTER User-Agent: Asterisk PBX 1.6.0.8 Authorization: Digest username="0033972112355", realm="sip.ovh.net", algorithm=MD5, uri="sip:91.121.129.17", nonce="0019c92d503f745637b43af4264a11db", response="04e848af655d00e03d032d9a1c2fae09", opaque="001934772ef6ed5" Expires: 120 Contact: <sip:0033972112...@172.20.1.1> Event: registration Content-Length: 0 --- Retransmitting #2 (no NAT) to 91.121.129.17:5060: OPTIONS sip:sip.ovh.net SIP/2.0 Via: SIP/2.0/UDP 172.20.1.1:5060;branch=z9hG4bK599f649d;rport Max-Forwards: 70 From: "asterisk" <sip:aster...@172.20.1.1>;tag=as1545fb99 To: <sip:sip.ovh.net> Contact: <sip:aster...@172.20.1.1> Call-ID: 578ac87b06eaa6526aa313e130be3...@172.20.1.1 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.0.8 Date: Wed, 08 Apr 2009 05:57:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 --- Retransmitting #3 (no NAT) to 91.121.129.17:5060: OPTIONS sip:sip.ovh.net SIP/2.0 Via: SIP/2.0/UDP 172.20.1.1:5060;branch=z9hG4bK599f649d;rport Max-Forwards: 70 From: "asterisk" <sip:aster...@172.20.1.1>;tag=as1545fb99 To: <sip:sip.ovh.net> Contact: <sip:aster...@172.20.1.1> Call-ID: 578ac87b06eaa6526aa313e130be3...@172.20.1.1 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.0.8 Date: Wed, 08 Apr 2009 05:57:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 --- Retransmitting #4 (no NAT) to 91.121.129.17:5060: OPTIONS sip:sip.ovh.net SIP/2.0 Via: SIP/2.0/UDP 172.20.1.1:5060;branch=z9hG4bK599f649d;rport Max-Forwards: 70 From: "asterisk" <sip:aster...@172.20.1.1>;tag=as1545fb99 To: <sip:sip.ovh.net> Contact: <sip:aster...@172.20.1.1> Call-ID: 578ac87b06eaa6526aa313e130be3...@172.20.1.1 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.0.8 Date: Wed, 08 Apr 2009 05:57:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 --- Really destroying SIP dialog '578ac87b06eaa6526aa313e130be3...@172.20.1.1' Method: OPTIONS [Apr 8 07:57:48] NOTICE[25949]: chan_sip.c:9490 sip_reg_timeout: -- Registration for '0033972112...@91.121.129.17' timed out, trying again (Attempt #1262) REGISTER 13 headers, 0 lines Reliably Transmitting (no NAT) to 91.121.129.17:5060: REGISTER sip:91.121.129.17 SIP/2.0 Via: SIP/2.0/UDP 172.20.1.1:5060;branch=z9hG4bK40d5f950;rport Max-Forwards: 70 From: <sip:0033972112...@91.121.129.17>;tag=as02687bc2 To: <sip:0033972112...@91.121.129.17> Call-ID: 165ff552001c7f1e202e67200ae67...@172.25.3.51 CSeq: 1466 REGISTER User-Agent: Asterisk PBX 1.6.0.8 Authorization: Digest username="0033972112355", realm="sip.ovh.net", algorithm=MD5, uri="sip:91.121.129.17", nonce="0019c92d503f745637b43af4264a11db", response="04e848af655d00e03d032d9a1c2fae09", opaque="001934772ef6ed5" Expires: 120 Contact: <sip:0033972112...@172.20.1.1> Event: registration Content-Length: 0 --- Really destroying SIP dialog '165ff552001c7f1e202e67200ae67...@172.25.3.51' Method: REGISTER Retransmitting #1 (no NAT) to 91.121.129.17:5060: REGISTER sip:91.121.129.17 SIP/2.0 Via: SIP/2.0/UDP 172.20.1.1:5060;branch=z9hG4bK40d5f950;rport Max-Forwards: 70 From: <sip:0033972112...@91.121.129.17>;tag=as02687bc2 To: <sip:0033972112...@91.121.129.17> Call-ID: 165ff552001c7f1e202e67200ae67...@172.25.3.51 CSeq: 1466 REGISTER User-Agent: Asterisk PBX 1.6.0.8 Authorization: Digest username="0033972112355", realm="sip.ovh.net", algorithm=MD5, uri="sip:91.121.129.17", nonce="0019c92d503f745637b43af4264a11db", response="04e848af655d00e03d032d9a1c2fae09", opaque="001934772ef6ed5" Expires: 120 Contact: <sip:0033972112...@172.20.1.1> Event: registration Content-Length: 0 --- Retransmitting #2 (no NAT) to 91.121.129.17:5060: REGISTER sip:91.121.129.17 SIP/2.0 Via: SIP/2.0/UDP 172.20.1.1:5060;branch=z9hG4bK40d5f950;rport Max-Forwards: 70 From: <sip:0033972112...@91.121.129.17>;tag=as02687bc2 To: <sip:0033972112...@91.121.129.17> Call-ID: 165ff552001c7f1e202e67200ae67...@172.25.3.51 CSeq: 1466 REGISTER User-Agent: Asterisk PBX 1.6.0.8 Authorization: Digest username="0033972112355", realm="sip.ovh.net", algorithm=MD5, uri="sip:91.121.129.17", nonce="0019c92d503f745637b43af4264a11db", response="04e848af655d00e03d032d9a1c2fae09", opaque="001934772ef6ed5" Expires: 120 Contact: <sip:0033972112...@172.20.1.1> Event: registration Content-Length: 0 --- Retransmitting #3 (no NAT) to 91.121.129.17:5060: REGISTER sip:91.121.129.17 SIP/2.0 Via: SIP/2.0/UDP 172.20.1.1:5060;branch=z9hG4bK40d5f950;rport Max-Forwards: 70 From: <sip:0033972112...@91.121.129.17>;tag=as02687bc2 To: <sip:0033972112...@91.121.129.17> Call-ID: 165ff552001c7f1e202e67200ae67...@172.25.3.51 CSeq: 1466 REGISTER User-Agent: Asterisk PBX 1.6.0.8 Authorization: Digest username="0033972112355", realm="sip.ovh.net", algorithm=MD5, uri="sip:91.121.129.17", nonce="0019c92d503f745637b43af4264a11db", response="04e848af655d00e03d032d9a1c2fae09", opaque="001934772ef6ed5" Expires: 120 Contact: <sip:0033972112...@172.20.1.1> Event: registration Content-Length: 0 --- Retransmitting #4 (no NAT) to 91.121.129.17:5060: REGISTER sip:91.121.129.17 SIP/2.0 Via: SIP/2.0/UDP 172.20.1.1:5060;branch=z9hG4bK40d5f950;rport Max-Forwards: 70 From: <sip:0033972112...@91.121.129.17>;tag=as02687bc2 To: <sip:0033972112...@91.121.129.17> Call-ID: 165ff552001c7f1e202e67200ae67...@172.25.3.51 CSeq: 1466 REGISTER User-Agent: Asterisk PBX 1.6.0.8 Authorization: Digest username="0033972112355", realm="sip.ovh.net", algorithm=MD5, uri="sip:91.121.129.17", nonce="0019c92d503f745637b43af4264a11db", response="04e848af655d00e03d032d9a1c2fae09", opaque="001934772ef6ed5" Expires: 120 Contact: <sip:0033972112...@172.20.1.1> Event: registration Content-Length: 0 --- thank you. Danny Nicholas a écrit : > And sip set debug peer ovh? > > > -----Original Message----- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Henry > Sent: Tuesday, April 07, 2009 4:18 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] i have a probleme and my asterisk and ovh > > [ovh] > type=peer > secret=xxxxxxxx > username=0033972xxxxxx > fromuser=0033972xxxxxx > host=sip.ovh.net > canreinvite=no > disallow=all > allow=g729 > tos_sip=1 ; Sets TOS for SIP packets. > tos_audio=1 ; Sets TOS for RTP audio packets. > tos_video=1 > dtmfmode=rfc28335 > relaxdtmf=yes > nat=yes > qualify=yes > insecure=port,invite > context=entrant-ovh > > thank you. > > Danny Nicholas a écrit : > >> Show us your sip.conf >> >> -----Original Message----- >> From: asterisk-users-boun...@lists.digium.com >> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Henry >> Sent: Tuesday, April 07, 2009 2:54 PM >> To: asterisk-users@lists.digium.com >> Subject: [asterisk-users] i have a probleme and my asterisk and ovh >> >> hello every body >> >> my connexion on ovh to pass in UNREACHABLE and not reidentified were not >> reboot the server. >> >> [Apr 7 20:17:21] NOTICE[19947]: chan_sip.c:15605 >> handle_response_peerpoke: Peer 'ovh' is now Lagged. (2067ms / 2000ms) >> [Apr 7 20:17:35] NOTICE[19947]: chan_sip.c:19829 sip_poke_noanswer: >> Peer 'ovh' is now UNREACHABLE! Last qualify: 2067 >> >> but my probleme is the adress ip 172.25.3.51 is not my adress. >> >> Really destroying SIP dialog >> '13ff06ae3e4bb3bf04987f5f5b497...@172.20.1.1' Method: OPTIONS >> Really destroying SIP dialog >> '6eac266b68dbc2566209fbb74aec7...@172.25.3.51' Method: REGISTER >> >> I do not know where she comes out, my asterisk ip is 172.20.1.1 and my >> router is 172.20.1.254. >> >> thank you for help >> >> _______________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> >> _______________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
