[asterisk-users] i have a probleme and my asterisk and ovh
Henry
henry at henpier.fr
Wed Apr 8 00:58:27 CDT 2009
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:asterisk at 172.20.1.1>;tag=as1545fb99
To: <sip:sip.ovh.net>
Contact: <sip:asterisk at 172.20.1.1>
Call-ID: 578ac87b06eaa6526aa313e130be3912 at 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:0033972112355 at 91.121.129.17>;tag=as16505dec
To: <sip:0033972112355 at 91.121.129.17>
Call-ID: 165ff552001c7f1e202e67200ae67e79 at 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:0033972112355 at 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:asterisk at 172.20.1.1>;tag=as1545fb99
To: <sip:sip.ovh.net>
Contact: <sip:asterisk at 172.20.1.1>
Call-ID: 578ac87b06eaa6526aa313e130be3912 at 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:asterisk at 172.20.1.1>;tag=as1545fb99
To: <sip:sip.ovh.net>
Contact: <sip:asterisk at 172.20.1.1>
Call-ID: 578ac87b06eaa6526aa313e130be3912 at 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:asterisk at 172.20.1.1>;tag=as1545fb99
To: <sip:sip.ovh.net>
Contact: <sip:asterisk at 172.20.1.1>
Call-ID: 578ac87b06eaa6526aa313e130be3912 at 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
'578ac87b06eaa6526aa313e130be3912 at 172.20.1.1' Method: OPTIONS
[Apr 8 07:57:48] NOTICE[25949]: chan_sip.c:9490 sip_reg_timeout: --
Registration for '0033972112355 at 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:0033972112355 at 91.121.129.17>;tag=as02687bc2
To: <sip:0033972112355 at 91.121.129.17>
Call-ID: 165ff552001c7f1e202e67200ae67e79 at 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:0033972112355 at 172.20.1.1>
Event: registration
Content-Length: 0
---
Really destroying SIP dialog
'165ff552001c7f1e202e67200ae67e79 at 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:0033972112355 at 91.121.129.17>;tag=as02687bc2
To: <sip:0033972112355 at 91.121.129.17>
Call-ID: 165ff552001c7f1e202e67200ae67e79 at 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:0033972112355 at 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:0033972112355 at 91.121.129.17>;tag=as02687bc2
To: <sip:0033972112355 at 91.121.129.17>
Call-ID: 165ff552001c7f1e202e67200ae67e79 at 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:0033972112355 at 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:0033972112355 at 91.121.129.17>;tag=as02687bc2
To: <sip:0033972112355 at 91.121.129.17>
Call-ID: 165ff552001c7f1e202e67200ae67e79 at 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:0033972112355 at 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:0033972112355 at 91.121.129.17>;tag=as02687bc2
To: <sip:0033972112355 at 91.121.129.17>
Call-ID: 165ff552001c7f1e202e67200ae67e79 at 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:0033972112355 at 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-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at 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-bounces at lists.digium.com
>> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Henry
>> Sent: Tuesday, April 07, 2009 2:54 PM
>> To: asterisk-users at 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
>> '13ff06ae3e4bb3bf04987f5f5b497269 at 172.20.1.1' Method: OPTIONS
>> Really destroying SIP dialog
>> '6eac266b68dbc2566209fbb74aec76cb at 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
>>
>> _______________________________________________
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>>
>>
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>>
>
>
> _______________________________________________
> -- 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
>
>
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