[asterisk-users] SIP trunk between two Asterisk servers
Vardan
hvardan71 at gmail.com
Wed May 12 12:44:49 CDT 2010
Please change the peers name in any server.
for example:
server1:
interboxsip1
server2:
interboxsip2
Vardan
Vieri wrote:
>
>
> --- On Wed, 5/12/10, Vardan<hvardan71 at gmail.com> wrote:
>
>> please show "sip show users" and sip
>> show peers"
>
> SERVER 2:
>
> sip show users (trimmed to just my sip test trunk):
>
> Username Secret Accountcode Def.Context ACL NAT
> interboxsip mycontext No RFC3581
>
> sip show peers (also trimmed):
>
> Name/username Host Dyn Nat ACL Port Status
> sipprovider/0000000001 w.x.y.z N 5060 OK (90 ms)
> interboxsip 192.168.250.111 5060 Unmonitored
> 7503/7503 10.215.146.190 D N A 5060 OK (20 ms)
> 7502/7502 10.215.146.203 D N A 5060 OK (20 ms)
> 7172/7172 192.168.250.7 D N A 13404 OK (40 ms)
> 7166/7166 10.215.146.200 D N A 5060 OK (20 ms)
> 7165/7165 10.215.248.12 D N A 5060 OK (1 ms)
> 7160/7160 10.215.146.182 D N A 5060 OK (20 ms)
> 7137/7137 192.168.250.6 D N A 25967 OK (10 ms)
> 7118/7118 192.168.250.10 D N A 14508 OK (1 ms)
> 7117/7117 10.215.146.185 D N A 5060 OK (20 ms)
> 7114/7114 192.168.250.8 D N A 12342 OK (10 ms)
> 7112/7112 192.168.250.31 D N A 19829 OK (10 ms)
> 7111/7111 192.168.250.32 D N A 35259 OK (80 ms)
> 7109/7109 (Unspecified) D N A 0 UNKNOWN
> 7097/7097 10.215.146.164 D N A 5060 OK (20 ms)
>
> SERVER 1:
>
> sip show users is identical.
>
> sip show peers (trimmed):
>
> Name/username Host Dyn Nat ACL Port Status
> sipprovider/0000000001 w.x.y.z N 5060 OK (79 ms)
> interboxsip 192.168.250.112 5060 Unmonitored
>
>>
>> vardan
>>
>> Vieri wrote:
>>>
>>>
>>> --- On Wed, 5/12/10, Philipp von Klitzing<klitzing at pool.informatik.rwth-aachen.de>
>> wrote:
>>>
>>>>> <--- SIP read from 192.168.250.111:5060
>> --->
>>>>> SIP/2.0 407 Proxy Authentication Required
>>>>
>>>> You need to run the SIP debug on 192.168.250.111
>> to learn
>>>> more about WHY
>>>> the 407 is issued. Have a close look and you are
>> likely to
>>>> understand it
>>>> right away.
>>>>
>>>> Also: Do not forget the "reload" after applying
>> changes to
>>>> sip.conf.
>>>
>>> I always do a "sip reload" after changes to sip
>> settings.
>>>
>>> Here are the SIP messages on 192.168.250.111 (Asterisk
>> server 1 - receiving end):
>>>
>>> <-- SIP read from 192.168.250.112:5060:
>>> INVITE sip:3666 at 192.168.250.111 SIP/2.0
>>> Via: SIP/2.0/UDP
>> 192.168.250.112:5060;branch=z9hG4bK504ddbc7;rport
>>> From:
>> "device"<sip:4053 at 192.168.250.112>;tag=as18a568d6
>>> To:<sip:3666 at 192.168.250.111>
>>> Contact:<sip:4053 at 192.168.250.112>
>>> Call-ID:
>> 328617546726e5d430538e80617716e1 at 192.168.250.112
>>> CSeq: 102 INVITE
>>> User-Agent: Asterisk PBX
>>> Max-Forwards: 70
>>> Date: Wed, 12 May 2010 09:20:26 GMT
>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
>> SUBSCRIBE, NOTIFY, INFO
>>> upported: replaces
>>> Content-Type: application/sdp
>>> Content-Length: 270
>>>
>>> v=0
>>> o=root 20611 20611 IN IP4 192.168.250.112
>>> s=session
>>> c=IN IP4 192.168.250.112
>>> t=0 0
>>> m=audio 14648 RTP/AVP 0 8 101
>>> a=rtpmap:0 PCMU/8000
>>> a=rtpmap:8 PCMA/8000
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-16
>>> a=silenceSupp:off - - - -
>>> a=ptime:20
>>> a=sendrecv
>>>
>>> --- (14 headers 13 lines) ---
>>> Using INVITE request as basis request -
>> 328617546726e5d430538e80617716e1 at 192.168.250.112
>>> Sending to 192.168.250.112 : 5060 (NAT)
>>> Reliably Transmitting (NAT) to 192.168.250.112:5060:
>>> SIP/2.0 407 Proxy Authentication Required
>>> Via: SIP/2.0/UDP
>> 192.168.250.112:5060;branch=z9hG4bK504ddbc7;received=192.168.250.112;rport=5060
>>> From:
>> "device"<sip:4053 at 192.168.250.112>;tag=as18a568d6
>>> To:<sip:3666 at 192.168.250.111>;tag=as57a19dac
>>> Call-ID:
>> 328617546726e5d430538e80617716e1 at 192.168.250.112
>>> CSeq: 102 INVITE
>>> User-Agent: Asterisk PBX
>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
>> SUBSCRIBE, NOTIFY
>>> Proxy-Authenticate: Digest algorithm=MD5,
>> realm="asterisk", nonce="1327c5b6"
>>> Content-Length: 0
>>>
>>>
>>> ---
>>> Scheduling destruction of call
>> '328617546726e5d430538e80617716e1 at 192.168.250.112' in 15000
>> ms
>>> Found user '4053'
>>>
>>> <-- SIP read from 192.168.250.112:5060:
>>> ACK sip:3666 at 192.168.250.111 SIP/2.0
>>> Via: SIP/2.0/UDP
>> 192.168.250.112:5060;branch=z9hG4bK504ddbc7;rport
>>> From:
>> "device"<sip:4053 at 192.168.250.112>;tag=as18a568d6
>>> To:<sip:3666 at 192.168.250.111>;tag=as57a19dac
>>> Contact:<sip:4053 at 192.168.250.112>
>>> Call-ID:
>> 328617546726e5d430538e80617716e1 at 192.168.250.112
>>> CSeq: 102 ACK
>>> User-Agent: Asterisk PBX
>>> Max-Forwards: 70
>>> Content-Length: 0
>>>
>>> Can you deduce from this what I'm doing wrong?
>>>
>>> Thanks,
>>>
>>> Vieri
>>>
>>>
>>>
>>>
>>>
>>
>>
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>
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