[asterisk-users] SIP trunk between two Asterisk servers

[Vardan]

please show "sip show users" and sip show peers"

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
>
>
>
>
>