[asterisk-users] SIP trunk between two Asterisk servers

Wed May 12 08:08:14 CDT 2010

I have forget to write for outcall in extension

server1:
[calltoserver2]
  exten =>  _X.,1,Noop(Call to server2)
  exten =>  _X.,2,Dial(SIP/interboxserver2/${EXTEN})
  exten =>  _X.,3,Hangup

server2:

[calltoserver1]
  exten =>  _X.,1,Noop(Call to server1)
  exten =>  _X.,2,Dial(SIP/interboxserver1/${EXTEN})
  exten =>  _X.,3,Hangup

:)

Vardan


Vardan wrote:
> Hello
>
> Server1:
>
> sip.conf
>
> [interboxserver2]
> type=friend
> host=192.168.250.112
> context=callfromserver2
> disallow=all
> allow=ulaw
> allow=alaw
> allow=g729
>
> extensions.conf
>
> [callfromserver2]
>
> exten =>  _X.,1,Noop(Call from server2)
> exten =>  _X.,2,Dial(SIP/${EXTEN})
> exten =>  _X.,3,Hangup
>
>
> Server2:
>
> sip.conf
>
> [interboxserver1]
> type=friend
> host=192.168.250.111
> context=callfromserver1
> disallow=all
> allow=ulaw
> allow=alaw
> allow=g729
>
> extensions.conf
>
> [callfromserver1]
>
> exten =>  _X.,1,Noop(Call from server1)
> exten =>  _X.,2,Dial(SIP/${EXTEN})
> exten =>  _X.,3,Hangup
>
>
> Try so, I think it must work.
> And also, look and delete any another records in both servers in
> sip.conf about this servers settings.
>
> Vardan
>
>
> Vieri wrote:
>> Hi,
>>
>> I'm trying to setup a SIP trunk between 2 Asterisk servers on the same LAN (no NAT, no firewalls).
>>
>> With IAX2 all's fine but I'm unable to setup SIP. I must be missing something obvious.
>>
>> I followed the simple example at http://www.panoramisk.com/90/sip-trunk-with-asterisk/en/.
>>
>> so Asterisk server 1 (192.168.250.111) sip.conf contains:
>>
>> [interboxsip]
>> type=peer
>> host=192.168.250.112
>> context=mycontext
>>
>> Asterisk server 2 (192.168.250.112) sip.conf contains:
>>
>> [interboxsip]
>> type=peer
>> host=192.168.250.111
>> context=mycontext
>>
>> I dialed from a SIP extension (4053) in server 2 (192.168.250.112) to 3666 in server 1 (192.168.250.111) via the interboxsip SIP trunk.
>>
>> The call fails and according to the SIP messages it seems to be an authentication problem.
>>
>> What am I missing?
>>
>> SIP messages on 192.168.250.112 (Asterisk server 2 - transmitting call):
>>
>>       -- Executing [3666 at from-internal:2] Dial("SIP/4053-00006dea", "SIP/interboxsip/3666|300|rt") in new stack
>> Audio is at 192.168.250.112 port 15850
>> Adding codec 0x4 (ulaw) to SDP
>> Adding codec 0x8 (alaw) to SDP
>> Adding non-codec 0x1 (telephone-event) to SDP
>> Reliably Transmitting (no NAT) to 192.168.250.111:5060:
>> INVITE sip:3666 at 192.168.250.111 SIP/2.0
>> Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK3c951a1d;rport
>> From: "device"<sip:4053 at 192.168.250.112>;tag=as4d17a185
>> To:<sip:3666 at 192.168.250.111>
>> Contact:<sip:4053 at 192.168.250.112>
>> Call-ID: 3770a8004ce882dd3c89c1d91b5aa2d2 at 192.168.250.112
>> CSeq: 102 INVITE
>> User-Agent: Asterisk PBX
>> Max-Forwards: 70
>> Date: Wed, 12 May 2010 09:13:06 GMT
>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
>> Supported: 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 15850 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
>>
>> ---
>>       -- Called interboxsip/3666
>>
>> <--- SIP read from 192.168.250.111:5060 --->
>> SIP/2.0 407 Proxy Authentication Required
>> Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK3c951a1d;received=192.168.250.112;rport=5060
>> From: "device"<sip:4053 at 192.168.250.112>;tag=as4d17a185
>> To:<sip:3666 at 192.168.250.111>;tag=as00842b82
>> Call-ID: 3770a8004ce882dd3c89c1d91b5aa2d2 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="2545a5dd"
>> Content-Length: 0
>>
>>
>> <------------->
>>
>> --- (10 headers 0 lines) ---
>> Transmitting (no NAT) to 192.168.250.111:5060:
>> ACK sip:3666 at 192.168.250.111 SIP/2.0
>> Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK3c951a1d;rport
>> From: "device"<sip:4053 at 192.168.250.112>;tag=as4d17a185
>> To:<sip:3666 at 192.168.250.111>;tag=as00842b82
>> Contact:<sip:4053 at 192.168.250.112>
>> Call-ID: 3770a8004ce882dd3c89c1d91b5aa2d2 at 192.168.250.112
>> CSeq: 102 ACK
>> User-Agent: Asterisk PBX
>> Max-Forwards: 70
>> Content-Length: 0
>>
>>
>> ---
>>       -- SIP/interboxsip-00006deb is circuit-busy
>>
>>
>> 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
>>
>>
>>
>>
>>
>>
>
>