[asterisk-users] SIP trunk between two Asterisk servers

Wed May 12 08:59:50 CDT 2010

Please look in any conf file that have any relations with sip.conf.
I think you have some records.
And one also, you take this message when calling in both direction? 
(server1 call server2 and server2 call server1)

Vardan

Vieri wrote:
>
>
> --- On Wed, 5/12/10, Vardan<hvardan71 at gmail.com>  wrote:
>
>> 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
>
>
> Hi,
>
> I tried your suggestion (then I even added the insecure param) but I still get the error:
>
> SIP/2.0 407 Proxy Authentication Required
>
>
> on server 2:
>
> [interboxsip]
> type=friend
> insecure=invite
> host=192.168.250.111
> context=mycontext
> disallow=all
> allow=ulaw
> allow=alaw
> allow=g729
> allow=gsm
>
> on server 1:
>
> [interboxsip]
> type=friend
> insecure=invite
> host=192.168.250.112
> context=mycontext
> disallow=all
> allow=ulaw
> allow=alaw
> allow=g729
> allow=gsm
>
> to call from one server to the other:
>
> exten =>  3666,1,Dial(SIP/interboxsip/${EXTEN},20,rt)
> exten =>  3666,n,HangUp()
>
> This should be simple but it puzzles me why it's not working.
>
> Thanks,
>
> Vieri
>
>
>
>
>