[asterisk-users] SIP trunk between two Asterisk servers

Wed May 12 08:46:44 CDT 2010

--- 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