[asterisk-users] SIP trunk between two Asterisk servers

Wed May 12 04:51:59 CDT 2010

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