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Vieri

2010-May-12 09:51 UTC

[asterisk-users] SIP trunk between two Asterisk servers

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

Philipp von Klitzing

2010-May-12 10:49 UTC

head link

[asterisk-users] SIP trunk between two Asterisk servers

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.
Either 

a) set a secret and use that on both sides, or 
b) look at allowguest= and the default context and maybe the domain= 
settings, or
c) use insecure=invite

Philipp

Vardan

2010-May-12 13:00 UTC

head link

[asterisk-users] SIP trunk between two Asterisk servers

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
>
>
>
>
>
>

Philipp von Klitzing

2010-May-12 13:36 UTC

head link

[asterisk-users] SIP trunk between two Asterisk servers

Hi again!
> <--- 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.

Philipp

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