Does each box show up in the others "SIP SHOW PEERS"?
-----Original Message-----
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ken D'Ambrosio
Sent: Monday, December 10, 2012 2:59 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Problem with SIP trunk I've set up between two *
boxes.
Hi! I'm trying to set up a SIP trunk so that I can test calls, etc.,
between a new Asterisk box, and an old 1.4 box.
---------------------------------------------------------------------------
New box:
root@asterisk1:/etc/asterisk# head -1 sip.conf #include siptrunk.conf
siptrunk.conf:
[box1] ; All box1 extensions; see extensions.conf type=peer
context=adhearsion
host=172.17.0.17 ; IP for old system
disallow=all
allow=g729
canreinvite=yes
qualify=no
Old box:
root@asterisk1:/etc/asterisk# head -1 sip.conf #include siptrunk.conf
siptrunk.conf:
[box2] ; All box2 extensions; see extensions.conf type=peer
context=local_SIP
host=172.17.145.145 ; IP for new system
disallow=all
allow=g729
canreinvite=yes
qualify=no
extensions.conf snippet:
[local_SIP]
include => aggregate
include => passthrough
exten => _7XXX,1,Dial(SIP/box2/${EXTEN}) exten => _7XXX,2,Hangup()
-----------------------------------------------------------------------
When I dial, all I get is (I'll attach the full dialog up to that point from
SIP debug, below.)
-- Executing [7444@local_SIP:1] Dial("SIP/6110-08291cb0",
"SIP/box2/7444") in new stack
-- Couldn't call box2/7444
Scheduling destruction of SIP dialog
'1f18dd4b4ee8f7583041de280f307c18@172.17.0.17' in 32000 ms (Method:
INVITE)
== Everyone is busy/congested at this time (0:0/0/0)
-----------------------------------------------------------------------
Where am I goofing up? Any pointers?
Thanks!
-Ken
-----------------------------------------------------------------------
INVITE sip:7444@172.17.0.17 SIP/2.0
Via: SIP/2.0/UDP
172.17.9.1:55388;rport;branch=z9hG4bKPj95Yj53gkGda17Af72e33RKrQBygd0iP0
Max-Forwards: 70
From: <sip:6110@172.17.0.17>;tag=UzpwMu8tx77dICqAkH0V9MpQn-bMS.zN
To: <sip:7444@172.17.0.17>
Contact: <sip:6110@172.17.9.1:55388;ob>
Call-ID: nUiGauUpyxjNOJfcZog476ws.Art7jZS
CSeq: 24152 INVITE
Route: <sip:172.17.0.17;transport=udp;lr>
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER,
MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: CSipSimple_d2vzw-16/r1916
Content-Type: application/sdp
Content-Length: 354
v=0
o=- 3564161970 3564161970 IN IP4 172.17.9.1 s=pjmedia c=IN IP4 172.17.9.1
t=0 0
m=audio 4006 RTP/AVP 96 3 0 8 101
c=IN IP4 172.17.9.1
a=rtcp:4007 IN IP4 172.17.9.1
a=sendrecv
a=rtpmap:96 SILK/8000
a=fmtp:96 useinbandfec=0
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (16 headers 16 lines) ---
Sending to 172.17.9.1 : 55388 (NAT)
Using INVITE request as basis request - nUiGauUpyxjNOJfcZog476ws.Art7jZS
<--- Reliably Transmitting (no NAT) to 172.17.9.1:55388 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
172.17.9.1:55388;branch=z9hG4bKPj95Yj53gkGda17Af72e33RKrQBygd0iP0;received=1
72.17.9.1;rport=55388
From: <sip:6110@172.17.0.17>;tag=UzpwMu8tx77dICqAkH0V9MpQn-bMS.zN
To: <sip:7444@172.17.0.17>;tag=as595faea1
Call-ID: nUiGauUpyxjNOJfcZog476ws.Art7jZS
CSeq: 24152 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="16883b72"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'nUiGauUpyxjNOJfcZog476ws.Art7jZS'
in 32000 ms (Method: INVITE)
Found user '6110'
<--- SIP read from 172.17.9.1:55388 ---> ACK sip:7444@172.17.0.17 SIP/2.0
Via: SIP/2.0/UDP
172.17.9.1:55388;rport;branch=z9hG4bKPj95Yj53gkGda17Af72e33RKrQBygd0iP0
Max-Forwards: 70
From: <sip:6110@172.17.0.17>;tag=UzpwMu8tx77dICqAkH0V9MpQn-bMS.zN
To: <sip:7444@172.17.0.17>;tag=as595faea1
Call-ID: nUiGauUpyxjNOJfcZog476ws.Art7jZS
CSeq: 24152 ACK
Route: <sip:172.17.0.17;transport=udp;lr>
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
<--- SIP read from 172.17.9.1:55388 --->
INVITE sip:7444@172.17.0.17 SIP/2.0
Via: SIP/2.0/UDP
172.17.9.1:55388;rport;branch=z9hG4bKPjvX4It3-WYRqMlyhU9peo5ewQRIgQ4qd1
Max-Forwards: 70
From: <sip:6110@172.17.0.17>;tag=UzpwMu8tx77dICqAkH0V9MpQn-bMS.zN
To: <sip:7444@172.17.0.17>
Contact: <sip:6110@172.17.9.1:55388;ob>
Call-ID: nUiGauUpyxjNOJfcZog476ws.Art7jZS
CSeq: 24153 INVITE
Route: <sip:172.17.0.17;transport=udp;lr>
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY,
REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: CSipSimple_d2vzw-16/r1916
Proxy-Authorization: Digest username="6110", realm="asterisk",
nonce="16883b72", uri="sip:7444@172.17.0.17",
response="b75389c5938b4f185b3d31bd4463abf3", algorithm=MD5
Content-Type: application/sdp
Content-Length: 354
v=0
o=- 3564161970 3564161970 IN IP4 172.17.9.1
s=pjmedia
c=IN IP4 172.17.9.1
t=0 0
m=audio 4006 RTP/AVP 96 3 0 8 101
c=IN IP4 172.17.9.1
a=rtcp:4007 IN IP4 172.17.9.1
a=sendrecv
a=rtpmap:96 SILK/8000
a=fmtp:96 useinbandfec=0
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (17 headers 16 lines) ---
Sending to 172.17.9.1 : 55388 (NAT)
Using INVITE request as basis request -
nUiGauUpyxjNOJfcZog476ws.Art7jZS
Found user '6110'
Found RTP audio format 96
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 172.17.9.1:4006
Found unknown media description format SILK for ID 96
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0xe
(gsm|ulaw|alaw)/video=0x0 (nothing), combined - 0x6 (gsm|ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 172.17.9.1:4006
Looking for 7444 in local_SIP (domain 172.17.0.17)
list_route: hop: <sip:6110@172.17.9.1:55388;ob>
<--- Transmitting (no NAT) to 172.17.9.1:55388 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
172.17.9.1:55388;branch=z9hG4bKPjvX4It3-WYRqMlyhU9peo5ewQRIgQ4qd1;received=1
72.17.9.1;rport=55388
From: <sip:6110@172.17.0.17>;tag=UzpwMu8tx77dICqAkH0V9MpQn-bMS.zN
To: <sip:7444@172.17.0.17>
Call-ID: nUiGauUpyxjNOJfcZog476ws.Art7jZS
CSeq: 24153 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:7444@172.17.0.17>
Content-Length: 0
<------------>
-- Executing [7444@local_SIP:1] Dial("SIP/6110-08293240",
"SIP/box2/7444") in new stack
-- Couldn't call box2/7444
Scheduling destruction of SIP dialog
'2e08d34c5211d82d7e9afa67550458cb@172.17.0.17' in 32000 ms (Method:
INVITE)
== Everyone is busy/congested at this time (0:0/0/0)
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