[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=172.17.9.1;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
Re: [asterisk-users] Problem with SIP trunk I've set up between two * boxes.
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
Re: [asterisk-users] Problem with SIP trunk I've set up between two * boxes.
On 2012-12-10 16:16, Danny Nicholas wrote:
Does each box show up in the others SIP SHOW PEERS?
Yup -- each shows in the other's. Sorry I didn't mention that.
-Ken
-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
Re: [asterisk-users] Problem with SIP trunk I've set up between two * boxes.
Looks like a connectivity issue, doesn't it?
IP of box2, 172.17.145.145, doesn't show up even once in the SIP dialogues.
What happens on box2 (asterisk -vvvr and tcpdump port 5060) in the
moment that you place a call through box1 to box2?
Also what's strange is that you are trying to call from box2 to box2?
Because local_SIP is the context on box2, and on box1 it's adhearsion.
The console message you pasted shows @local_SIP however, so it looks
like you are calling from box2 to box2?
Am 10.12.2012 22:53, schrieb Ken D'Ambrosio:
On 2012-12-10 16:16, Danny Nicholas wrote:
Does each box show up in the others SIP SHOW PEERS?
Yup -- each shows in the other's. Sorry I didn't mention that.
-Ken
-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
Re: [asterisk-users] Problem with SIP trunk I've set up between two * boxes.
Hi, Ken
I have almost the same setup as yours: new
asterisk-SIP-Trixbox(Asterisk 1.4)---PRIpots
Here are my configs:
new box sip.conf:
[126]
directmedia=no
type=friend
host=trixbox_IP_addr
secret=my_secret
username=126 ;this is for outgoing calls from new asterisk via trixbox
fromuser=126 ;this is for outgoing calls from new asterisk via trixbox
context=default
disallow=all
allow=alaw
allow=ulaw
qualify=yes
qualifyfreq=60
nat=yes
pickupgroup=1
callgroup=1
trixbox
[126]
type=friend
secret=mysecret
record_out=Adhoc
record_in=Adhoc
qualify=yes
port=5060
pickupgroup=
nat=yes
mailbox=
host=dynamic
dtmfmode=rfc2833
dial=SIP/126
context=from-internal
canreinvite=no
callgroup=
callerid=device 126
accountcode=
call-limit=50
New box's account (126) registers to the Trixbox so as to make incoming calls
from trixbox to new box possible.
The config in the new box implies that the trixbox require authorization in
calls from the new box (username and fromuser options are necessary for this).
Actually looking through the sip.conf in 1.8 asterisk I found that there are
auth option as well as remotesecret and remoteuser - but I can not
understand how they work in case if I need to authorise my outgoing calls
(probably sip.conf will be more logical in the future 12th version).
Hope this helps.
Dmitry Pavlenko
From: Ken D'Ambrosio k...@jots.org
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, December 11, 2012 3:53 AM
Subject: Re: [asterisk-users] Problem with SIP trunk I've set up between two *
boxes.
On 2012-12-10 16:16, Danny Nicholas wrote:
Does each box show up in the others SIP SHOW PEERS?
Yup -- each shows in the other's. Sorry I didn't mention that.
-Ken
-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
