403 Forbidden while trying to call from one * to another

Asterisk Asterisk Support
1

I have 2 servers with Asterisks on them: 192.168.241.98 and 192.168.243.112.

There is a valid registration on the first:

CLI output:

CLI> sip show registry Host dnsmgr Username Refresh State Reg.Time 192.168.243.112:5060 N wagateway 105 Registered Wed, 26 Jun 2013 16:42:42

And peers on 243.112 are just fine:

CLI> sip show peers Name/username Host Dyn Forcerport ACL Port Status Description wacaller/wacaller 192.168.242.235 D a 5062 OK (13 ms) wagateway/s 192.168.241.98 D a 5060 OK (1 ms)

extensions.conf on 243.112:

[code][watest]

exten => 123123123,1,NoOp()
exten => 123123123,n,Dial(SIP/wagateway)
exten => 123123123,n,Hangup()[/code]

sip.conf on 243.112:

[code][wacaller]
type=friend
secret=qwerty
host=dynamic
context=watest
qualify=yes
allow=ulaw
allow=alaw

[wagateway]
type=friend
secret=qwerty
fromuser=wagateway
host=dynamic
context=watest
qualify=yes
allow=ulaw
allow=alaw
insecure=port,invite[/code]

Now I try to call 123123123 from wacaller’s Grandstream phone:

[code]<— SIP read from UDP:192.168.242.235:5062 —>
INVITE sip:123123123@192.168.243.112 SIP/2.0
Via: SIP/2.0/UDP 192.168.242.235:5062;branch=z9hG4bK97733114;rport
From: “WACaller” sip:wacaller@192.168.243.112;tag=1014197566
To: sip:123123123@192.168.243.112
Call-ID: 298833112-5062-25@BJC.BGI.CEC.CDF
CSeq: 240 INVITE
Contact: “WACaller” sip:wacaller@192.168.242.235:5062
Max-Forwards: 70
User-Agent: Grandstream GXP1400 1.0.4.13
Privacy: none
P-Preferred-Identity: “WACaller” sip:wacaller@192.168.243.112
Supported: replaces, path, timer
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length: 412

v=0
o=wacaller 8000 8000 IN IP4 192.168.242.235
s=SIP Call
c=IN IP4 192.168.242.235
t=0 0
m=audio 5004 RTP/AVP 0 8 4 18 9 97 2 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:2 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
— (16 headers 19 lines) —
Sending to 192.168.242.235:5062 (no NAT)
Sending to 192.168.242.235:5062 (no NAT)
Using INVITE request as basis request - 298833112-5062-25@BJC.BGI.CEC.CDF
Found peer ‘wacaller’ for ‘wacaller’ from 192.168.242.235:5062

<— Reliably Transmitting (no NAT) to 192.168.242.235:5062 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.242.235:5062;branch=z9hG4bK97733114;received=192.168.242.235;rport=5062
From: “WACaller” sip:wacaller@192.168.243.112;tag=1014197566
To: sip:123123123@192.168.243.112;tag=as5a3de236
Call-ID: 298833112-5062-25@BJC.BGI.CEC.CDF
CSeq: 240 INVITE
Server: Asterisk PBX SVN-trunk-r385782
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="4f84bef0"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘298833112-5062-25@BJC.BGI.CEC.CDF’ in 6400 ms (Method: INVITE)

<— SIP read from UDP:192.168.242.235:5062 —>
ACK sip:123123123@192.168.243.112 SIP/2.0
Via: SIP/2.0/UDP 192.168.242.235:5062;branch=z9hG4bK97733114;rport
From: “WACaller” sip:wacaller@192.168.243.112;tag=1014197566
To: sip:123123123@192.168.243.112;tag=as5a3de236
Call-ID: 298833112-5062-25@BJC.BGI.CEC.CDF
CSeq: 240 ACK
Content-Length: 0

<------------->
— (7 headers 0 lines) —

<— SIP read from UDP:192.168.242.235:5062 —>
INVITE sip:123123123@192.168.243.112 SIP/2.0
Via: SIP/2.0/UDP 192.168.242.235:5062;branch=z9hG4bK1881861609;rport
From: “WACaller” sip:wacaller@192.168.243.112;tag=1014197566
To: sip:123123123@192.168.243.112
Call-ID: 298833112-5062-25@BJC.BGI.CEC.CDF
CSeq: 241 INVITE
Contact: “WACaller” sip:wacaller@192.168.242.235:5062
Authorization: Digest username=“wacaller”, realm=“asterisk”, nonce=“4f84bef0”, uri="sip:123123123@192.168.243.112", response=“53cdb5b8c1822c80870faab15a6dc6d2”, algorithm=MD5
Max-Forwards: 70
User-Agent: Grandstream GXP1400 1.0.4.13
Privacy: none
P-Preferred-Identity: “WACaller” sip:wacaller@192.168.243.112
Supported: replaces, path, timer
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length: 412

v=0
o=wacaller 8000 8000 IN IP4 192.168.242.235
s=SIP Call
c=IN IP4 192.168.242.235
t=0 0
m=audio 5004 RTP/AVP 0 8 4 18 9 97 2 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:2 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
— (17 headers 19 lines) —
Sending to 192.168.242.235:5062 (no NAT)
Using INVITE request as basis request - 298833112-5062-25@BJC.BGI.CEC.CDF
Found peer ‘wacaller’ for ‘wacaller’ from 192.168.242.235:5062
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 4
Found RTP audio format 18
Found RTP audio format 9
Found RTP audio format 97
Found RTP audio format 2
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G723 for ID 4
Found audio description format G729 for ID 18
Found audio description format G722 for ID 9
Found audio description format iLBC for ID 97
Found audio description format G726-32 for ID 2
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw|alaw|h263|testlaw), peer - audio=(g723|ulaw|alaw|g726|g729|ilbc|g722)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.242.235:5004
Looking for 123123123 in watest (domain 192.168.243.112)
list_route: route/path hop: sip:wacaller@192.168.242.235:5062

<— Transmitting (no NAT) to 192.168.242.235:5062 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.242.235:5062;branch=z9hG4bK1881861609;received=192.168.242.235;rport=5062
From: “WACaller” sip:wacaller@192.168.243.112;tag=1014197566
To: sip:123123123@192.168.243.112
Call-ID: 298833112-5062-25@BJC.BGI.CEC.CDF
CSeq: 241 INVITE
Server: Asterisk PBX SVN-trunk-r385782
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:123123123@192.168.243.112:5060
Content-Length: 0

<------------>
Audio is at 17372
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100017 (testlaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.241.98:5060:
INVITE sip:s@192.168.241.98:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.243.112:5060;branch=z9hG4bK67b16b32;rport
Max-Forwards: 70
From: “WACaller” sip:wagateway@192.168.243.112;tag=as3f5f372a
To: sip:s@192.168.241.98:5060
Contact: sip:wagateway@192.168.243.112:5060
Call-ID: 758899861bee35980dadd87912ef805a@192.168.243.112:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX SVN-trunk-r385782
Date: Wed, 26 Jun 2013 08:31:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 326

v=0
o=root 2059284449 2059284449 IN IP4 192.168.243.112
s=Asterisk PBX SVN-trunk-r385782
c=IN IP4 192.168.243.112
t=0 0
m=audio 17372 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<— SIP read from UDP:192.168.241.98:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.243.112:5060;branch=z9hG4bK67b16b32;received=192.168.243.112;rport=5060
From: “WACaller” sip:wagateway@192.168.243.112;tag=as3f5f372a
To: sip:s@192.168.241.98:5060;tag=as22eeeac0
Call-ID: 758899861bee35980dadd87912ef805a@192.168.243.112:5060
CSeq: 102 INVITE
Server: Asterisk PBX 1.8.12.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="603b4bbf"
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Transmitting (NAT) to 192.168.241.98:5060:
ACK sip:s@192.168.241.98:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.243.112:5060;branch=z9hG4bK67b16b32;rport
Max-Forwards: 70
From: “WACaller” sip:wagateway@192.168.243.112;tag=as3f5f372a
To: sip:s@192.168.241.98:5060;tag=as22eeeac0
Contact: sip:wagateway@192.168.243.112:5060
Call-ID: 758899861bee35980dadd87912ef805a@192.168.243.112:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX SVN-trunk-r385782
Content-Length: 0

Audio is at 17372
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100017 (testlaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.241.98:5060:
INVITE sip:s@192.168.241.98:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.243.112:5060;branch=z9hG4bK6cd34725;rport
Max-Forwards: 70
From: “WACaller” sip:wagateway@192.168.243.112;tag=as3f5f372a
To: sip:s@192.168.241.98:5060
Contact: sip:wagateway@192.168.243.112:5060
Call-ID: 758899861bee35980dadd87912ef805a@192.168.243.112:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX SVN-trunk-r385782
Authorization: Digest username=“s”, realm=“asterisk”, algorithm=MD5, uri=“sip:s@192.168.241.98:5060”, nonce=“603b4bbf”, response="059cae207fb81fb76ea9061f71258895"
Date: Wed, 26 Jun 2013 08:31:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 326

v=0
o=root 2059284449 2059284450 IN IP4 192.168.243.112
s=Asterisk PBX SVN-trunk-r385782
c=IN IP4 192.168.243.112
t=0 0
m=audio 17372 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<— SIP read from UDP:192.168.241.98:5060 —>
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.243.112:5060;branch=z9hG4bK6cd34725;received=192.168.243.112;rport=5060
From: “WACaller” sip:wagateway@192.168.243.112;tag=as3f5f372a
To: sip:s@192.168.241.98:5060;tag=as22eeeac0
Call-ID: 758899861bee35980dadd87912ef805a@192.168.243.112:5060
CSeq: 103 INVITE
Server: Asterisk PBX 1.8.12.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
— (10 headers 0 lines) —
Transmitting (NAT) to 192.168.241.98:5060:
ACK sip:s@192.168.241.98:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.243.112:5060;branch=z9hG4bK6cd34725;rport
Max-Forwards: 70
From: “WACaller” sip:wagateway@192.168.243.112;tag=as3f5f372a
To: sip:s@192.168.241.98:5060;tag=as22eeeac0
Contact: sip:wagateway@192.168.243.112:5060
Call-ID: 758899861bee35980dadd87912ef805a@192.168.243.112:5060
CSeq: 103 ACK
User-Agent: Asterisk PBX SVN-trunk-r385782
Content-Length: 0

[Jun 26 16:31:48] WARNING[20447][C-0000000a]: chan_sip.c:23213 handle_response_invite: Received response: “Forbidden” from '“WACaller” sip:wagateway@192.168.243.112;tag=as3f5f372a’
Scheduling destruction of SIP dialog ‘758899861bee35980dadd87912ef805a@192.168.243.112:5060’ in 6400 ms (Method: INVITE)
Scheduling destruction of SIP dialog ‘298833112-5062-25@BJC.BGI.CEC.CDF’ in 6400 ms (Method: INVITE)[/code]

Any help?

2

You need to provide the logs and configuration from the destination system.

Usually it is better to use static addresses, if you can. insecure=invite means there is no authentication inbound. If you control both ends, set them to the same password, or use remotesecret, if it is available in your Asterisk version. Few systems need insecure=port, and it is bad practice to include it without a good reason.

3

[quote=“david55”]You need to provide the logs and configuration from the destination system.

Usually it is better to use static addresses, if you can. insecure=invite means there is no authentication inbound. If you control both ends, set them to the same password, or use remotesecret, if it is available in your Asterisk version. Few systems need insecure=port, and it is bad practice to include it without a good reason.[/quote]

Destination system uses mysql database for configuration. This is for extension using (not sure it goes until this point):

636,"AC000001_in","123123123","1","GotoIf","${name}?${CONTEXT},${EXTEN},4:" 637,"AC000001_in","123123123","2","Set","name=${CALLERID(name)}" 638,"AC000001_in","123123123","3","Set","__GLOBAL_NAME=${name}" 639,"AC000001_in","123123123","4","GotoIf","${num}?${CONTEXT},${EXTEN},6:" 640,"AC000001_in","123123123","5","Set","num=${CALLERID(num)}" 641,"AC000001_in","123123123","6","Verbose","0,End name and num substitute for CALLERID" 642,"AC000001_in","123123123","7","GotoIf","${TRANSFER_COUNT}?${CONTEXT},${EXTEN},10:" 643,"AC000001_in","123123123","8","Set","__TRANSFER_COUNT=1" 644,"AC000001_in","123123123","9","Goto","AC000001_in,${EXTEN},11" 645,"AC000001_in","123123123","10","Set","__TRANSFER_COUNT=$[${TRANSFER_COUNT}+1]" 646,"AC000001_in","123123123","11","Verbose","0,TRANSFER_COUNT is: ${TRANSFER_COUNT}" 647,"AC000001_in","123123123","12","Hangup","3"
The Register string was already provided.

Sip debug from destination server:

[code]<— SIP read from UDP:192.168.243.112:5060 —>
INVITE sip:s@192.168.241.98:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.243.112:5060;branch=z9hG4bK3159e4b1;rport
Max-Forwards: 70
From: “WACaller” sip:wagateway@192.168.243.112;tag=as30b27eae
To: sip:s@192.168.241.98:5060
Contact: sip:wagateway@192.168.243.112:5060
Call-ID: 5dc37059030845ca3d974c513993876d@192.168.243.112:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX SVN-trunk-r385782
Date: Thu, 27 Jun 2013 01:27:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 326

v=0
o=root 1301894386 1301894386 IN IP4 192.168.243.112
s=Asterisk PBX SVN-trunk-r385782
c=IN IP4 192.168.243.112
t=0 0
m=audio 15838 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------->
— (14 headers 14 lines) —
Sending to 192.168.243.112:5060 (NAT)
Using INVITE request as basis request - 5dc37059030845ca3d974c513993876d@192.168.243.112:5060
Found peer ‘wagateway’ for ‘wagateway’ from 192.168.243.112:5060

<— Reliably Transmitting (no NAT) to 192.168.243.112:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.243.112:5060;branch=z9hG4bK3159e4b1;received=192.168.243.112;rport=5060
From: “WACaller” sip:wagateway@192.168.243.112;tag=as30b27eae
To: sip:s@192.168.241.98:5060;tag=as671c0824
Call-ID: 5dc37059030845ca3d974c513993876d@192.168.243.112:5060
CSeq: 102 INVITE
Server: Asterisk PBX 1.8.12.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="0b63a660"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘5dc37059030845ca3d974c513993876d@192.168.243.112:5060’ in 32000 ms (Method: INVITE)

<— SIP read from UDP:192.168.243.112:5060 —>
ACK sip:s@192.168.241.98:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.243.112:5060;branch=z9hG4bK3159e4b1;rport
Max-Forwards: 70
From: “WACaller” sip:wagateway@192.168.243.112;tag=as30b27eae
To: sip:s@192.168.241.98:5060;tag=as671c0824
Contact: sip:wagateway@192.168.243.112:5060
Call-ID: 5dc37059030845ca3d974c513993876d@192.168.243.112:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX SVN-trunk-r385782
Content-Length: 0

<------------->
— (10 headers 0 lines) —

<— SIP read from UDP:192.168.243.112:5060 —>
INVITE sip:s@192.168.241.98:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.243.112:5060;branch=z9hG4bK40e56655;rport
Max-Forwards: 70
From: “WACaller” sip:wagateway@192.168.243.112;tag=as30b27eae
To: sip:s@192.168.241.98:5060
Contact: sip:wagateway@192.168.243.112:5060
Call-ID: 5dc37059030845ca3d974c513993876d@192.168.243.112:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX SVN-trunk-r385782
Authorization: Digest username=“s”, realm=“asterisk”, algorithm=MD5, uri=“sip:s@192.168.241.98:5060”, nonce=“0b63a660”, response="537f37fe2fb8d0fd40733cb190ea70c8"
Date: Thu, 27 Jun 2013 01:27:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 326

v=0
o=root 1301894386 1301894387 IN IP4 192.168.243.112
s=Asterisk PBX SVN-trunk-r385782
c=IN IP4 192.168.243.112
t=0 0
m=audio 15838 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------->
— (15 headers 14 lines) —
Sending to 192.168.243.112:5060 (no NAT)
Using INVITE request as basis request - 5dc37059030845ca3d974c513993876d@192.168.243.112:5060
Found peer ‘wagateway’ for ‘wagateway’ from 192.168.243.112:5060

<— Reliably Transmitting (no NAT) to 192.168.243.112:5060 —>
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.243.112:5060;branch=z9hG4bK40e56655;received=192.168.243.112;rport=5060
From: “WACaller” sip:wagateway@192.168.243.112;tag=as30b27eae
To: sip:s@192.168.241.98:5060;tag=as671c0824
Call-ID: 5dc37059030845ca3d974c513993876d@192.168.243.112:5060
CSeq: 103 INVITE
Server: Asterisk PBX 1.8.12.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘5dc37059030845ca3d974c513993876d@192.168.243.112:5060’ in 32000 ms (Method: INVITE)

<— SIP read from UDP:192.168.243.112:5060 —>
ACK sip:s@192.168.241.98:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.243.112:5060;branch=z9hG4bK40e56655;rport
Max-Forwards: 70
From: “WACaller” sip:wagateway@192.168.243.112;tag=as30b27eae
To: sip:s@192.168.241.98:5060;tag=as671c0824
Contact: sip:wagateway@192.168.243.112:5060
Call-ID: 5dc37059030845ca3d974c513993876d@192.168.243.112:5060
CSeq: 103 ACK
User-Agent: Asterisk PBX SVN-trunk-r385782
Content-Length: 0

<------------->
— (10 headers 0 lines) —
dev-ast*CLI> sip set debug off
SIP Debugging Disabled[/code]
I used insecure parameter just in case it may help, but it didn’t. Static adresses didn’t help either.

4

You need to increase the debugging level and/or enable the full log and take output from that, not the console. The console doesn’t include debug levels. sip set debug actually generates “verbose output”.

Also, this is almost certainly a sip.conf setting, problem, not an extensions.conf one.

5

[quote=“david55”]You need to increase the debugging level and/or enable the full log and take output from that, not the console. The console doesn’t include debug levels. sip set debug actually generates “verbose output”.

Also, this is almost certainly a sip.conf setting, problem, not an extensions.conf one.[/quote]

logger.conf on both servers has:

Still, on destination server there is no any full log, and on the caller server one line appears:

UPD: Changed verbosity to maximum and got this on caller server:

[Jun 27 05:16:27] WARNING[13663] chan_sip.c: Received response: "Forbidden" from '"WACaller" <sip:wacaller@192.168.242.125>;tag=as5911b9f9' [Jun 27 05:21:27] VERBOSE[13663] netsock2.c: == Using SIP RTP CoS mark 5 [Jun 27 05:21:27] VERBOSE[14529] pbx.c: -- Executing [123123123@watest:1] NoOp("SIP/wacaller-00000019", "") in new stack [Jun 27 05:21:27] VERBOSE[14529] pbx.c: -- Executing [123123123@watest:2] Dial("SIP/wacaller-00000019", "SIP/wagateway/123123123") in new stack [Jun 27 05:21:27] VERBOSE[14529] netsock2.c: == Using SIP RTP CoS mark 5 [Jun 27 05:21:27] VERBOSE[14529] app_dial.c: -- Called SIP/wagateway/123123123 [Jun 27 05:21:27] WARNING[13663] chan_sip.c: Received response: "Forbidden" from '"WACaller" <sip:wacaller@192.168.242.125>;tag=as7231f32a' [Jun 27 05:21:27] VERBOSE[14529] app_dial.c: == Everyone is busy/congested at this time (1:0/1/0) [Jun 27 05:21:27] VERBOSE[14529] pbx.c: -- Executing [123123123@watest:3] Hangup("SIP/wacaller-00000019", "") in new stack [Jun 27 05:21:27] VERBOSE[14529] pbx.c: == Spawn extension (watest, 123123123, 3) exited non-zero on 'SIP/wacaller-00000019'

By the way, changing caller server from one to another with fresh asterisk 1.8 didn’t help.

6

You need the detailed logs from the called machine.