[SOLVED] Problem with call "handshaking" with a SIP provider

Hello guys,

There is one Russian SIP provider called comtube and I’m trying to adjust my settings in Asterisk. When I try to make a call, their server returns BYE. When I try to make a call from 3CX softphone, everything is OK. I have successful registration (register string in sip.conf) to comtube in Asterisk. I ran sip debug and here are the packets from both calls.

Generally speaking this is what happens:
Asterisk<->comtube:
Asterisk -> comtube: INVITE
Asterisk <- comtube: Giving a try
Asterisk <- comtube: OK
Asterisk -> comtube: ACK
Asterisk <- comtube: BYE

Softphone<->comtube:
Softphone -> comtube: INVITE
Softphone <- comtube: Proxy Authentication Required
Softphone -> comtube: ACK
Softphone -> comtube: INVITE (with proxy authorization)
Softphone <- comtube: Giving a try
Softphone <- comtube: Ringing
and so on…

This is the full SIP capture:

Asterisk to comtube:

[code]Reliably Transmitting (no NAT) to 85.192.44.73:5060:
INVITE sip:00359885103431@sip.comtube.com:5060 SIP/2.0
Via: SIP/2.0/UDP 85.91.139.50:5060;branch=z9hG4bK205ba9dc
Max-Forwards: 70
From: “103” sip:197241@85.91.139.50;tag=as45e16371
To: sip:00359885103431@sip.comtube.com:5060
Contact: sip:197241@85.91.139.50:5060
Call-ID: 59ac275d7b5bf983118d85cd2d56ffc7@85.91.139.50:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.23.0
Date: Fri, 13 Sep 2013 07:29:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Remote-Party-ID: “103” sip:103@85.91.139.50;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 383

v=0
o=root 1489791177 1489791177 IN IP4 85.91.139.50
s=Asterisk PBX 1.8.23.0
c=IN IP4 85.91.139.50
b=CT:384
t=0 0
m=audio 10066 RTP/AVP 8 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/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
m=video 10002 RTP/AVP 34
a=rtpmap:34 H263/90000
a=sendrecv


[2013-09-13 10:29:13] – Called SIP/comtube/00359885103431
[2013-09-13 10:29:13]
<— SIP read from UDP:85.192.44.73:5060 —>
SIP/2.0 100 Giving a try
Via: SIP/2.0/UDP 85.91.139.50:5060;received=77.71.39.149;rport=5060;branch=z9hG4bK205ba9dc
From: “103” sip:197241@85.91.139.50;tag=as45e16371
To: sip:00359885103431@sip.comtube.com:5060
Call-ID: 59ac275d7b5bf983118d85cd2d56ffc7@85.91.139.50:5060
CSeq: 102 INVITE
Server: Comtube SIP Proxy
Content-Length: 0

<------------->
[2013-09-13 10:29:13] — (8 headers 0 lines) —
[2013-09-13 10:29:13]
<— SIP read from UDP:85.192.44.73:5060 —>
SIP/2.0 200 OK
Content-Type:application/sdp
Contact:sip:00359885103431@192.168.6.11:5061;nat=yes
Allow:INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,UPDATE,PRACK,REFER,SUBSCRIBE,NOTIFY
Supported:100rel
Accept:application/sdp
From:sip:197241@85.91.139.50;tag=as45e16371
To:sip:00359885103431@sip.comtube.com:5060;tag=288E303035303932004DB125
Call-ID:59ac275d7b5bf983118d85cd2d56ffc7@85.91.139.50:5060
CSeq:102 INVITE
Record-Route:sip:85.192.44.73;lr;ftag=as45e16371;did=888.48574c71
Server:TB005092/2.1
Via:SIP/2.0/UDP 85.91.139.50:5060;rport=5060;received=77.71.39.149;branch=z9hG4bK205ba9dc
Content-Length: 155

v=0
o=tb640 5 1 IN IP4 85.192.44.73
s=-
c=IN IP4 85.192.44.73
t=0 0
m=audio 26376 RTP/AVP 8 101
a=rtpmap:101 telephone-event/8000
a=nortpproxy:yes
<------------->
[2013-09-13 10:29:13] — (14 headers 8 lines) —
[2013-09-13 10:29:13] Found RTP audio format 8
[2013-09-13 10:29:13] Found RTP audio format 101
[2013-09-13 10:29:13] Found audio description format telephone-event for ID 101
[2013-09-13 10:29:13] Capabilities: us - 0x28000f (g723|gsm|ulaw|alaw|h263|h264), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
[2013-09-13 10:29:13] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[2013-09-13 10:29:13] Peer audio RTP is at port 85.192.44.73:26376
[2013-09-13 10:29:13] Peer doesn’t provide video
[2013-09-13 10:29:13] list_route: hop: sip:85.192.44.73;lr;ftag=as45e16371;did=888.48574c71
[2013-09-13 10:29:13] Transmitting (no NAT) to 85.192.44.73:5060:
ACK sip:00359885103431@192.168.6.11:5061;nat=yes SIP/2.0
Via: SIP/2.0/UDP 85.91.139.50:5060;branch=z9hG4bK15fb7666
Route: sip:85.192.44.73;lr;ftag=as45e16371;did=888.48574c71
Max-Forwards: 70
From: “103” sip:197241@85.91.139.50;tag=as45e16371
To: sip:00359885103431@sip.comtube.com:5060;tag=288E303035303932004DB125
Contact: sip:197241@85.91.139.50:5060
Call-ID: 59ac275d7b5bf983118d85cd2d56ffc7@85.91.139.50:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.23.0
Content-Length: 0


[2013-09-13 10:29:13] – SIP/comtube-00000010 answered SIP/103-0000000f
[2013-09-13 10:29:14]
<— SIP read from UDP:85.192.44.73:5060 —>
BYE sip:197241@192.168.1.5:5060 SIP/2.0
Record-Route: sip:85.192.44.73;lr;ftag=288E303035303932004DB125
Date:Thu, 06 Jan 2000 20:01:16 GMT
To:sip:197241@85.91.139.50;tag=as45e16371
From:sip:00359885103431@sip.comtube.com:5060;tag=288E303035303932004DB125
Call-ID:59ac275d7b5bf983118d85cd2d56ffc7@85.91.139.50:5060
CSeq:1 BYE
Max-Forwards:69
Timestamp:5091633
Via: SIP/2.0/UDP 85.192.44.73:5060;branch=z9hG4bKfb8c.287b013.0
Via:SIP/2.0/UDP 192.168.6.11:5061;received=192.168.6.11;branch=z9hG4bK4CF541B0E7E3496B9DBACE145DAFE0F5;rport=5061
Content-Length:0
P-hint: rr-enforced

<------------->
[2013-09-13 10:29:14] — (13 headers 0 lines) —
[2013-09-13 10:29:14] Sending to 85.192.44.73:5060 (no NAT)
[2013-09-13 10:29:14] Scheduling destruction of SIP dialog ‘59ac275d7b5bf983118d85cd2d56ffc7@85.91.139.50:5060’ in 6400 ms (Method: BYE)
[2013-09-13 10:29:14]
<— Transmitting (no NAT) to 85.192.44.73:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 85.192.44.73:5060;branch=z9hG4bKfb8c.287b013.0;received=85.192.44.73
Via: SIP/2.0/UDP 192.168.6.11:5061;received=192.168.6.11;branch=z9hG4bK4CF541B0E7E3496B9DBACE145DAFE0F5;rport=5061
Record-Route: sip:85.192.44.73;lr;ftag=288E303035303932004DB125
From: sip:00359885103431@sip.comtube.com:5060;tag=288E303035303932004DB125
To: sip:197241@85.91.139.50;tag=as45e16371
Call-ID: 59ac275d7b5bf983118d85cd2d56ffc7@85.91.139.50:5060
CSeq: 1 BYE
Server: Asterisk PBX 1.8.23.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------>
[/code]

Softphone to comtube (captured from Wireshark):

[code]Frame 1: 1053 bytes on wire (8424 bits), 1053 bytes captured (8424 bits) on interface 0
Ethernet II, Src: AsrockIn_d2:f7:d1 (00:25:22:d2:f7:d1), Dst: Routerbo_26:de:4b (d4:ca:6d:26:de:4b)
Internet Protocol Version 4, Src: 192.168.1.87 (192.168.1.87), Dst: 85.192.44.73 (85.192.44.73)
User Datagram Protocol, Src Port: taskman-port (2470), Dst Port: sip (5060)
Session Initiation Protocol (INVITE)
Request-Line: INVITE sip:00359885103431@sip.comtube.com:5060 SIP/2.0
Message Header
Via: SIP/2.0/UDP 192.168.1.87:2470;branch=z9hG4bK-d8754z-5b3b22190f137436-1—d8754z-;rport
Max-Forwards: 70
Contact: sip:197241@77.71.39.149:2470;transport=UDP
To: sip:00359885103431@sip.comtube.com:5060
From: "Stanimir"sip:197241@sip.comtube.com:5060;tag=0c7d710c
Call-ID: ZGJiOGJkYjYwYjRlMDNhZjg1ODc2YTMxYWI1MzU3NTI.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Content-Type: application/sdp
Supported: replaces
User-Agent: 3CXPhone 6.0.19548.0
Content-Length: 405
Message Body
Session Description Protocol

Frame 2: 587 bytes on wire (4696 bits), 587 bytes captured (4696 bits) on interface 0
Ethernet II, Src: Routerbo_26:de:4b (d4:ca:6d:26:de:4b), Dst: AsrockIn_d2:f7:d1 (00:25:22:d2:f7:d1)
Internet Protocol Version 4, Src: 85.192.44.73 (85.192.44.73), Dst: 192.168.1.87 (192.168.1.87)
User Datagram Protocol, Src Port: sip (5060), Dst Port: taskman-port (2470)
Session Initiation Protocol (407)
Status-Line: SIP/2.0 407 Proxy Authentication Required
Message Header
Via: SIP/2.0/UDP 192.168.1.87:2470;received=77.71.39.149;branch=z9hG4bK-d8754z-5b3b22190f137436-1—d8754z-;rport=2470
To: sip:00359885103431@sip.comtube.com:5060;tag=9766e6cd6366e4381d550735b7db114a.4a6d
From: "Stanimir"sip:197241@sip.comtube.com:5060;tag=0c7d710c
Call-ID: ZGJiOGJkYjYwYjRlMDNhZjg1ODc2YTMxYWI1MzU3NTI.
CSeq: 1 INVITE
Proxy-Authenticate: Digest realm=“sip.comtube.com”, nonce="5232bcac00001f0475be5b5441755e7fc0a3708ba16141fd"
Server: Comtube SIP Proxy
Content-Length: 0

Frame 3: 448 bytes on wire (3584 bits), 448 bytes captured (3584 bits) on interface 0
Ethernet II, Src: AsrockIn_d2:f7:d1 (00:25:22:d2:f7:d1), Dst: Routerbo_26:de:4b (d4:ca:6d:26:de:4b)
Internet Protocol Version 4, Src: 192.168.1.87 (192.168.1.87), Dst: 85.192.44.73 (85.192.44.73)
User Datagram Protocol, Src Port: taskman-port (2470), Dst Port: sip (5060)
Session Initiation Protocol (ACK)
Request-Line: ACK sip:00359885103431@sip.comtube.com:5060 SIP/2.0
Message Header
Via: SIP/2.0/UDP 192.168.1.87:2470;branch=z9hG4bK-d8754z-5b3b22190f137436-1—d8754z-;rport
Max-Forwards: 70
To: sip:00359885103431@sip.comtube.com:5060;tag=9766e6cd6366e4381d550735b7db114a.4a6d
From: "Stanimir"sip:197241@sip.comtube.com:5060;tag=0c7d710c
Call-ID: ZGJiOGJkYjYwYjRlMDNhZjg1ODc2YTMxYWI1MzU3NTI.
CSeq: 1 ACK
Content-Length: 0

Frame 4: 1285 bytes on wire (10280 bits), 1285 bytes captured (10280 bits) on interface 0
Ethernet II, Src: AsrockIn_d2:f7:d1 (00:25:22:d2:f7:d1), Dst: Routerbo_26:de:4b (d4:ca:6d:26:de:4b)
Internet Protocol Version 4, Src: 192.168.1.87 (192.168.1.87), Dst: 85.192.44.73 (85.192.44.73)
User Datagram Protocol, Src Port: taskman-port (2470), Dst Port: sip (5060)
Session Initiation Protocol (INVITE)
Request-Line: INVITE sip:00359885103431@sip.comtube.com:5060 SIP/2.0
Message Header
Via: SIP/2.0/UDP 192.168.1.87:2470;branch=z9hG4bK-d8754z-ba5bf9791c68ee0c-1—d8754z-;rport
Max-Forwards: 70
Contact: sip:197241@77.71.39.149:2470;transport=UDP
To: sip:00359885103431@sip.comtube.com:5060
From: "Stanimir"sip:197241@sip.comtube.com:5060;tag=0c7d710c
Call-ID: ZGJiOGJkYjYwYjRlMDNhZjg1ODc2YTMxYWI1MzU3NTI.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
Content-Type: application/sdp
Proxy-Authorization: Digest username=“197241”,realm=“sip.comtube.com”,nonce=“5232bcac00001f0475be5b5441755e7fc0a3708ba16141fd”,uri=“sip:00359885103431@sip.comtube.com:5060”,response=“e0c8cf3c449f2d21165afb3256f85eeb”,algorithm=MD5
Supported: replaces
User-Agent: 3CXPhone 6.0.19548.0
Content-Length: 405
Message Body
Session Description Protocol

Frame 5: 418 bytes on wire (3344 bits), 418 bytes captured (3344 bits) on interface 0
Ethernet II, Src: Routerbo_26:de:4b (d4:ca:6d:26:de:4b), Dst: AsrockIn_d2:f7:d1 (00:25:22:d2:f7:d1)
Internet Protocol Version 4, Src: 85.192.44.73 (85.192.44.73), Dst: 192.168.1.87 (192.168.1.87)
User Datagram Protocol, Src Port: sip (5060), Dst Port: taskman-port (2470)
Session Initiation Protocol (100)
Status-Line: SIP/2.0 100 Giving a try
Message Header
Via: SIP/2.0/UDP 192.168.1.87:2470;received=77.71.39.149;branch=z9hG4bK-d8754z-ba5bf9791c68ee0c-1—d8754z-;rport=2470
To: sip:00359885103431@sip.comtube.com:5060
From: "Stanimir"sip:197241@sip.comtube.com:5060;tag=0c7d710c
Call-ID: ZGJiOGJkYjYwYjRlMDNhZjg1ODc2YTMxYWI1MzU3NTI.
CSeq: 2 INVITE
Server: Comtube SIP Proxy
Content-Length: 0

Frame 6: 807 bytes on wire (6456 bits), 807 bytes captured (6456 bits) on interface 0
Ethernet II, Src: Routerbo_26:de:4b (d4:ca:6d:26:de:4b), Dst: AsrockIn_d2:f7:d1 (00:25:22:d2:f7:d1)
Internet Protocol Version 4, Src: 85.192.44.73 (85.192.44.73), Dst: 192.168.1.87 (192.168.1.87)
User Datagram Protocol, Src Port: sip (5060), Dst Port: taskman-port (2470)
Session Initiation Protocol (180)
Status-Line: SIP/2.0 180 Ringing
Message Header
Content-Type:application/sdp
Allow:INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,UPDATE,PRACK,REFER,SUBSCRIBE,NOTIFY
Date:Thu, 06 Jan 2000 19:51:50 GMT
From:"Stanimir"sip:197241@sip.comtube.com:5060;tag=0c7d710c
To:sip:00359885103431@sip.comtube.com:5060;tag=4C7F303035303932004D9AA9
Call-ID:ZGJiOGJkYjYwYjRlMDNhZjg1ODc2YTMxYWI1MzU3NTI.
CSeq:2 INVITE
Record-Route:sip:85.192.44.73;lr;ftag=0c7d710c;did=d7c.b59d894
Server:TB005092/2.1
Via:SIP/2.0/UDP 192.168.1.87:2470;received=77.71.39.149;branch=z9hG4bK-d8754z-ba5bf9791c68ee0c-1—d8754z-;rport=2470
Content-Length: 155
Message Body
Session Description Protocol

Frame 7: 678 bytes on wire (5424 bits), 678 bytes captured (5424 bits) on interface 0
Ethernet II, Src: AsrockIn_d2:f7:d1 (00:25:22:d2:f7:d1), Dst: Routerbo_26:de:4b (d4:ca:6d:26:de:4b)
Internet Protocol Version 4, Src: 192.168.1.87 (192.168.1.87), Dst: 85.192.44.73 (85.192.44.73)
User Datagram Protocol, Src Port: taskman-port (2470), Dst Port: sip (5060)
Session Initiation Protocol (CANCEL)
Request-Line: CANCEL sip:00359885103431@sip.comtube.com:5060 SIP/2.0
Message Header
Via: SIP/2.0/UDP 192.168.1.87:2470;branch=z9hG4bK-d8754z-ba5bf9791c68ee0c-1—d8754z-;rport
Max-Forwards: 70
To: sip:00359885103431@sip.comtube.com:5060
From: "Stanimir"sip:197241@sip.comtube.com:5060;tag=0c7d710c
Call-ID: ZGJiOGJkYjYwYjRlMDNhZjg1ODc2YTMxYWI1MzU3NTI.
CSeq: 2 CANCEL
Proxy-Authorization: Digest username=“197241”,realm=“sip.comtube.com”,nonce=“5232bcac00001f0475be5b5441755e7fc0a3708ba16141fd”,uri=“sip:00359885103431@sip.comtube.com:5060”,response=“6d0147a5cbf42289e29cce8632e5f000”,algorithm=MD5
User-Agent: 3CXPhone 6.0.19548.0
Content-Length: 0

Frame 8: 457 bytes on wire (3656 bits), 457 bytes captured (3656 bits) on interface 0
Ethernet II, Src: Routerbo_26:de:4b (d4:ca:6d:26:de:4b), Dst: AsrockIn_d2:f7:d1 (00:25:22:d2:f7:d1)
Internet Protocol Version 4, Src: 85.192.44.73 (85.192.44.73), Dst: 192.168.1.87 (192.168.1.87)
User Datagram Protocol, Src Port: sip (5060), Dst Port: taskman-port (2470)
Session Initiation Protocol (200)
Status-Line: SIP/2.0 200 canceling
Message Header
Via: SIP/2.0/UDP 192.168.1.87:2470;received=77.71.39.149;branch=z9hG4bK-d8754z-ba5bf9791c68ee0c-1—d8754z-;rport=2470
To: sip:00359885103431@sip.comtube.com:5060;tag=2c188135ad61a91c40572184412e3209-72a5
From: "Stanimir"sip:197241@sip.comtube.com:5060;tag=0c7d710c
Call-ID: ZGJiOGJkYjYwYjRlMDNhZjg1ODc2YTMxYWI1MzU3NTI.
CSeq: 2 CANCEL
Server: Comtube SIP Proxy
Content-Length: 0

Frame 9: 610 bytes on wire (4880 bits), 610 bytes captured (4880 bits) on interface 0
Ethernet II, Src: Routerbo_26:de:4b (d4:ca:6d:26:de:4b), Dst: AsrockIn_d2:f7:d1 (00:25:22:d2:f7:d1)
Internet Protocol Version 4, Src: 85.192.44.73 (85.192.44.73), Dst: 192.168.1.87 (192.168.1.87)
User Datagram Protocol, Src Port: sip (5060), Dst Port: taskman-port (2470)
Session Initiation Protocol (487)
Status-Line: SIP/2.0 487 Request Terminated
Message Header
Content-Type:application/sdp
From:"Stanimir"sip:197241@sip.comtube.com:5060;tag=0c7d710c
To:sip:00359885103431@sip.comtube.com:5060;tag=4C7F303035303932004D9AA9
Call-ID:ZGJiOGJkYjYwYjRlMDNhZjg1ODc2YTMxYWI1MzU3NTI.
CSeq:2 INVITE
Server:TB005092/2.1
Via:SIP/2.0/UDP 192.168.1.87:2470;received=77.71.39.149;branch=z9hG4bK-d8754z-ba5bf9791c68ee0c-1—d8754z-;rport=2470
Content-Length:137
Message Body
Session Description Protocol

Frame 10: 435 bytes on wire (3480 bits), 435 bytes captured (3480 bits) on interface 0
Ethernet II, Src: AsrockIn_d2:f7:d1 (00:25:22:d2:f7:d1), Dst: Routerbo_26:de:4b (d4:ca:6d:26:de:4b)
Internet Protocol Version 4, Src: 192.168.1.87 (192.168.1.87), Dst: 85.192.44.73 (85.192.44.73)
User Datagram Protocol, Src Port: taskman-port (2470), Dst Port: sip (5060)
Session Initiation Protocol (ACK)
Request-Line: ACK sip:00359885103431@sip.comtube.com:5060 SIP/2.0
Message Header
Via: SIP/2.0/UDP 192.168.1.87:2470;branch=z9hG4bK-d8754z-ba5bf9791c68ee0c-1—d8754z-;rport
Max-Forwards: 70
To: sip:00359885103431@sip.comtube.com:5060;tag=4C7F303035303932004D9AA9
From: "Stanimir"sip:197241@sip.comtube.com:5060;tag=0c7d710c
Call-ID: ZGJiOGJkYjYwYjRlMDNhZjg1ODc2YTMxYWI1MzU3NTI.
CSeq: 2 ACK
Content-Length: 0

Frame 11: 46 bytes on wire (368 bits), 46 bytes captured (368 bits) on interface 0
Ethernet II, Src: AsrockIn_d2:f7:d1 (00:25:22:d2:f7:d1), Dst: Routerbo_26:de:4b (d4:ca:6d:26:de:4b)
Internet Protocol Version 4, Src: 192.168.1.87 (192.168.1.87), Dst: 85.192.44.73 (85.192.44.73)
User Datagram Protocol, Src Port: taskman-port (2470), Dst Port: sip (5060)
Data (4 bytes)

0000 0d 0a 0d 0a …
[/code]

pbx*CLI> sip show peer comtube

[code] * Name : comtube
Secret :
MD5Secret :
Remote Secret:
Context : from-external
Subscr.Cont. :
Language :
AMA flags : Unknown
Transfer mode: open
CallingPres : Presentation Allowed, Not Screened
FromUser : 197241
Callgroup : 1
Pickupgroup : 1
MOH Suggest : default
Mailbox :
VM Extension : asterisk
LastMsgsSent : 32767/65535
Call limit : 5
Max forwards : 0
Dynamic : No
Callerid : “” <>
MaxCallBR : 384 kbps
Expire : -1
Insecure : invite
Force rport : No
ACL : No
DirectMedACL : No
T.38 support : Yes
T.38 EC mode : FEC
T.38 MaxDtgrm: -1
DirectMedia : No
PromiscRedir : No
User=Phone : No
Video Support: Yes
Text Support : No
Ign SDP ver : No
Trust RPID : Yes
Send RPID : Yes
Subscriptions: Yes
Overlap dial : Yes
DTMFmode : auto
Timer T1 : 500
Timer B : 32000
ToHost : sip.comtube.com
Addr->IP : 85.192.44.73:5060
Defaddr->IP : (null)
Prim.Transp. : UDP
Allowed.Trsp : UDP
Def. Username: 197241
SIP Options : (none)
Codecs : 0x28000f (g723|gsm|ulaw|alaw|h263|h264)
Codec Order : (alaw:20,ulaw:20,gsm:20,g723:30)
Auto-Framing : No
Status : OK (81 ms)
Useragent :
Reg. Contact :
Qualify Freq : 30000 ms
Sess-Timers : Accept
Sess-Refresh : uas
Sess-Expires : 1800 secs
Min-Sess : 90 secs
RTP Engine : asterisk
Parkinglot :
Use Reason : No
Encryption : No

[/code]

pbx*CLI> sip show settings

[code]Global Settings:

UDP Bindaddress: 0.0.0.0:5060
TCP SIP Bindaddress: Disabled
TLS SIP Bindaddress: Disabled
Videosupport: Yes
Textsupport: No
Ignore SDP sess. ver.: No
AutoCreate Peer: No
Match Auth Username: No
Allow unknown access: No
Allow subscriptions: Yes
Allow overlap dialing: Yes
Allow promisc. redir: No
Enable call counters: No
SIP domain support: No
Realm. auth: No
Our auth realm asterisk
Use domains as realms: No
Call to non-local dom.: Yes
URI user is phone no: No
Always auth rejects: Yes
Direct RTP setup: No
User Agent: Asterisk PBX 1.8.23.0
SDP Session Name: Asterisk PBX 1.8.23.0
SDP Owner Name: root
Reg. context: (not set)
Regexten on Qualify: No
Legacy userfield parse: No
Caller ID: Asterisk
From: Domain:
Record SIP history: Off
Call Events: Off
Auth. Failure Events: Off
T.38 support: Yes
T.38 EC mode: FEC
T.38 MaxDtgrm: -1
SIP realtime: Disabled
Qualify Freq : 10000 ms
Q.850 Reason header: No
Store SIP_CAUSE: No

Network QoS Settings:

IP ToS SIP: CS3
IP ToS RTP audio: EF
IP ToS RTP video: AF41
IP ToS RTP text: CS0
802.1p CoS SIP: 4
802.1p CoS RTP audio: 5
802.1p CoS RTP video: 6
802.1p CoS RTP text: 5
Jitterbuffer enabled: No

Network Settings:

SIP address remapping: Enabled using externaddr
Externhost:
Externaddr: 85.91.139.50:0
Externrefresh: 10
Localnet: 192.168.1.0/255.255.255.0

Global Signalling Settings:

Codecs: 0x0 (nothing)
Codec Order: none
Relax DTMF: No
RFC2833 Compensation: Yes
Symmetric RTP: No
Compact SIP headers: No
RTP Keepalive: 0 (Disabled)
RTP Timeout: 300
RTP Hold Timeout: 300
MWI NOTIFY mime type: application/simple-message-summary
DNS SRV lookup: Yes
Pedantic SIP support: Yes
Reg. min duration 60 secs
Reg. max duration: 3600 secs
Reg. default duration: 120 secs
Outbound reg. timeout: 20 secs
Outbound reg. attempts: 0
Notify ringing state: Yes
Include CID: No
Notify hold state: No
SIP Transfer mode: open
Max Call Bitrate: 384 kbps
Auto-Framing: No
Outb. proxy:
Session Timers: Accept
Session Refresher: uas
Session Expires: 1800 secs
Session Min-SE: 90 secs
Timer T1: 500
Timer T1 minimum: 100
Timer B: 32000
No premature media: Yes
Max forwards: 70

Default Settings:

Allowed transports: UDP
Outbound transport: UDP
Context: default
Force rport: No
DTMF: auto
Qualify: 10000
Use ClientCode: No
Progress inband: Never
Language:
MOH Interpret: default
MOH Suggest: default
Voice Mail Extension: asterisk

----[/code]

I wanted to post in the comtube’s forum but I don’t have registration and their registration form is buggy, so here is my only option to search for help. What am I missing?

They appear to have decided to abort the call before sending OK (otherwise they would have sent Ringing). That is stupid as you should abort the call at that point with a 4xx, 5xx, or 6xx response, which will give a clue as to the reason. (In any case, there is nothing in the ACK that could cause any reasonable system to change its mind.

You need to sort out your support registration problem or use an ITSP that does support their users.

(The fact that they don’t request authentication either makes me think that this is an attempt to confuse an attacker and they are treating you as hostile for some reason, possibly because you are not using an approved user agent. Have you successfully registered?)

Hi David,

I don’t know if this does make any sense but I managed to fix the issue. I had to put fromdomain field in my sip.conf for this account. After that I was able to make calls. Thank you for your time!