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
<------------>
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 ....
pbx*CLI> sip show peer comtube
Code:
* Name : comtube
Secret : <Set>
MD5Secret : <Not set>
Remote Secret: <Not set>
Context : from-external
Subscr.Cont. : <Not set>
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
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: <none>
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: <not set>
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
----
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?
Statistics : Posted by stanimir_ss • on Fri Sep 13, 2013 2:12 am • Replies 0 • Views 13