Hi,
I have install a new Asterisk serveur on Ubuntu 10.4 LTS. My actually serveur is on Debian and have Asterisk 1.4.
I want to use te new server, I can do outgoing all, but when I call my number with an external phone, I have an hang up after 5 sec, and this 2 line in cli verbose mode 5 :
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
This a my log with debug mode on.
Any idea?
Thank's
Jean-Seb
Log debug on:
PBX*CLI> sip set debug on
SIP Debugging enabled
Really destroying SIP dialog '08219e58068dadd9556e1f3d6e4ea615@192.168.50.4' Method: REGISTER
== Manager 'admin' logged on from 127.0.0.1
== Manager 'admin' logged off from 127.0.0.1
== Manager 'admin' logged on from 127.0.0.1
== Manager 'admin' logged off from 127.0.0.1
PBX*CLI>
<--- SIP read from UDP:209.217.98.194:5060 --->
INVITE sip:[email protected].69 SIP/2.0
Via: SIP/2.0/UDP 209.217.98.194:5060;branch=z9hG4bK51fe72dc;rport
From: "Unknown Name" <sip:[email protected].194>;tag=as07d9c92c
To: <sip:@173.179.85.69>
Contact: <sip:[email protected]94>
Call-ID: 178cc86029e751936457fbea1b169b6a@209.217.9.194
CSeq: 102 INVITE
User-Agent: HMNet SIP 1.02
Max-Forwards: 70
Date: Thu, 22 Jul 2010 03:00:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 265
v=0
o=root 4654 4654 IN IP4 209.217.98.194
s=session
c=IN IP4 209.217.8.194
t=0 0
m=audio 17904 RTP/AVP 0 3 101
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
<------------->
--- (14 headers 13 lines) ---
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Sending to 209.217.98.194 : 5060 (NAT)
Using INVITE request as basis request - 178cc86029e751936457fbea1b169b6a@209.217.98.194
Found peer 'from-sip-external' for 'anonymous' from 209.217.98.194:5060
Found RTP audio format 0
Found RTP audio format 3
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format GSM for ID 3
Found audio description format telephone-event for ID 101
Capabilities: us - 0x4 (ulaw), peer - audio=0x6 (gsm|ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 209.217.9.194:17904
Looking for myphone in default (domain 153.179.85.69)
<--- Reliably Transmitting (NAT) to 209.217.98.194:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 209.217.98.194:5060;branch=z9hG4bK51fe72dc;received=209.217.98.194;rport=5060
From: "Unknown Name" <sip:[email protected].194>;tag=as07d9c92c
To: <sip:[email protected]>;tag=as41bae57e
Call-ID: 178cc86029e751936457fbea1b169b6a@209.217.98.194
CSeq: 102 INVITE
Server: Asterisk PBX 1.6.2.6
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '178cc86029e751936457fbea1b169b6a@209.217.98.194' in 6400 ms (Method: INVITE)
PBX*CLI>
<--- SIP read from UDP:209.217.98.194:5060 --->
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 209.217.98.194:5060;branch=z9hG4bK51fe72dc;rport
From: "Unknown Name" <sip:[email protected].194>;tag=as07d9c92c
To: <sip:[email protected].69>;tag=as41bae57e
Contact: <sip:[email protected].194>
Call-ID: 178cc86029e751936457fbea1b169b6a@209.217.98.194
CSeq: 102 ACK
User-Agent: HMNet SIP 1.02
Max-Forwards: 70
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '178cc86029e751936457fbea1b169b6a@209.217.98.194' Method: ACK
== Manager 'admin' logged on from 127.0.0.1
Reliably Transmitting (NAT) to 209.217.98.194:5060:
OPTIONS sip:sip02.unlimitel.ca SIP/2.0
Via: SIP/2.0/UDP 153.179.85.69:5060;branch=z9hG4bK0133b7b7;rport
Max-Forwards: 70
From: "Unknown" <sip:[email protected]>;tag=as4d82b8ef
To: <sip:sip02.unlimitel.ca>
Contact: <sip:[email protected]>
Call-ID: 6c2a61592081caee62e2384004af070a@173.179.85.69
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.6
Date: Thu, 22 Jul 2010 03:00:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
---
Reliably Transmitting (NAT) to 209.217.98.194:5060:
OPTIONS sip:sip02.unlimitel.ca SIP/2.0
Via: SIP/2.0/UDP 153.179.85.69:5060;branch=z9hG4bK1ae05670;rport
Max-Forwards: 70
From: "Unknown" <sip:[email protected]>;tag=as5c86a597
To: <sip:sip02.unlimitel.ca>
Contact: <sip:[email protected]>
Call-ID: 4c4a89253768a88d3f921ac56e1a7a50@153.179.85.69
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.6
Date: Thu, 22 Jul 2010 03:00:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
---
PBX*CLI>
<--- SIP read from UDP:209.217.98.194:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 153.179.85.69:5060;branch=z9hG4bK0133b7b7;received=153.179.85.69;rport=5060
From: "Unknown" <sip:[email protected]>;tag=as4d82b8ef
To: <sip:sip02.unlimitel.ca>;tag=as3996a98a
Call-ID: 6c2a61592081caee62e2384004af070a@173.179.85.69
CSeq: 102 OPTIONS
User-Agent: HMNet SIP 1.02
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Accept: application/sdp
Content-Length: 0
PBX*CLI>
<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '6c2a61592081caee62e2384004af070a@153.179.85.69' Method: OPTIONS
PBX*CLI>
<--- SIP read from UDP:209.217.98.194:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 153.179.85.69:5060;branch=z9hG4bK1ae05670;received=173.179.85.69;rport=5060
From: "Unknown" <sip:[email protected]>;tag=as5c86a597
To: <sip:sip02.unlimitel.ca>;tag=as39988e9b
Call-ID: 4c4a89253768a88d3f921ac56e1a7a50@153.179.85.69
CSeq: 102 OPTIONS
User-Agent: HMNet SIP 1.02
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Accept: application/sdp
Content-Length: 0
PBX*CLI>
<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '4c4a89253768a88d3f921ac56e1a7a50@173.179.85.69' Method: OPTIONS
Reliably Transmitting (NAT) to 192.168.50.138:37726:
OPTIONS sip:[email protected]:37726;rinstance=1ef7bc8373d31faa SIP/2.0
Via: SIP/2.0/UDP 192.168.50.4:5060;branch=z9hG4bK41a18401;rport
Max-Forwards: 70
From: "Unknown" <sip:[email protected]>;tag=as6ddb2797
To: <sip:[email protected]:37726;rinstance=1ef7bc8373d31faa>
Contact: <sip:[email protected]>
Call-ID: 0759c54c2c458c54710e64ac71ce9b82@192.168.50.4
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.6
Date: Thu, 22 Jul 2010 03:00:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
---
== Manager 'admin' logged off from 127.0.0.1
PBX*CLI>
<--- SIP read from UDP:192.168.50.138:37726 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.50.4:5060;branch=z9hG4bK41a18401;rport=5060
Contact: <sip:192.168.50.138:37726>
To: <sip:[email protected]:37726;rinstance=1ef7bc8373d31faa>;tag=e064a84f
From: "Unknown"<sip:Unknown@192.168.50.4>;tag=as6ddb2797
Call-ID: 0759c54c2c458c54710e64ac71ce9b82@192.168.50.4
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 0
<------------->
--- (12 headers 0 lines) ---
Really destroying SIP dialog '0759c54c2c458c54710e64ac71ce9b82@192.168.50.4' Method: OPTIONS
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