Hello!
I am using Asterisk 12.0, softphone-softphone calls are finally working, but, when calling IP Phone, I got the error below
chan_sip.c:10865 process_sdp_o: SDP syntax error in o= line version == Everyone is busy/congested at this time (1:0/0/1)
What am I missing? what is the difference between softphone and IP phone configurations?
Thanks!
softphone is a subset of IPPhone
The, unnamed, softphone is probably broken, but without the content of the SDP it is offering (e.g. sip set debug on) we can’t say how.
Sometimes disabling pedantic SIP processing can work round broken peers, but I’m not sure that it does anything for broken SDP.
Thank you for your reply!
Retransmitting #10 (NAT) to xxx.x.xxx.xx:5060:
NOTIFY sip:user@xxx.x.xxx.xx SIP/2.0
Via: SIP/2.0/UDP AstServerIP:5060;branch=z9hG4bK024db9ab;rport
Max-Forwards: 70
Route: sip:user@xxx.x.xxx.xx
From: “asterisk” sip:asterisk@AstServerIP;tag=as0609715b
To: sip:user@xxx.x.xxx.xx;tag=ca482779
Contact: sip:asterisk@AstServerIP:5060
Call-ID: MmM0ZDQ0Yjg0NTkxNWI2MmFlMTJhOTRiYzdhMjRjYjA
CSeq: 156 NOTIFY
User-Agent: Asterisk PBX 12.8.0-rc2
Event: message-summary
Content-Type: application/simple-message-summary
Subscription-State: active
Content-Length: 96
Messages-Waiting: yes
Message-Account: sip:asterisk@AstServerIP
Voice-Message: 23/0 (0/0)
<— SIP read from UDP:xxx.x.xxx.xx:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP AstServerIP:5060;branch=z9hG4bK024db9ab;rport
From: “asterisk” sip:asterisk@AstServerIP;tag=as0609715b
To: sip:user@xxx.x.xxx.xx;tag=ca482779
Call-ID: MmM0ZDQ0Yjg0NTkxNWI2MmFlMTJhOTRiYzdhMjRjYjA
CSeq: 156 NOTIFY
Contact: sip:user@xxx.x.xxx.xx
user-agent: X-Lite release 4.7.1 stamp 74247
Content-Length: 0
<------------->
— (9 headers 0 lines) —
Retransmitting #4 (NAT) to xxx.x.xxx.xx:5060:
NOTIFY sip:user@xxx.x.xxx.xx SIP/2.0
Via: SIP/2.0/UDP AstServerIP:5060;branch=z9hG4bK024db9ab;rport
Max-Forwards: 70
Route: sip:user@xxx.x.xxx.xx
From: “asterisk” sip:asterisk@AstServerIP;tag=as0609715b
To: sip:user@xxx.x.xxx.xx;tag=ca482779
Contact: sip:asterisk@AstServerIP:5060
Call-ID: MmM0ZDQ0Yjg0NTkxNWI2MmFlMTJhOTRiYzdhMjRjYjA
CSeq: 157 NOTIFY
User-Agent: Asterisk PBX 12.8.0-rc2
Event: message-summary
Content-Type: application/simple-message-summary
Subscription-State: active
Content-Length: 96
Messages-Waiting: yes
Message-Account: sip:asterisk@AstServerIP
Voice-Message: 23/0 (0/0)
Retransmitting #5 (NAT) to xxx.x.xxx.xx:5060:
NOTIFY sip:user@xxx.x.xxx.xx SIP/2.0
Via: SIP/2.0/UDP AstServerIP:5060;branch=z9hG4bK024db9ab;rport
Max-Forwards: 70
Route: sip:user@xxx.x.xxx.xx
From: “asterisk” sip:asterisk@AstServerIP;tag=as0609715b
To: sip:user@xxx.x.xxx.xx;tag=ca482779
Contact: sip:asterisk@AstServerIP:5060
Call-ID: MmM0ZDQ0Yjg0NTkxNWI2MmFlMTJhOTRiYzdhMjRjYjA
CSeq: 157 NOTIFY
User-Agent: Asterisk PBX 12.8.0-rc2
Event: message-summary
Content-Type: application/simple-message-summary
Subscription-State: active
Content-Length: 96
Messages-Waiting: yes
Message-Account: sip:asterisk@AstServerIP
Voice-Message: 23/0 (0/0)
<— SIP read from UDP:xxx.x.xxx.xx:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP AstServerIP:5060;branch=z9hG4bK024db9ab;rport
From: “asterisk” sip:asterisk@AstServerIP;tag=as0609715b
To: sip:user@xxx.x.xxx.xx;tag=ca482779
Call-ID: MmM0ZDQ0Yjg0NTkxNWI2MmFlMTJhOTRiYzdhMjRjYjA
CSeq: 156 NOTIFY
Contact: sip:user@xxx.x.xxx.xx
user-agent: X-Lite release 4.7.1 stamp 74247
Content-Length: 0
<------------->
— (9 headers 0 lines) —
Retransmitting #6 (NAT) to xxx.x.xxx.xx:5060:
NOTIFY sip:user@xxx.x.xxx.xx SIP/2.0
Via: SIP/2.0/UDP AstServerIP:5060;branch=z9hG4bK024db9ab;rport
Max-Forwards: 70
Route: sip:user@xxx.x.xxx.xx
From: “asterisk” sip:asterisk@AstServerIP;tag=as0609715b
To: sip:user@xxx.x.xxx.xx;tag=ca482779
Contact: sip:asterisk@AstServerIP:5060
Call-ID: MmM0ZDQ0Yjg0NTkxNWI2MmFlMTJhOTRiYzdhMjRjYjA
CSeq: 157 NOTIFY
User-Agent: Asterisk PBX 12.8.0-rc2
Event: message-summary
Content-Type: application/simple-message-summary
Subscription-State: active
Content-Length: 96
Messages-Waiting: yes
Message-Account: sip:asterisk@AstServerIP
Voice-Message: 23/0 (0/0)
<— SIP read from UDP:xxx.x.xxx.xx:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP AstServerIP:5060;branch=z9hG4bK024db9ab;rport
From: “asterisk” sip:asterisk@AstServerIP;tag=as0609715b
To: sip:user@xxx.x.xxx.xx;tag=ca482779
Call-ID: MmM0ZDQ0Yjg0NTkxNWI2MmFlMTJhOTRiYzdhMjRjYjA
CSeq: 156 NOTIFY
Contact: sip:user@xxx.x.xxx.xx
user-agent: X-Lite release 4.7.1 stamp 74247
Content-Length: 0
Retransmitting #7 (NAT) to xxx.x.xxx.xx:5060:
NOTIFY sip:user@xxx.x.xxx.xx SIP/2.0
Via: SIP/2.0/UDP AstServerIP:5060;branch=z9hG4bK024db9ab;rport
Max-Forwards: 70
Route: sip:user@xxx.x.xxx.xx
From: “asterisk” sip:asterisk@AstServerIP;tag=as0609715b
To: sip:user@xxx.x.xxx.xx;tag=ca482779
Contact: sip:asterisk@AstServerIP:5060
Call-ID: MmM0ZDQ0Yjg0NTkxNWI2MmFlMTJhOTRiYzdhMjRjYjA
CSeq: 157 NOTIFY
User-Agent: Asterisk PBX 12.8.0-rc2
Event: message-summary
Content-Type: application/simple-message-summary
Subscription-State: active
Content-Length: 96
Messages-Waiting: yes
Message-Account: sip:asterisk@AstServerIP
Voice-Message: 23/0 (0/0)
<— SIP read from UDP:xxx.x.xxx.xx:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP AstServerIP:5060;branch=z9hG4bK024db9ab;rport
From: “asterisk” sip:asterisk@AstServerIP;tag=as0609715b
To: sip:user@xxx.x.xxx.xx;tag=ca482779
Call-ID: MmM0ZDQ0Yjg0NTkxNWI2MmFlMTJhOTRiYzdhMjRjYjA
CSeq: 156 NOTIFY
Contact: sip:user@xxx.x.xxx.xx
user-agent: X-Lite release 4.7.1 stamp 74247
Content-Length: 0
<------------->
— (9 headers 0 lines) —
<— SIP read from UDP:xxx.x.xxx.xx:5060 —>
INVITE sip:100@AstServerIP SIP/2.0
Via: SIP/2.0/UDP xxx.x.xxx.xx:5060;branch=z9hG4bK7637ba6c28882398f6907a8f57d7e851
Via: SIP/2.0/UDP 192.168.1.2:37518;branch=z9hG4bK-d8754z-4508337cd837255e-1—d8754z-;rport
From: “user” sip:user@AstServerIP;tag=91c0881e
To: sip:100@AstServerIP
Call-ID: NGNiMTFmZGZiZDM2ODk5NWIzN2NjYjBjYzg0NWNkMTk
CSeq: 1 INVITE
Contact: sip:user@xxx.x.xxx.xx
max-forwards: 69
supported: replaces
user-agent: X-Lite 4.7.1 74247-bf7d9a67-W6.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Content-Length: 300
v=0
o=- 13067379963294747 1 IN IP4 xxx.x.xxx.xx
s=X-Lite release 4.7.1 stamp 74247
c=IN IP4 xxx.x.xxx.xx
t=0 0
m=audio 7070 RTP/AVP 125 100 0 9 8 101
a=rtpmap:125 opus/48000/2
a=fmtp:125 useinbandfec=1
a=rtpmap:100 speex/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
— (14 headers 12 lines) —
Sending to xxx.x.xxx.xx:5060 (NAT)
Sending to xxx.x.xxx.xx:5060 (NAT)
Using INVITE request as basis request - NGNiMTFmZGZiZDM2ODk5NWIzN2NjYjBjYzg0NWNkMTk
Found peer ‘user’ for ‘user’ from xxx.x.xxx.xx:5060
<— Reliably Transmitting (NAT) to xxx.x.xxx.xx:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP xxx.x.xxx.xx:5060;branch=z9hG4bK7637ba6c28882398f6907a8f57d7e851;received=xxx.x.xxx.xx;rport=5060
Via: SIP/2.0/UDP 192.168.1.2:37518;branch=z9hG4bK-d8754z-4508337cd837255e-1—d8754z-;rport
From: “user” sip:user@AstServerIP;tag=91c0881e
To: sip:100@AstServerIP;tag=as72a1f10c
Call-ID: NGNiMTFmZGZiZDM2ODk5NWIzN2NjYjBjYzg0NWNkMTk
CSeq: 1 INVITE
Server: Asterisk PBX 12.8.0-rc2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="6b6095f3"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘NGNiMTFmZGZiZDM2ODk5NWIzN2NjYjBjYzg0NWNkMTk’ in 32000 ms (Method: INVITE)
[Feb 2 13:40:18] WARNING[28990]: chan_sip.c:4103 retrans_pkt: Retransmission timeout reached on transmission 6931bfb7503171bd330037fe97c8fb2a for seqno 1 (Critical Response) – See wiki.asterisk.org/wiki/display/ … nsmissions
Packet timed out after 32000ms with no response
Really destroying SIP dialog ‘6931bfb7503171bd330037fe97c8fb2a’ Method: INVITE
<— SIP read from UDP:xxx.x.xxx.xx:5060 —>
ACK sip:100@AstServerIP SIP/2.0
Via: SIP/2.0/UDP xxx.x.xxx.xx:5060;branch=z9hG4bK7637ba6c28882398f6907a8f57d7e851
Via: SIP/2.0/UDP 192.168.1.2:37518;branch=z9hG4bK-d8754z-4508337cd837255e-1—d8754z-;rport
From: “user” sip:user@AstServerIP;tag=91c0881e
To: sip:100@AstServerIP;tag=as72a1f10c
Call-ID: NGNiMTFmZGZiZDM2ODk5NWIzN2NjYjBjYzg0NWNkMTk
CSeq: 1 ACK
max-forwards: 69
Content-Length: 0
<------------->
— (9 headers 0 lines) —
<— SIP read from UDP:xxx.x.xxx.xx:5060 —>
INVITE sip:100@AstServerIP SIP/2.0
Via: SIP/2.0/UDP xxx.x.xxx.xx:5060;branch=z9hG4bK202fc71fab024132f97a0e7dddaa6d1e
Via: SIP/2.0/UDP 192.168.1.2:37518;branch=z9hG4bK-d8754z-da3c4554641fbf78-1—d8754z-;rport
From: “user” sip:user@AstServerIP;tag=91c0881e
To: sip:100@AstServerIP
Call-ID: NGNiMTFmZGZiZDM2ODk5NWIzN2NjYjBjYzg0NWNkMTk
CSeq: 2 INVITE
Contact: sip:user@xxx.x.xxx.xx
Authorization: Digest username=“user”, realm=“asterisk”, nonce=“6b6095f3”, uri=“sip:100@AstServerIP”, response=“fde3b7f2492d2afdcc5c62401f030b7b”, algorithm=MD5
max-forwards: 69
supported: replaces
user-agent: X-Lite 4.7.1 74247-bf7d9a67-W6.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Content-Length: 300
v=0
o=- 13067379963294747 1 IN IP4 xxx.x.xxx.xx
s=X-Lite release 4.7.1 stamp 74247
c=IN IP4 xxx.x.xxx.xx
t=0 0
m=audio 7072 RTP/AVP 125 100 0 9 8 101
a=rtpmap:125 opus/48000/2
a=fmtp:125 useinbandfec=1
a=rtpmap:100 speex/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
— (15 headers 12 lines) —
Sending to xxx.x.xxx.xx:5060 (NAT)
Using INVITE request as basis request - NGNiMTFmZGZiZDM2ODk5NWIzN2NjYjBjYzg0NWNkMTk
Found peer ‘user’ for ‘user’ from xxx.x.xxx.xx:5060
== Using SIP RTP CoS mark 5
Found RTP audio format 125
Found RTP audio format 100
Found RTP audio format 0
Found RTP audio format 9
Found RTP audio format 8
Found RTP audio format 101
Found audio description format opus for ID 125
Found audio description format speex for ID 100
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|g726|g722|h261|h263|h263p|h264), peer - audio=(ulaw|alaw|speex16|g722|opus)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|g722)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port xxx.x.xxx.xx:7072
Looking for 100 in realtybox (domain AstServerIP)
list_route: route/path hop: sip:user@xxx.x.xxx.xx
<— Transmitting (NAT) to xxx.x.xxx.xx:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP xxx.x.xxx.xx:5060;branch=z9hG4bK202fc71fab024132f97a0e7dddaa6d1e;received=xxx.x.xxx.xx;rport=5060
Via: SIP/2.0/UDP 192.168.1.2:37518;branch=z9hG4bK-d8754z-da3c4554641fbf78-1—d8754z-;rport
From: “user” sip:user@AstServerIP;tag=91c0881e
To: sip:100@AstServerIP
Call-ID: NGNiMTFmZGZiZDM2ODk5NWIzN2NjYjBjYzg0NWNkMTk
CSeq: 2 INVITE
Server: Asterisk PBX 12.8.0-rc2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:100@AstServerIP:5060
Content-Length: 0
<------------>
– Executing [100@realtybox:1] Dial(“SIP/user-00000008”, “SIP/Ouser,60”) in new stack
== Using SIP RTP CoS mark 5
Audio is at 15560
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100012 (g722) to SDP
Adding codec 100011 (g726) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to xxxxxxxxxxxxxx:5060:
INVITE sip:Ouser@1xxxxxxxxxx SIP/2.0
Via: SIP/2.0/UDP xxxxxxxxxxx:5060;branch=z9hG4bK3de8aba6;rport
Max-Forwards: 70
From: “user” sip:103@xxxxxxxxxxx;tag=as07478637
To: sip:Ouser@xxxxxxxxxxx
Contact: sip:103@xxxxxxxxxx:5060
Call-ID: 5abf2cda5c1dba735248f76c0ab80e2b@xxxxxxxxxxxxxxx:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 12.8.0-rc2
Date: Mon, 02 Feb 2015 19:40:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 334
v=0
o=root 445099944 445099944 IN IP4 xxxxxxxxxxxxxx
s=Asterisk PBX 12.8.0-rc2
c=IN IP4 xxxxxxxxxxxxxx
t=0 0
m=audio 15560 RTP/AVP 0 8 9 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
-- Called SIP/Ouser
<— SIP read from UDP:xxxxxxxxxxxxx:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP xxxxxxxxxxx:5060;branch=z9hG4bK3de8aba6;rport
Call-ID: 5abf2cda5c1dba735248f76c0ab80e2b@xxxxxxxxxxx:5060
CSeq: 102 INVITE
From: “user” sip:103@xxxxxxxxxxx;tag=as07478637
To: sip:Ouser@xxxxxxxxxxx;tag=iZiLOwPw4DBhL8C1
Contact: sip:xxxxxxx:5060
Content-Length: 0
<------------->
— (8 headers 0 lines) —
list_route: route/path hop: sip:xxxxxxxxxx:5060
– SIP/Ouser-00000009 is ringing
<— Transmitting (NAT) to xxx.x.xxx.xx:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP xxx.x.xxx.xx:5060;branch=z9hG4bK202fc71fab024132f97a0e7dddaa6d1e;received=xxx.x.xxx.xx;rport=5060
Via: SIP/2.0/UDP 192.168.1.2:37518;branch=z9hG4bK-d8754z-da3c4554641fbf78-1—d8754z-;rport
From: “user” sip:user@AstServerIP;tag=91c0881e
To: sip:100@AstServerIP;tag=as0293562d
Call-ID: NGNiMTFmZGZiZDM2ODk5NWIzN2NjYjBjYzg0NWNkMTk
CSeq: 2 INVITE
Server: Asterisk PBX 12.8.0-rc2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:100@AstServerIP:5060
Content-Length: 0
<------------>
<— SIP read from UDP:xxxxxxx:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP xxxxxxxxxxxxxx:5060;branch=z9hG4bK3de8aba6;rport
Call-ID: 5abf2cda5c1dba735248f76c0ab80e2b@xxxxxxxxxxxxxx:5060
CSeq: 102 INVITE
From: “user” sip:103@xxxxxxxxxxxxxx;tag=as07478637
To: sip:Ouser@xxxx;tag=iZiLOwPw4DBhL8C1
Contact: sip:xxxxxxxxxxxxxx:5060
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REGISTER, REFER, NOTIFY, INFO, PRACK, UPDATE
Supported: 100rel, replaces
Content-Type: application/sdp
Content-Length: 193
v=0
o= 63158019 07514591 IN IP4 xxxxxxxxxxxxxx
s=SIP CALL
c=IN IP4 xxxxxxxxxxxxxx
t=0 0
m=audio 1722 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
— (11 headers 9 lines) —
[Feb 2 13:40:20] WARNING[28990][C-0000000e]: chan_sip.c:10865 process_sdp_o: SDP syntax error in o= line version
list_route: route/path hop: sip:xxxxxxxxxxxxxx:5060
Transmitting (NAT) to xxxxxxxxxxxxxx:5060:
ACK sip:@xxxxxxxxxxxxxx:5060 SIP/2.0
Via: SIP/2.0/UDP xxxxxxxxxxxxxx:5060;branch=z9hG4bK72afb7f2;rport
Max-Forwards: 70
From: “user” sip:103@xxxxxxxxxxxxxx;tag=as07478637
To: sip:Ouser@xxxxxxxxxxxxxx;tag=iZiLOwPw4DBhL8C1
Contact: sip:103@xxxxxxxxxxxxxx:5060
Call-ID: 5abf2cda5c1dba735248f76c0ab80e2b@xxxxxxxxxxxxxx:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 12.8.0-rc2
Content-Length: 0
Reliably Transmitting (NAT) to xxxxxxxxxxxxxx:5060:
BYE sip:@xxxxxxxxxxxxxx:5060 SIP/2.0
Via: SIP/2.0/UDP xxxxxxxxxxxxxx:5060;branch=z9hG4bK20e25d8f;rport
Max-Forwards: 70
From: “user” sip:103@xxxxxxxxxxxxxx;tag=as07478637
To: sip:Ouser@xxxxxxxxxxxxxx;tag=iZiLOwPw4DBhL8C1
Call-ID: 5abf2cda5c1dba735248f76c0ab80e2b@xxxxxxxxxxxxxx:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX 12.8.0-rc2
X-Asterisk-HangupCause: Bearer capability not available
X-Asterisk-HangupCauseCode: 58
Content-Length: 0
Scheduling destruction of SIP dialog ‘5abf2cda5c1dba735248f76c0ab80e2b@xxxxxxxxxxxxxx:5060’ in 32000 ms (Method: INVITE)
Scheduling destruction of SIP dialog ‘5abf2cda5c1dba735248f76c0ab80e2b@xxxxxxxxxxxxxx:5060’ in 32000 ms (Method: INVITE)
== Everyone is busy/congested at this time (1:0/0/1)
– Executing [100@realtybox:2] VoiceMail(“SIP/user-00000008”, “100”) in new stack
Audio is at 11216
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100012 (g722) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<— Reliably Transmitting (NAT) to xxx.x.xxx.xx:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP xxx.x.xxx.xx:5060;branch=z9hG4bK202fc71fab024132f97a0e7dddaa6d1e;received=xxx.x.xxx.xx;rport=5060
Via: SIP/2.0/UDP 192.168.1.2:37518;branch=z9hG4bK-d8754z-da3c4554641fbf78-1—d8754z-;rport
From: “user” sip:user@AstServerIP;tag=91c0881e
To: sip:100@AstServerIP;tag=as0293562d
Call-ID: NGNiMTFmZGZiZDM2ODk5NWIzN2NjYjBjYzg0NWNkMTk
CSeq: 2 INVITE
Server: Asterisk PBX 12.8.0-rc2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:100@AstServerIP:5060
Content-Type: application/sdp
Content-Length: 307
v=0
o=root 1539505467 1539505467 IN IP4 AstServerIP
s=Asterisk PBX 12.8.0-rc2
c=IN IP4 AstServerIP
t=0 0
m=audio 11216 RTP/AVP 0 8 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
<------------>
<— SIP read from UDP:xxxxxxxxxxxxxx:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP xxxxxxxxxxxxxx:5060;branch=z9hG4bK20e25d8f;rport
Call-ID: 5abf2cda5c1dba735248f76c0ab80e2b@xxxxxxxxxxxxxx:5060
CSeq: 103 BYE
From: “user” sip:103@xxxxxxxxxxxxxx;tag=as07478637
To: sip:Ouser@xxxxxxxxxxxxxx;tag=iZiLOwPw4DBhL8C1
Contact: sip:xxxxxxxxxxxxxx:5060
Content-Length: 0
<------------->
— (8 headers 0 lines) —
Really destroying SIP dialog ‘5abf2cda5c1dba735248f76c0ab80e2b@xxxxxxxxxxxxxx:5060’ Method: INVITE
Retransmitting #1 (NAT) to xxx.x.xxx.xx:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP xxx.x.xxx.xx:5060;branch=z9hG4bK202fc71fab024132f97a0e7dddaa6d1e;received=xxx.x.xxx.xx;rport=5060
Via: SIP/2.0/UDP 192.168.1.2:37518;branch=z9hG4bK-d8754z-da3c4554641fbf78-1—d8754z-;rport
From: “user” sip:user@AstServerIP;tag=91c0881e
To: sip:100@AstServerIP;tag=as0293562d
Call-ID: NGNiMTFmZGZiZDM2ODk5NWIzN2NjYjBjYzg0NWNkMTk
CSeq: 2 INVITE
Server: Asterisk PBX 12.8.0-rc2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:100@AstServerIP:5060
Content-Type: application/sdp
Content-Length: 307
v=0
o=root 1539505467 1539505467 IN IP4 AstServerIP
s=Asterisk PBX 12.8.0-rc2
c=IN IP4 AstServerIP
t=0 0
m=audio 11216 RTP/AVP 0 8 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
-- <SIP/user-00000008> Playing 'vm-intro.gsm' (language 'en')
Retransmitting #8 (NAT) to xxx.x.xxx.xx:5060:
NOTIFY sip:user@xxx.x.xxx.xx SIP/2.0
Via: SIP/2.0/UDP AstServerIP:5060;branch=z9hG4bK024db9ab;rport
Max-Forwards: 70
Route: sip:user@xxx.x.xxx.xx
From: “asterisk” sip:asterisk@AstServerIP;tag=as0609715b
To: sip:user@xxx.x.xxx.xx;tag=ca482779
Contact: sip:asterisk@AstServerIP:5060
Call-ID: MmM0ZDQ0Yjg0NTkxNWI2MmFlMTJhOTRiYzdhMjRjYjA
CSeq: 157 NOTIFY
User-Agent: Asterisk PBX 12.8.0-rc2
Event: message-summary
Content-Type: application/simple-message-summary
Subscription-State: active
Content-Length: 96
Messages-Waiting: yes
Message-Account: sip:asterisk@AstServerIP
Voice-Message: 23/0 (0/0)
<— SIP read from UDP:xxx.x.xxx.xx:5060 —>
ACK sip:100@AstServerIP:5060 SIP/2.0
Via: SIP/2.0/UDP xxx.x.xxx.xx:5060;branch=z9hG4bK131783bee4459cb288bb5637635498e6
Via: SIP/2.0/UDP 192.168.1.2:37518;branch=z9hG4bK-d8754z-80ad8b0125210d03-1—d8754z-;rport
From: “user” sip:user@AstServerIP;tag=91c0881e
To: sip:100@AstServerIP;tag=as0293562d
Call-ID: NGNiMTFmZGZiZDM2ODk5NWIzN2NjYjBjYzg0NWNkMTk
CSeq: 2 ACK
Contact: sip:user@xxx.x.xxx.xx
max-forwards: 69
user-agent: X-Lite 4.7.1 74247-bf7d9a67-W6.1
Content-Length: 0
<------------->
— (11 headers 0 lines) —
<— SIP read from UDP:xxx.x.xxx.xx:5060 —>
ACK sip:100@AstServerIP:5060 SIP/2.0
Via: SIP/2.0/UDP xxx.x.xxx.xx:5060;branch=z9hG4bK131783bee4459cb288bb5637635498e6
Via: SIP/2.0/UDP 192.168.1.2:37518;branch=z9hG4bK-d8754z-80ad8b0125210d03-1—d8754z-;rport
From: “user” sip:user@AstServerIP;tag=91c0881e
To: sip:100@AstServerIP;tag=as0293562d
Call-ID: NGNiMTFmZGZiZDM2ODk5NWIzN2NjYjBjYzg0NWNkMTk
CSeq: 2 ACK
Contact: sip:user@xxx.x.xxx.xx
max-forwards: 69
user-agent: X-Lite 4.7.1 74247-bf7d9a67-W6.1
Content-Length: 0
<------------->
— (11 headers 0 lines) —
<— SIP read from UDP:xxx.x.xxx.xx:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP AstServerIP:5060;branch=z9hG4bK024db9ab;rport
From: “asterisk” sip:asterisk@AstServerIP;tag=as0609715b
To: sip:user@xxx.x.xxx.xx;tag=ca482779
Call-ID: MmM0ZDQ0Yjg0NTkxNWI2MmFlMTJhOTRiYzdhMjRjYjA
CSeq: 156 NOTIFY
Contact: sip:user@xxx.x.xxx.xx
user-agent: X-Lite release 4.7.1 stamp 74247
Content-Length: 0
<------------->
— (9 headers 0 lines) —
<— SIP read from UDP:xxx.x.xxx.xx:5060 —>
REGISTER sip:AstServerIP SIP/2.0
Via: SIP/2.0/UDP xxx.x.xxx.xx:5060;branch=z9hG4bKb3a115b60d9325ea645253aaa525a8e2
Via: SIP/2.0/UDP 192.168.1.2:37518;branch=z9hG4bK-d8754z-274eed7af8b0003c-1—d8754z-;rport
From: “user” sip:user@AstServerIP;tag=a1f1b617
To: “user” sip:user@AstServerIP
Call-ID: Y2IzNzczOTc1Zjc4MmIwYWQxMDA0MzQwZWJjMGI3NjA
CSeq: 104 REGISTER
Contact: sip:user@xxx.x.xxx.xx
Authorization: Digest username=“user”, realm=“asterisk”, nonce=“0bd1433b”, uri=“sip:AstServerIP”, response=“82bae596261ee482c1febc478223eb75”, algorithm=MD5
max-forwards: 69
expires: 30
user-agent: X-Lite 4.7.1 74247-bf7d9a67-W6.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Length: 0
<------------->
— (14 headers 0 lines) —
Sending to xxx.x.xxx.xx:5060 (NAT)
[Feb 2 13:40:23] NOTICE[28990]: chan_sip.c:16778 check_auth: Correct auth, but based on stale nonce received from ‘“user” sip:user@AstServerIP;tag=a1f1b617’
<— Transmitting (NAT) to xxx.x.xxx.xx:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP xxx.x.xxx.xx:5060;branch=z9hG4bKb3a115b60d9325ea645253aaa525a8e2;received=xxx.x.xxx.xx;rport=5060
Via: SIP/2.0/UDP 192.168.1.2:37518;branch=z9hG4bK-d8754z-274eed7af8b0003c-1—d8754z-;rport
From: “user” sip:user@AstServerIP;tag=a1f1b617
To: “user” sip:user@AstServerIP;tag=as20d31114
Call-ID: Y2IzNzczOTc1Zjc4MmIwYWQxMDA0MzQwZWJjMGI3NjA
CSeq: 104 REGISTER
Server: Asterisk PBX 12.8.0-rc2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“042115bf”, stale=true
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘Y2IzNzczOTc1Zjc4MmIwYWQxMDA0MzQwZWJjMGI3NjA’ in 32000 ms (Method: REGISTER)
[Feb 2 13:40:23] NOTICE[31675][C-0000000e]: res_rtp_asterisk.c:4467 ast_rtp_read: Unknown RTP codec 126 received from ‘xxx.x.xxx.xx:7072’
<— SIP read from UDP:xxx.x.xxx.xx:5060 —>
REGISTER sip:AstServerIP SIP/2.0
Via: SIP/2.0/UDP xxx.x.xxx.xx:5060;branch=z9hG4bK0708ab5295dfb488b7ec65e9c88ffe00
Via: SIP/2.0/UDP 192.168.1.2:37518;branch=z9hG4bK-d8754z-827f52270976962d-1—d8754z-;rport
From: “user” sip:user@AstServerIP;tag=a1f1b617
To: “user” sip:user@AstServerIP
Call-ID: Y2IzNzczOTc1Zjc4MmIwYWQxMDA0MzQwZWJjMGI3NjA
CSeq: 105 REGISTER
Contact: sip:user@xxx.x.xxx.xx
Authorization: Digest username=“user”, realm=“asterisk”, nonce=“042115bf”, uri=“sip:AstServerIP”, response=“639d7b7b4de8faf64ac40859d1d16b94”, algorithm=MD5
max-forwards: 69
expires: 30
user-agent: X-Lite 4.7.1 74247-bf7d9a67-W6.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Length: 0
<------------->
— (14 headers 0 lines) —
Sending to xxx.x.xxx.xx:5060 (NAT)
<— Transmitting (NAT) to xxx.x.xxx.xx:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP xxx.x.xxx.xx:5060;branch=z9hG4bK0708ab5295dfb488b7ec65e9c88ffe00;received=xxx.x.xxx.xx;rport=5060
Via: SIP/2.0/UDP 192.168.1.2:37518;branch=z9hG4bK-d8754z-827f52270976962d-1—d8754z-;rport
From: “user” sip:user@AstServerIP;tag=a1f1b617
To: “user” sip:user@AstServerIP;tag=as20d31114
Call-ID: Y2IzNzczOTc1Zjc4MmIwYWQxMDA0MzQwZWJjMGI3NjA
CSeq: 105 REGISTER
Server: Asterisk PBX 12.8.0-rc2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 60
Contact: sip:user@xxx.x.xxx.xx;expires=60
Date: Mon, 02 Feb 2015 19:40:23 GMT
Content-Length: 0
<------------>
Reliably Transmitting (NAT) to xxx.x.xxx.xx:5060:
NOTIFY sip:user@xxx.x.xxx.xx SIP/2.0
Via: SIP/2.0/UDP AstServerIP:5060;branch=z9hG4bK024db9ab;rport
Max-Forwards: 70
Route: sip:user@xxx.x.xxx.xx
From: “asterisk” sip:asterisk@AstServerIP;tag=as0609715b
To: sip:user@xxx.x.xxx.xx;tag=ca482779
Contact: sip:asterisk@AstServerIP:5060
Call-ID: MmM0ZDQ0Yjg0NTkxNWI2MmFlMTJhOTRiYzdhMjRjYjA
CSeq: 158 NOTIFY
User-Agent: Asterisk PBX 12.8.0-rc2
Event: message-summary
Content-Type: application/simple-message-summary
Subscription-State: active
Content-Length: 96
Messages-Waiting: yes
Message-Account: sip:asterisk@AstServerIP
Voice-Message: 23/0 (0/0)
–
<— SIP read from UDP:xxx.x.xxx.xx:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP AstServerIP:5060;branch=z9hG4bK024db9ab;rport
From: “asterisk” sip:asterisk@AstServerIP;tag=as0609715b
To: sip:user@xxx.x.xxx.xx;tag=ca482779
Call-ID: MmM0ZDQ0Yjg0NTkxNWI2MmFlMTJhOTRiYzdhMjRjYjA
CSeq: 156 NOTIFY
Contact: sip:user@xxx.x.xxx.xx
user-agent: X-Lite release 4.7.1 stamp 74247
Content-Length: 0
<------------->
— (9 headers 0 lines) —
<— SIP read from UDP:xxx.x.xxx.xx:5060 —>
BYE sip:100@AstServerIP:5060 SIP/2.0
Via: SIP/2.0/UDP xxx.x.xxx.xx:5060;branch=z9hG4bK49d504dc855fbc251f5afe17ce928eaa
Via: SIP/2.0/UDP 192.168.1.2:37518;branch=z9hG4bK-d8754z-152744520d506571-1—d8754z-;rport
From: “user” sip:user@AstServerIP;tag=91c0881e
To: sip:100@AstServerIP;tag=as0293562d
Call-ID: NGNiMTFmZGZiZDM2ODk5NWIzN2NjYjBjYzg0NWNkMTk
CSeq: 3 BYE
Contact: sip:user@xxx.x.xxx.xx
Authorization: Digest username=“user”, realm=“asterisk”, nonce=“6b6095f3”, uri=“sip:100@AstServerIP:5060”, response=“dfaaf1a577c55d82bb74e5eb7d4019d0”, algorithm=MD5
max-forwards: 69
user-agent: X-Lite 4.7.1 74247-bf7d9a67-W6.1
Content-Length: 0
<------------->
— (12 headers 0 lines) —
Sending to xxx.x.xxx.xx:5060 (NAT)
Scheduling destruction of SIP dialog ‘NGNiMTFmZGZiZDM2ODk5NWIzN2NjYjBjYzg0NWNkMTk’ in 32000 ms (Method: BYE)
<— Transmitting (NAT) to xxx.x.xxx.xx:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP xxx.x.xxx.xx:5060;branch=z9hG4bK49d504dc855fbc251f5afe17ce928eaa;received=xxx.x.xxx.xx;rport=5060
Via: SIP/2.0/UDP 192.168.1.2:37518;branch=z9hG4bK-d8754z-152744520d506571-1—d8754z-;rport
From: “user” sip:user@AstServerIP;tag=91c0881e
To: sip:100@AstServerIP;tag=as0293562d
Call-ID: NGNiMTFmZGZiZDM2ODk5NWIzN2NjYjBjYzg0NWNkMTk
CSeq: 3 BYE
Server: Asterisk PBX 12.8.0-rc2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------>
== Spawn extension (realtybox, 100, 2) exited non-zero on 'SIP/user-00000008’
Retransmitting #1 (NAT) to xxx.x.xxx.xx:5060:
NOTIFY sip:user@xxx.x.xxx.xx SIP/2.0
Via: SIP/2.0/UDP AstServerIP:5060;branch=z9hG4bK024db9ab;rport
Max-Forwards: 70
Route: sip:user@xxx.x.xxx.xx
From: “asterisk” sip:asterisk@AstServerIP;tag=as0609715b
To: sip:user@xxx.x.xxx.xx;tag=ca482779
Contact: sip:asterisk@AstServerIP:5060
Call-ID: MmM0ZDQ0Yjg0NTkxNWI2MmFlMTJhOTRiYzdhMjRjYjA
CSeq: 158 NOTIFY
User-Agent: Asterisk PBX 12.8.0-rc2
Event: message-summary
Content-Type: application/simple-message-summary
Subscription-State: active
Content-Length: 96
Messages-Waiting: yes
Message-Account: sip:asterisk@AstServerIP
Voice-Message: 23/0 (0/0)
<— SIP read from UDP:xxx.x.xxx.xx:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP AstServerIP:5060;branch=z9hG4bK024db9ab;rport
From: “asterisk” sip:asterisk@AstServerIP;tag=as0609715b
To: sip:user@xxx.x.xxx.xx;tag=ca482779
Call-ID: MmM0ZDQ0Yjg0NTkxNWI2MmFlMTJhOTRiYzdhMjRjYjA
CSeq: 156 NOTIFY
Contact: sip:user@xxx.x.xxx.xx
user-agent: X-Lite release 4.7.1 stamp 74247
Content-Length: 0
<------------->
— (9 headers 0 lines) —
Retransmitting #2 (NAT) to xxx.x.xxx.xx:5060:
NOTIFY sip:user@xxx.x.xxx.xx SIP/2.0
Via: SIP/2.0/UDP AstServerIP:5060;branch=z9hG4bK024db9ab;rport
Max-Forwards: 70
Route: sip:user@xxx.x.xxx.xx
From: “asterisk” sip:asterisk@AstServerIP;tag=as0609715b
To: sip:user@xxx.x.xxx.xx;tag=ca482779
Contact: sip:asterisk@AstServerIP:5060
Call-ID: MmM0ZDQ0Yjg0NTkxNWI2MmFlMTJhOTRiYzdhMjRjYjA
CSeq: 158 NOTIFY
User-Agent: Asterisk PBX 12.8.0-rc2
Event: message-summary
Content-Type: application/simple-message-summary
Subscription-State: active
Content-Length: 96
Messages-Waiting: yes
Message-Account: sip:asterisk@AstServerIP
Voice-Message: 23/0 (0/0)
<— SIP read from UDP:xxx.x.xxx.xx:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP AstServerIP:5060;branch=z9hG4bK024db9ab;rport
From: “asterisk” sip:asterisk@AstServerIP;tag=as0609715b
To: sip:user@xxx.x.xxx.xx;tag=ca482779
Call-ID: MmM0ZDQ0Yjg0NTkxNWI2MmFlMTJhOTRiYzdhMjRjYjA
CSeq: 156 NOTIFY
Contact: sip:user@xxx.x.xxx.xx
user-agent: X-Lite release 4.7.1 stamp 74247
Content-Length: 0
<------------->
— (9 headers 0 lines) —
Retransmitting #9 (NAT) to xxx.x.xxx.xx:5060:
NOTIFY sip:user@xxx.x.xxx.xx SIP/2.0
Via: SIP/2.0/UDP AstServerIP:5060;branch=z9hG4bK024db9ab;rport
Max-Forwards: 70
Route: sip:user@xxx.x.xxx.xx
From: “asterisk” sip:asterisk@AstServerIP;tag=as0609715b
To: sip:user@xxx.x.xxx.xx;tag=ca482779
Contact: sip:asterisk@AstServerIP:5060
Call-ID: MmM0ZDQ0Yjg0NTkxNWI2MmFlMTJhOTRiYzdhMjRjYjA
CSeq: 157 NOTIFY
User-Agent: Asterisk PBX 12.8.0-rc2
Event: message-summary
Content-Type: application/simple-message-summary
Subscription-State: active
Content-Length: 96
Messages-Waiting: yes
Message-Account: sip:asterisk@AstServerIP
Voice-Message: 23/0 (0/0)
<— SIP read from UDP:xxx.x.xxx.xx:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP AstServerIP:5060;branch=z9hG4bK024db9ab;rport
From: “asterisk” sip:asterisk@AstServerIP;tag=as0609715b
To: sip:user@xxx.x.xxx.xx;tag=ca482779
Call-ID: MmM0ZDQ0Yjg0NTkxNWI2MmFlMTJhOTRiYzdhMjRjYjA
CSeq: 156 NOTIFY
Contact: sip:user@xxx.x.xxx.xx
user-agent: X-Lite release 4.7.1 stamp 74247
Content-Length: 0
<------------->
— (9 headers 0 lines) —
Retransmitting #3 (NAT) to xxx.x.xxx.xx:5060:
NOTIFY sip:user@xxx.x.xxx.xx SIP/2.0
Via: SIP/2.0/UDP AstServerIP:5060;branch=z9hG4bK024db9ab;rport
Max-Forwards: 70
Route: sip:user@xxx.x.xxx.xx
From: “asterisk” sip:asterisk@AstServerIP;tag=as0609715b
To: sip:user@xxx.x.xxx.xx;tag=ca482779
Contact: sip:asterisk@AstServerIP:5060
Call-ID: MmM0ZDQ0Yjg0NTkxNWI2MmFlMTJhOTRiYzdhMjRjYjA
CSeq: 158 NOTIFY
User-Agent: Asterisk PBX 12.8.0-rc2
Event: message-summary
Content-Type: application/simple-message-summary
Subscription-State: active
Content-Length: 96
Messages-Waiting: yes
Message-Account: sip:asterisk@AstServerIP
Voice-Message: 23/0 (0/0)
<— SIP read from UDP:xxx.x.xxx.xx:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP AstServerIP:5060;branch=z9hG4bK024db9ab;rport
From: “asterisk” sip:asterisk@AstServerIP;tag=as0609715b
To: sip:user@xxx.x.xxx.xx;tag=ca482779
Call-ID: MmM0ZDQ0Yjg0NTkxNWI2MmFlMTJhOTRiYzdhMjRjYjA
CSeq: 156 NOTIFY
Contact: sip:user@xxx.x.xxx.xx
user-agent: X-Lite release 4.7.1 stamp 74247
Content-Length: 0
[Feb 2 13:40:27] WARNING[28990]: chan_sip.c:4103 retrans_pkt: Retransmission timeout reached on transmission 15b703b62304d95f32ece998cd5e8acb for seqno 1 (Critical Response) – See wiki.asterisk.org/wiki/display/ … nsmissions
Packet timed out after 31999ms with no response
Really destroying SIP dialog ‘15b703b62304d95f32ece998cd5e8acb’ Method: INVITE
Retransmitting #10 (NAT) to xxx.x.xxx.xx:5060:
NOTIFY sip:user@xxx.x.xxx.xx SIP/2.0
Via: SIP/2.0/UDP AstServerIP:5060;branch=z9hG4bK024db9ab;rport
Max-Forwards: 70
Route: sip:user@xxx.x.xxx.xx
From: “asterisk” sip:asterisk@AstServerIP;tag=as0609715b
To: sip:user@xxx.x.xxx.xx;tag=ca482779
Contact: sip:asterisk@AstServerIP:5060
Call-ID: MmM0ZDQ0Yjg0NTkxNWI2MmFlMTJhOTRiYzdhMjRjYjA
CSeq: 157 NOTIFY
User-Agent: Asterisk PBX 12.8.0-rc2
Event: message-summary
Content-Type: application/simple-message-summary
Subscription-State: active
Content-Length: 96
Messages-Waiting: yes
Message-Account: sip:asterisk@AstServerIP
Voice-Message: 23/0 (0/0)
<— SIP read from UDP:xxx.x.xxx.xx:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP AstServerIP:5060;branch=z9hG4bK024db9ab;rport
From: “asterisk” sip:asterisk@AstServerIP;tag=as0609715b
To: sip:user@xxx.x.xxx.xx;tag=ca482779
Call-ID: MmM0ZDQ0Yjg0NTkxNWI2MmFlMTJhOTRiYzdhMjRjYjA
CSeq: 156 NOTIFY
Contact: sip:user@xxx.x.xxx.xx
user-agent: X-Lite release 4.7.1 stamp 74247
Content-Length: 0
<------------->
— (9 headers 0 lines) —
[Feb 2 13:40:29] WARNING[28990]: chan_sip.c:4103 retrans_pkt: Retransmission timeout reached on transmission MmM0ZDQ0Yjg0NTkxNWI2MmFlMTJhOTRiYzdhMjRjYjA for seqno 157 (Non-critical Request) – See wiki.asterisk.org/wiki/display/ … nsmissions
Packet timed out after 32000ms with no response
Retransmitting #4 (NAT) to xxx.x.xxx.xx:5060:
NOTIFY sip:user@xxx.x.xxx.xx SIP/2.0
Via: SIP/2.0/UDP AstServerIP:5060;branch=z9hG4bK024db9ab;rport
Max-Forwards: 70
Route: sip:user@xxx.x.xxx.xx
From: “asterisk” sip:asterisk@AstServerIP;tag=as0609715b
To: sip:user@xxx.x.xxx.xx;tag=ca482779
Contact: sip:asterisk@AstServerIP:5060
Call-ID: MmM0ZDQ0Yjg0NTkxNWI2MmFlMTJhOTRiYzdhMjRjYjA
CSeq: 158 NOTIFY
User-Agent: Asterisk PBX 12.8.0-rc2
Event: message-summary
Content-Type: application/simple-message-summary
Subscription-State: active
Content-Length: 96
Messages-Waiting: yes
Message-Account: sip:asterisk@AstServerIP
Voice-Message: 23/0 (0/0)
<— SIP read from UDP:xxxxxxxxxxxxxx:5060 —>
NOTIFY sip:s@AstServerIP:5060 SIP/2.0
Via: SIP/2.0/UDP xxxxxxxxxx;branch=z9hG4bKb902.9a843653.0
Via: SIP/2.0/UDP xxxxxxxxxxxxxx:8003;branch=z9hG4bK.973251529a3577;rport=8003;alias
Via: SIP/2.0/UDP xxxxxxxxxxxxxxx:8003;rport
From: sip:mwi@xxxxxxxxxxxxxx;tag=aa2c6dec
To: sip:3202577777@xxxxxxxxxxxxxx
Call-ID: dc5dd7133116b68d-54cf@xxxxxxxxxx
CSeq: 260 NOTIFY
Max-Forwards: 9
Content-Length: 103
User-Agent: MWI Agent
Event: message-summary
Content-Type: application/simple-message-summary
Messages-Waiting: Yes
Message-Account: sip:3202577777@xxxxxxxxxxx:5060
Voice-Messages: 4/0 (4/0)
<------------->
— (13 headers 3 lines) —
Sending to xxxxxxxxxx:5060 (NAT)
<— Transmitting (NAT) to xxxxxxxxxx:5060 —>
SIP/2.0 489 Bad event
Via: SIP/2.0/UDP xxxxxxxxxxx;branch=z9hG4bKb902.9a843653.0;received=6xxxxxxxxxxx;rport=5060
Via: SIP/2.0/UDP xxxxxxxxxxxx:8003;branch=z9hG4bK.973251529a3577;rport=8003;alias
Via: SIP/2.0/UDP xxxxxxxxxxx:8003;rport
From: sip:mwi@xxxxxxx;tag=aa2c6dec
To: sip:xxxxxxxxxxx7@xxxxxxx;tag=as6b7be3fa
Call-ID: dc5dd7133116b68d-54cf@xxxxxxx
CSeq: 260 NOTIFY
Server: Asterisk PBX 12.8.0-rc2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘dc5dd7133116b68d-54cf@50.97.129.82’ in 32000 ms (Method: NOTIFY)
<— SIP read from UDP:xxx.x.xxx.xx:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP AstServerIP:5060;branch=z9hG4bK024db9ab;rport
From: “asterisk” sip:asterisk@AstServerIP;tag=as0609715b
To: sip:user@xxx.x.xxx.xx;tag=ca482779
Call-ID: MmM0ZDQ0Yjg0NTkxNWI2MmFlMTJhOTRiYzdhMjRjYjA
CSeq: 156 NOTIFY
Contact: sip:user@xxx.x.xxx.xx
user-agent: X-Lite release 4.7.1 stamp 74247
Content-Length: 0
Thank you!
There is a syntax error in the peer’s SDP user name. Section 5.2 of tools.ietf.org/html/rfc4566 says:
is the user’s login on the originating host, or it is "-"
if the originating host does not support the concept of user IDs.
The MUST NOT contain spaces.
but the user name is empty.
I think you have to fix the peer.
Hi!
The error message is about version : SDP syntax error in o= line version
Here is the peer definition (but why is it appearing empty? and sometimes showing peer username? (sip:ouser…):
[Ouser]
username = Ouser
type = friend
context = mycontext
host = xxx.xx.xxx.xx
qualify = no
nat=force_rport,comedia
dtmfmode = rfc2833
disallow = all
allow=ulaw
allow=alaw
allow=g722
allow=g726
allow=h261
allow=h263
allow=h263p
allow=h264
callerid = “Ouser” <100>
mailbox=100@default
Thank you so much!
The peer is broken. This is nothing to do with the Asterisk configuration. This needs to be fixed outside of Asterisk.
Sorry but this means that the ip phone is not working correctly? needs update? or what?
Thank you very much.
Yes. Assuming the device reporting various addresses consisting of different numbers of x’s is an IP phone.
Thank you very much!!! thank you!