Probation passed - setting RTP source address

Asterisk Asterisk Support
1

I have an Debian 7, and I use an Asterisk 11 on it.

I tried to make call with my softphone using Zoiper to my phone that use ATA cisco spa 8000.

But I can’t get any audio.

WIFI phone ==> ATA Cisco SPA 8000 phone (Can’t hear sound/Just Ringing)

 == Using SIP RTP CoS mark 5
    -- Executing [3020@ramais:1] Dial("SIP/3028-0000004f", "SIP/3020,60,tT") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/3020
    -- SIP/3020-00000050 is ringing
    -- SIP/3020-00000050 answered SIP/3028-0000004f
       > 0x90fd5e8 -- Probation passed - setting RTP source address to XXX.X.X.XX:16479
[May  8 15:47:35] WARNING[3048]: chan_sip.c:4024 retrans_pkt: Retransmission timeout reached on transmission Luu1OPUGYl60fdj3rodhzw.. for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6399ms with no response
[May  8 15:47:35] WARNING[3048]: chan_sip.c:4053 retrans_pkt: Hanging up call Luu1OPUGYl60fdj3rodhzw.. - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
  == Spawn extension (ramais, 3020, 1) exited non-zero on 'SIP/3028-0000004f'

Sip Debug On

    -- SIP/3028-0000004e answered SIP/3020-0000004d
Audio is at 14858
Adding codec 100004 (alaw) to SDP
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to XXX.X.X.XX:5089 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP XXX.X.X.XX5089;branch=z9hG4bK-f473ef29;received=XXX.X.X.XX;rport=5089
From: "3020" <sip:3020@XXX.X.X.X>;tag=498e957dd1963555o1
To: <sip:3028@XXX.X.X.X>;tag=as03ed3577
Call-ID: 1d63169d-dcd6a6b5@XXX.X.X.X
CSeq: 102 INVITE
Server: Asterisk PBX 11.17.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:3028@XXX.X.X.X:5089>
Content-Type: application/sdp
Content-Length: 254

v=0
o=root 1026738788 1026738788 IN IP4 XXX.X.X.X
s=Asterisk PBX 11.17.1
c=IN IP4 XXX.X.X.X
t=0 0
m=audio 14858 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>

<--- SIP read from UDP:XXX.X.X.XX:5089 --->
ACK sip:3028@XXX.X.X.X:5089 SIP/2.0
Via: SIP/2.0/UDP XXX.X.X.X:5089;branch=z9hG4bK-b41b580c;rport
From: "3020" <sip:3020@XXX.X.X.X>;tag=498e957dd1963555o1
To: <sip:3028@XXX.X.X.X>;tag=as03ed3577
Call-ID: 1d63169d-dcd6a6b5@XXX.X.X.X
CSeq: 102 ACK
Max-Forwards: 70
Authorization: Digest username="3020",realm="asterisk",nonce="0cdb90b9",uri="sip:3028@XXX.X.X.X:5089",algorithm=MD5,response="f33d5c86422e51594b92440632335a6e"
Contact: "3020" <sip:3020@XXX.X.X.X:5089>
User-Agent: Linksys/SPA8000-6.1.12(XU)
Allow-Events: talk, hold, conference
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
       > 0x90fc3d8 -- Probation passed - setting RTP source address to XXX.X.X.X:16477

<--- SIP read from UDP:XXX.X.X.X:5089 --->
BYE sip:3028@XXX.X.X.X:5089 SIP/2.0
Via: SIP/2.0/UDP XXX.X.X.X:5089;branch=z9hG4bK-1e7a43a7;rport
From: "3020" <sip:3020@XXX.X.X.X>;tag=498e957dd1963555o1
To: <sip:3028@XXX.X.X.X>;tag=as03ed3577
Call-ID: 1d63169d-dcd6a6b5@XXX.X.X.X
CSeq: 103 BYE
Max-Forwards: 70
Authorization: Digest username="3020",realm="asterisk",nonce="0cdb90b9",uri="sip:3028@XXX.X.X.X:5089",algorithm=MD5,response="3c8d0f1530c50848f2b44dea4ee87b5c"
User-Agent: Linksys/SPA8000-6.1.12(XU)
Allow-Events: talk, hold, conference
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Sending to XXX.X.X.X:5089 (NAT)
Scheduling destruction of SIP dialog '1d63169d-dcd6a6b5@XXX.X.X.X' in 6400 ms (Method: BYE)

<--- Transmitting (NAT) to XXX.X.X.X:5089 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP XXX.X.X.X:5089;branch=z9hG4bK-1e7a43a7;received=XXX.X.X.X;rport=5089
From: "3020" <sip:3020@XXX.X.X.X>;tag=498e957dd1963555o1
To: <sip:3028@XXX.X.X.X>;tag=as03ed3577
Call-ID: 1d63169d-dcd6a6b5@XXX.X.X.X
CSeq: 103 BYE
Server: Asterisk PBX 11.17.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '59d761cd4ba61f305e3bc2ba6d069108@xxx.xxx.x.x:5089' in 6400 ms (Method: INVITE)
set_destination: Parsing <sip:3028@xxx.xxx.xxx.xxx:51508;transport=UDP> for address/port to send to
set_destination: set destination to xxx.xxx.xxx.xxx:51508
Reliably Transmitting (NAT) to xxx.xxx.xxx.xxx:51508:
BYE sip:3028@xxx.xxx.xxx.xxx:51508;transport=UDP SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.x.x:5089;branch=z9hG4bK17ff2c6c;rport
Max-Forwards: 70
From: "3020" <sip:3020@xxx.xxx.x.x:5089>;tag=as2179a192
To: <sip:3028@xxx.xxx.xxx.xxx:51508;transport=UDP;rinstance=34563efa1eb36438>;tag=44f69c34
Call-ID: 59d761cd4ba61f305e3bc2ba6d069108@xxx.xxx.x.x:5089
CSeq: 103 BYE
User-Agent: Asterisk PBX 11.17.1
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
  == Spawn extension (ramais, 3028, 1) exited non-zero on 'SIP/3020-0000004d'
Retransmitting #1 (NAT) to xxx.xxx.xxx.xxx:51508:
BYE sip:3028@xxx.xxx.xxx.xxx:51508;transport=UDP SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.x.x:5089;branch=z9hG4bK17ff2c6c;rport
Max-Forwards: 70
From: "3020" <sip:3020@xxx.xxx.x.x:5089>;tag=as2179a192
To: <sip:3028@xxx.xxx.xxx.xxx:51508;transport=UDP;rinstance=34563efa1eb36438>;tag=44f69c34
Call-ID: 59d761cd4ba61f305e3bc2ba6d069108@xxx.xxx.x.x:5089
CSeq: 103 BYE
User-Agent: Asterisk PBX 11.17.1
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---

<--- SIP read from UDP:xxx.xxx.xxx.xxx:51508 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP xxx.xxx.x.x:5089;branch=z9hG4bK17ff2c6c;rport=5089;received=xxx.xx.xxx.x
Contact: <sip:3028@xxx.xxx.xxx.xxx:51508;transport=UDP>
To: <sip:3028@xxx.xxx.xxx.xxx:51508;transport=UDP;rinstance=34563efa1eb36438>;tag=44f69c34
From: "3020" <sip:3020@xxx.xxx.x.x:5089>;tag=as2179a192
Call-ID: 59d761cd4ba61f305e3bc2ba6d069108@xxx.xxx.x.x:5089
CSeq: 103 BYE
User-Agent: Zoiper r30798
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '59d761cd4ba61f305e3bc2ba6d069108@xxx.xxx.x.x:5089' Method: INVITE
Reliably Transmitting (no NAT) to XXX.X.X.X:50001:
OPTIONS sip:3024@XXX.X.X.X:50001 SIP/2.0
Via: SIP/2.0/UDP XXX.X.X.X:5089;branch=z9hG4bK428cca17
Max-Forwards: 70
From: "asterisk" <sip:asterisk@XXX.X.X.X:5089>;tag=as657e3ac3
To: <sip:3024@XXX.X.X.X:50001>
Contact: <sip:asterisk@XXX.X.X.X:5089>
Call-ID: 089015527cd07f3868152701072c5a9d@XXX.X.X.X:5089
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.17.1
Date: Fri, 08 May 2015 18:44:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:XXX.X.X.X:50001 --->
SIP/2.0 200 OK
To: <sip:3024@XXX.X.X.X:50001>;tag=365c01b6386f3791i0
From: "asterisk" <sip:asterisk@XXX.X.X.X:5089>;tag=as657e3ac3
Call-ID: 089015527cd07f3868152701072c5a9d@XXX.X.X.X:5089
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP XXX.X.X.X:5089;branch=z9hG4bK428cca17
Server: Linksys/SPA8000-6.1.12(XU)
Allow-Events: talk, hold, conference
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces

<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '089015527cd07f3868152701072c5a9d@XXX.X.X.X:5089' Method: OPTIONS
Reliably Transmitting (no NAT) to XXX.X.X.X:50000:
OPTIONS sip:3022@XXX.X.X.X:50000 SIP/2.0
Via: SIP/2.0/UDP XXX.X.X.X:5089;branch=z9hG4bK39ecc14c
Max-Forwards: 70
From: "asterisk" <sip:asterisk@XXX.X.X.X:5089>;tag=as0183bc1a
To: <sip:3022@XXX.X.X.X:50000>
Contact: <sip:asterisk@XXX.X.X.X:5089>
Call-ID: 30a7735612f39c4221d4cf0e67197ea0@XXX.X.X.X:5089
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.17.1
Date: Fri, 08 May 2015 18:44:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
Reliably Transmitting (no NAT) to XXX.X.X.X:50000:
OPTIONS sip:3023@XXX.X.X.X:50000 SIP/2.0
Via: SIP/2.0/UDP XXX.X.X.X:5089;branch=z9hG4bK6902da7d
Max-Forwards: 70
From: "asterisk" <sip:asterisk@XXX.X.X.X:5089>;tag=as780797c0
To: <sip:3023@XXX.X.X.X:50000>
Contact: <sip:asterisk@XXX.X.X.X:5089>
Call-ID: 364f38503c0dd924149bed282e2cdf29@XXX.X.X.X:5089
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.17.1
Date: Fri, 08 May 2015 18:44:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:XXX.X.X.X:50000 --->
SIP/2.0 200 OK
To: <sip:3022@XXX.X.X.X:50000>;tag=d92ada5df11e5f27i0
From: "asterisk" <sip:asterisk@XXX.X.X.X:5089>;tag=as0183bc1a
Call-ID: 30a7735612f39c4221d4cf0e67197ea0@XXX.X.X.X:5089
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP XXX.X.X.X:5089;branch=z9hG4bK39ecc14c
Server: Linksys/SPA8000-6.1.12(XU)
Allow-Events: talk, hold, conference
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces

<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '30a7735612f39c4221d4cf0e67197ea0@XXX.X.X.X:5089' Method: OPTIONS

<--- SIP read from UDP:XXX.X.X.X:50000 --->
SIP/2.0 200 OK
To: <sip:3023@XXX.X.X.X:50000>;tag=84e90ad77fd9d17i1
From: "asterisk" <sip:asterisk@XXX.X.X.X:5089>;tag=as780797c0
Call-ID: 364f38503c0dd924149bed282e2cdf29@XXX.X.X.X:5089
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP XXX.X.X.X:5089;branch=z9hG4bK6902da7d
Server: Linksys/SPA8000-6.1.12(XU)
Allow-Events: talk, hold, conference
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces

<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '364f38503c0dd924149bed282e2cdf29@XXX.X.X.X:5089' Method: OPTIONS
Reliably Transmitting (no NAT) to XXX.X.X.X:50001:
OPTIONS sip:3025@XXX.X.X.X:50001 SIP/2.0
Via: SIP/2.0/UDP XXX.X.X.X:5089;branch=z9hG4bK2a938745
Max-Forwards: 70
From: "asterisk" <sip:asterisk@XXX.X.X.X:5089>;tag=as257439fd
To: <sip:3025@XXX.X.X.X:50001>
Contact: <sip:asterisk@XXX.X.X.X:5089>
Call-ID: 2ff277c27c7ca308360f15c933939949@XXX.X.X.X:5089
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.17.1
Date: Fri, 08 May 2015 18:44:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:XXX.X.X.X:50001 --->
SIP/2.0 200 OK
To: <sip:3025@XXX.X.X.X:50001>;tag=5afbe88e7a8be99i1
From: "asterisk" <sip:asterisk@XXX.X.X.X:5089>;tag=as257439fd
Call-ID: 2ff277c27c7ca308360f15c933939949@XXX.X.X.X:5089
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP XXX.X.X.X:5089;branch=z9hG4bK2a938745
Server: Linksys/SPA8000-6.1.12(XU)
Allow-Events: talk, hold, conference
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces

<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '2ff277c27c7ca308360f15c933939949@XXX.X.X.X:5089' Method: OPTIONS
Reliably Transmitting (no NAT) to XXX.X.X.X:50002:
OPTIONS sip:3026@XXX.X.X.X:50002 SIP/2.0
Via: SIP/2.0/UDP XXX.X.X.X:5089;branch=z9hG4bK7df232df
Max-Forwards: 70
From: "asterisk" <sip:asterisk@XXX.X.X.X:5089>;tag=as6dacb67e
To: <sip:3026@XXX.X.X.X:50002>
Contact: <sip:asterisk@XXX.X.X.X:5089>
Call-ID: 3f3c70e568b23f763561b8ba6e165b71@XXX.X.X.X:5089
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.17.1
Date: Fri, 08 May 2015 18:44:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:XXX.X.X.X:50002 --->
SIP/2.0 200 OK
To: <sip:3026@XXX.X.X.X:50002>;tag=f10bf90bed21208ai0
From: "asterisk" <sip:asterisk@XXX.X.X.X:5089>;tag=as6dacb67e
Call-ID: 3f3c70e568b23f763561b8ba6e165b71@XXX.X.X.X:5089
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP XXX.X.X.X:5089;branch=z9hG4bK7df232df
Server: Linksys/SPA8000-6.1.12(XU)
Allow-Events: talk, hold, conference
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces

<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '3f3c70e568b23f763561b8ba6e165b71@XXX.X.X.X:5089' Method: OPTIONS
Really destroying SIP dialog '1d63169d-dcd6a6b5@XXX.X.X.X' Method: BYE

<--- SIP read from UDP:xxx.xxx.xxx.xxx:51508 --->
REGISTER sip:mydomain.noip.us:5089;transport=UDP SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:51508;branch=z9hG4bK-524287-1---adf26173531adb88;rport
Max-Forwards: 70
Contact: <sip:3028@xxx.xxx.xxx.xxx:51508;transport=UDP;rinstance=34563efa1eb36438>
To: <sip:3028@mydomain.noip.us:5089;transport=UDP>
From: <sip:3028@mydomain.noip.us:5089;transport=UDP>;tag=273a653a
Call-ID: UPpXzH9pJosLX-HtJ2DxnA..
CSeq: 7 REGISTER
Expires: 60
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Zoiper r30798
Authorization: Digest username="3028",realm="asterisk",nonce="1f8fe5bf",uri="sip:mydomain.noip.us:5089;transport=UDP",response="0bf89b9d7b56cabda6f80be178973a45",algorithm=MD5
Allow-Events: presence, kpml
Content-Length: 0

<------------->
--- (15 headers 0 lines) ---
Sending to xxx.xxx.xxx.xxx:51508 (no NAT)
Sending to xxx.xxx.xxx.xxx:51508 (no NAT)

<--- Transmitting (NAT) to xxx.xxx.xxx.xxx:51508 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:51508;branch=z9hG4bK-524287-1---adf26173531adb88;received=xxx.xxx.xxx.xxx;rport=51508
From: <sip:3028@mydomain.noip.us:5089;transport=UDP>;tag=273a653a
To: <sip:3028@mydomain.noip.us:5089;transport=UDP>;tag=as4bf4be2d
Call-ID: UPpXzH9pJosLX-HtJ2DxnA..
CSeq: 7 REGISTER
Server: Asterisk PBX 11.17.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4512deef"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'UPpXzH9pJosLX-HtJ2DxnA..' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:xxx.xxx.xxx.xxx:51508 --->
REGISTER sip:mydomain.noip.us:5089;transport=UDP SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:51508;branch=z9hG4bK-524287-1---a344623e434c16c2;rport
Max-Forwards: 70
Contact: <sip:3028@xxx.xxx.xxx.xxx:51508;transport=UDP;rinstance=34563efa1eb36438>
To: <sip:3028@mydomain.noip.us:5089;transport=UDP>
From: <sip:3028@mydomain.noip.us:5089;transport=UDP>;tag=273a653a
Call-ID: UPpXzH9pJosLX-HtJ2DxnA..
CSeq: 8 REGISTER
Expires: 60
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Zoiper r30798
Authorization: Digest username="3028",realm="asterisk",nonce="4512deef",uri="sip:mydomain.noip.us:5089;transport=UDP",response="f975c731194606c14561096b82bbb4d0",algorithm=MD5
Allow-Events: presence, kpml
Content-Length: 0

<------------->
--- (15 headers 0 lines) ---
Sending to xxx.xxx.xxx.xxx:51508 (no NAT)
Reliably Transmitting (NAT) to xxx.xxx.xxx.xxx:51508:
OPTIONS sip:3028@xxx.xxx.xxx.xxx:51508;transport=UDP;rinstance=34563efa1eb36438 SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.x.x:5089;branch=z9hG4bK03c3904f;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@xxx.xxx.x.x:5089>;tag=as7940564a
To: <sip:3028@xxx.xxx.xxx.xxx:51508;transport=UDP;rinstance=34563efa1eb36438>
Contact: <sip:asterisk@xxx.xxx.x.x:5089>
Call-ID: 1359e1cf0235650912c9d87620d84baa@xxx.xxx.x.x:5089
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.17.1
Date: Fri, 08 May 2015 18:44:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- Transmitting (NAT) to xxx.xxx.xxx.xxx:51508 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:51508;branch=z9hG4bK-524287-1---a344623e434c16c2;received=xxx.xxx.xxx.xxx;rport=51508
From: <sip:3028@mydomain.noip.us:5089;transport=UDP>;tag=273a653a
To: <sip:3028@mydomain.noip.us:5089;transport=UDP>;tag=as4bf4be2d
Call-ID: UPpXzH9pJosLX-HtJ2DxnA..
CSeq: 8 REGISTER
Server: Asterisk PBX 11.17.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 60
Contact: <sip:3028@xxx.xxx.xxx.xxx:51508;transport=UDP;rinstance=34563efa1eb36438>;expires=60
Date: Fri, 08 May 2015 18:44:57 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'UPpXzH9pJosLX-HtJ2DxnA..' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:xxx.xxx.xxx.xxx:51508 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP xxx.xxx.x.x:5089;branch=z9hG4bK03c3904f;rport=5089;received=xxx.xx.xxx.x
Contact: <sip:xxx.x.x.xxx:51508>
To: <sip:3028@xxx.xxx.xxx.xxx:51508;transport=UDP;rinstance=34563efa1eb36438>;tag=10709013
From: "asterisk" <sip:asterisk@xxx.xxx.x.x:5089>;tag=as7940564a
Call-ID: 1359e1cf0235650912c9d87620d84baa@xxx.xxx.x.x:5089
CSeq: 102 OPTIONS
Accept: application/sdp, application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
User-Agent: Zoiper r30798
Allow-Events: presence, kpml
Content-Length: 0

<------------->
--- (14 headers 0 lines) ---
Really destroying SIP dialog '1359e1cf0235650912c9d87620d84baa@xxx.xxx.x.x:5089' Method: OPTIONS

<--- SIP read from UDP:xxx.xxx.xxx.xxx:51508 --->


<------------->
Really destroying SIP dialog 'UPpXzH9pJosLX-HtJ2DxnA..' Method: REGISTER
Reliably Transmitting (NAT) to XXX.X.X.X:5089:
OPTIONS sip:3020@XXX.X.X.X:5089 SIP/2.0
Via: SIP/2.0/UDP XXX.X.X.X:5089;branch=z9hG4bK60c8ab9d;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@XXX.X.X.X:5089>;tag=as73db79cc
To: <sip:3020@XXX.X.X.X:5089>
Contact: <sip:asterisk@XXX.X.X.X:5089>
Call-ID: 5c8e5fce37f559520d9b41b25e39f037@XXX.X.X.X:5089
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.17.1
Date: Fri, 08 May 2015 18:45:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:XXX.X.X.X:5089 --->
SIP/2.0 200 OK
To: <sip:3020@XXX.X.X.X:5089>;tag=6d13a8d67eb6cda6i1
From: "asterisk" <sip:asterisk@XXX.X.X.X:5089>;tag=as73db79cc
Call-ID: 5c8e5fce37f559520d9b41b25e39f037@XXX.X.X.X:5089
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP XXX.X.X.X:5089;branch=z9hG4bK60c8ab9d;rport=5089
Server: Linksys/SPA8000-6.1.12(XU)
Allow-Events: talk, hold, conference
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces

<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '5c8e5fce37f559520d9b41b25e39f037@XXX.X.X.X:5089' Method: OPTIONS

<--- SIP read from UDP:xxx.xxx.xxx.xxx:51508 --->

How can I fix it?

2

The address used by Asterisk and that used by the phone need to be different. Currently they are both XXX.X.X.XX:5089.

“Probation passed - setting RTP source address” is a normal message. It indicates that nothing is wrong at that point. The actual error message is “Retransmission timeout reached on transmission Luu1OPUGYl60fdj3rodhzw… for seqno 2 (Critical Response)” and it is followed by a URL which you should have read before coming here.

Basically your system has sent OK to the phone, at XXX.X.X.XX:5089 telling it to acknowledge to XXX.X.X.XX:5089. That acknowledgement has never been received.

I’m guessing you have destroyed a vital distinction between these two addresses when sanitising your trace. The actual values of these addresses in relation to the specifics of your network are critical in debugging this.

3

Ok, but how can I fix it?

4

By ensuring that the ACK reaches Asterisk.

There aren’t simple fixes for things like this; you need to understand how your network is mis-configured, and that is not something that can be done on the information you have provided. In particular, you have provided no information about the network, and have removed any clues about it from the logs…

5

Most of the people tend to hide the IP address when posting their log files or cli output, but it just make more difficult to diagnose the issue.

Make sure you have setup your sip.conf file with the correct NAT setting.

Check the section called ----------------------------------------- NAT SUPPORT ------------------------ from the sample conf file
svn.digium.com/svn/asterisk/trun … onf.sample

6

[quote=“david55”]By ensuring that the ACK reaches Asterisk.

There aren’t simple fixes for things like this; you need to understand how your network is mis-configured, and that is not something that can be done on the information you have provided. In particular, you have provided no information about the network, and have removed any clues about it from the logs…[/quote]

OK if I show IP adress? Is possible to fix it?

7

Asterisk works. If you configure your network correctly and configure Asterisk to match your network, it should be OK. The IP addresses may give more clues, and people may be able to suggest things to try next, but this is best solved by actually knowing how your network is set up, and I think you need to hire someone who has the expertise to find that out. It is probably best if you find someone local, as doing this sort of thing over email is likely to be painful.

Note that the solution may lie outside of Asterisk, and the choice of solution certainly does.