[asterisk-users] Upgrade from version 1.6.24 to 1.8.12 - Retransmission timeout error

Sun May 27 12:11:48 CDT 2012

Hi list,

we are upgrading our Asterisk production server from 1.6.24 to 1.8.12 
version and face the following problem: one of our peer 
(voicetrading.com) doesn't accept our calls anymore, we receive a 
timeout error "Packet timed out after 32000ms with no response".

Switching back to 1.6 make things working again!

In sip.conf we have nat=no, peer conf is:

[myPeerDef]
type=peer
host=111.111.1.111
context=honeypot 

insecure=invite 

directmedia=no 

disallow=all 

allow=ulaw,alaw 

dtmfmode=inband

We aren't registered, our IP is authorized by their system.

Debug of sessions (222.222.22.22 is our server 111.111.1.111 is their)

Working one with 1.6:

Audio is at 222.222.22.22 port 26002
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Reliably Transmitting (no NAT) to 111.111.1.111:5060:
INVITE sip:0000033666666666 at 111.111.1.111 SIP/2.0
Via: SIP/2.0/UDP 222.222.22.22:5060;branch=z9hG4bK58aef527;rport
Max-Forwards: 70
From: "TOOTAi" <sip:0033333333333 at 222.222.22.22>;tag=as52190c5c
To: <sip:0000033666666666 at 111.111.1.111>
Contact: <sip:0033333333333 at 222.222.22.22>
Call-ID: 2c974a0a2b08abe320ed388433e47d7e at 222.222.22.22
CSeq: 102 INVITE
User-Agent: TOOTAiAudio
Date: Sun, 27 May 2012 16:10:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 199

v=0
o=root 284043376 284043376 IN IP4 222.222.22.22
s=TOOTAiAudio PBX
c=IN IP4 222.222.22.22
t=0 0
m=audio 26002 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

---
     -- Called myPeerDef/0000033666666666

<--- SIP read from UDP:111.111.1.111:5060 --->
SIP/2.0 183 Session progress
Via: SIP/2.0/UDP 222.222.22.22:5060;branch=z9hG4bK58aef527;rport
From: "TOOTAi" <sip:0033333333333 at 222.222.22.22>;tag=as52190c5c
To: <sip:0000033666666666 at 111.111.1.111>;tag=4e0313ac670313ac4f9920c3173f554
Contact: sip:0000033666666666 at 111.111.1.111:5060
Call-ID: 2c974a0a2b08abe320ed388433e47d7e at 222.222.22.22
CSeq: 102 INVITE
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
Content-Type: application/sdp
Content-Length: 159

v=0
o=CARRIER 1338135040 1338135040 IN IP4 77.72.168.74
s=SIP Call
c=IN IP4 77.72.168.74
t=0 0
m=audio 18456 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20

<------------->
--- (11 headers 8 lines) ---
Found RTP audio format 0
Found audio description format PCMU for ID 0
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 
(nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), 
combined - 0x0 (nothing)
Peer audio RTP is at port 77.72.168.74:18456
Peer doesn't provide video
     -- SIP/myPeerDef-00000007 is making progress passing it to 
SIP/104-00000006

<--- SIP read from UDP:111.111.1.111:5060 --->
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 222.222.22.22:5060;branch=z9hG4bK58aef527;rport
From: "TOOTAi" <sip:0033333333333 at 222.222.22.22>;tag=as52190c5c
To: <sip:0000033666666666 at 111.111.1.111>;tag=4e0313ac670313ac4f9920c3173f554
Contact: sip:0000033666666666 at 111.111.1.111:5060
Call-ID: 2c974a0a2b08abe320ed388433e47d7e at 222.222.22.22
CSeq: 102 INVITE
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
Content-Type: application/sdp
Content-Length: 159

v=0
o=CARRIER 1338135052 1338135052 IN IP4 77.72.168.74
s=SIP Call
c=IN IP4 77.72.168.74
t=0 0
m=audio 18456 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20

<------------->
--- (11 headers 8 lines) ---
Found RTP audio format 0
Found audio description format PCMU for ID 0
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 
(nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), 
combined - 0x0 (nothing)
Peer audio RTP is at port 77.72.168.74:18456
Peer doesn't provide video
list_route: hop: <sip:0000033666666666 at 111.111.1.111:5060>
set_destination: Parsing <sip:0000033666666666 at 111.111.1.111:5060> for 
address/port to send to
set_destination: set destination to 111.111.1.111, port 5060
Transmitting (no NAT) to 111.111.1.111:5060:
ACK sip:0000033666666666 at 111.111.1.111:5060 SIP/2.0
Via: SIP/2.0/UDP 222.222.22.22:5060;branch=z9hG4bK5afa4cc0;rport
Max-Forwards: 70
From: "TOOTAi" <sip:0033333333333 at 222.222.22.22>;tag=as52190c5c
To: <sip:0000033666666666 at 111.111.1.111>;tag=4e0313ac670313ac4f9920c3173f554
Contact: <sip:0033333333333 at 222.222.22.22>
Call-ID: 2c974a0a2b08abe320ed388433e47d7e at 222.222.22.22
CSeq: 102 ACK
User-Agent: TOOTAiAudio
Content-Length: 0

---
     -- SIP/myPeerDef-00000007 answered SIP/104-00000006
Scheduling destruction of SIP dialog 
'2c974a0a2b08abe320ed388433e47d7e at 222.222.22.22' in 32000 ms (Method: 
INVITE)
set_destination: Parsing <sip:0000033666666666 at 111.111.1.111:5060> for 
address/port to send to
set_destination: set destination to 111.111.1.111, port 5060
Reliably Transmitting (no NAT) to 111.111.1.111:5060:
BYE sip:0000033666666666 at 111.111.1.111:5060 SIP/2.0
Via: SIP/2.0/UDP 222.222.22.22:5060;branch=z9hG4bK2e708816;rport
Max-Forwards: 70
From: "TOOTAi" <sip:0033333333333 at 222.222.22.22>;tag=as52190c5c
To: <sip:0000033666666666 at 111.111.1.111>;tag=4e0313ac670313ac4f9920c3173f554
Call-ID: 2c974a0a2b08abe320ed388433e47d7e at 222.222.22.22
CSeq: 103 BYE
User-Agent: TOOTAiAudio
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

 From 1.8 not working:

Audio is at 26704
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Reliably Transmitting (no NAT) to 111.111.1.111:5060:
INVITE sip:0000033666666666 at 111.111.1.111 SIP/2.0
Via: SIP/2.0/UDP 222.222.22.22:5060;branch=z9hG4bK0c8907be
Max-Forwards: 70
From: "TOOTAi" <sip:0033333333333 at 222.222.22.22>;tag=as61c0d9a7
To: <sip:0000033666666666 at 111.111.1.111>
Contact: <sip:0033333333333 at 222.222.22.22:5060>
Call-ID: 2f80a1b0013b3c9200ad6cb1178f7e9b at 222.222.22.22:5060
CSeq: 102 INVITE
User-Agent: TOOTAiAudio
Date: Sun, 27 May 2012 16:14:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, 
INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 199

v=0
o=root 298186421 298186421 IN IP4 222.222.22.22
s=TOOTAiAudio PBX
c=IN IP4 222.222.22.22
t=0 0
m=audio 26704 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

---
     -- Called SIP/myPeerDef/0000033666666666
Retransmitting #1 (no NAT) to 111.111.1.111:5060:
INVITE sip:0000033666666666 at 111.111.1.111 SIP/2.0
Via: SIP/2.0/UDP 222.222.22.22:5060;branch=z9hG4bK0c8907be
Max-Forwards: 70
From: "TOOTAi" <sip:0033333333333 at 222.222.22.22>;tag=as61c0d9a7
To: <sip:0000033666666666 at 111.111.1.111>
Contact: <sip:0033333333333 at 222.222.22.22:5060>
Call-ID: 2f80a1b0013b3c9200ad6cb1178f7e9b at 222.222.22.22:5060
CSeq: 102 INVITE
User-Agent: TOOTAiAudio
Date: Sun, 27 May 2012 16:14:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, 
INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 199

v=0
o=root 298186421 298186421 IN IP4 222.222.22.22
s=TOOTAiAudio PBX
c=IN IP4 222.222.22.22
t=0 0
m=audio 26704 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

---
Retransmitting #2 (no NAT) to 111.111.1.111:5060:
INVITE sip:0000033666666666 at 111.111.1.111 SIP/2.0
Via: SIP/2.0/UDP 222.222.22.22:5060;branch=z9hG4bK0c8907be
Max-Forwards: 70
From: "TOOTAi" <sip:0033333333333 at 222.222.22.22>;tag=as61c0d9a7
To: <sip:0000033666666666 at 111.111.1.111>
Contact: <sip:0033333333333 at 222.222.22.22:5060>
Call-ID: 2f80a1b0013b3c9200ad6cb1178f7e9b at 222.222.22.22:5060
CSeq: 102 INVITE
User-Agent: TOOTAiAudio
Date: Sun, 27 May 2012 16:14:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, 
INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 199

v=0
o=root 298186421 298186421 IN IP4 222.222.22.22
s=TOOTAiAudio PBX
c=IN IP4 222.222.22.22
t=0 0
m=audio 26704 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

---
Retransmitting #3 (no NAT) to 111.111.1.111:5060:
INVITE sip:0000033666666666 at 111.111.1.111 SIP/2.0
Via: SIP/2.0/UDP 222.222.22.22:5060;branch=z9hG4bK0c8907be
Max-Forwards: 70
From: "TOOTAi" <sip:0033333333333 at 222.222.22.22>;tag=as61c0d9a7
To: <sip:0000033666666666 at 111.111.1.111>
Contact: <sip:0033333333333 at 222.222.22.22:5060>
Call-ID: 2f80a1b0013b3c9200ad6cb1178f7e9b at 222.222.22.22:5060
CSeq: 102 INVITE
User-Agent: TOOTAiAudio
Date: Sun, 27 May 2012 16:14:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, 
INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 199

v=0
o=root 298186421 298186421 IN IP4 222.222.22.22
s=TOOTAiAudio PBX
c=IN IP4 222.222.22.22
t=0 0
m=audio 26704 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

---
Retransmitting #4 (no NAT) to 111.111.1.111:5060:
INVITE sip:0000033666666666 at 111.111.1.111 SIP/2.0
Via: SIP/2.0/UDP 222.222.22.22:5060;branch=z9hG4bK0c8907be
Max-Forwards: 70
From: "TOOTAi" <sip:0033333333333 at 222.222.22.22>;tag=as61c0d9a7
To: <sip:0000033666666666 at 111.111.1.111>
Contact: <sip:0033333333333 at 222.222.22.22:5060>
Call-ID: 2f80a1b0013b3c9200ad6cb1178f7e9b at 222.222.22.22:5060
CSeq: 102 INVITE
User-Agent: TOOTAiAudio
Date: Sun, 27 May 2012 16:14:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, 
INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 199

v=0
o=root 298186421 298186421 IN IP4 222.222.22.22
s=TOOTAiAudio PBX
c=IN IP4 222.222.22.22
t=0 0
m=audio 26704 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

---
Retransmitting #5 (no NAT) to 111.111.1.111:5060:
INVITE sip:0000033666666666 at 111.111.1.111 SIP/2.0
Via: SIP/2.0/UDP 222.222.22.22:5060;branch=z9hG4bK0c8907be
Max-Forwards: 70
From: "TOOTAi" <sip:0033333333333 at 222.222.22.22>;tag=as61c0d9a7
To: <sip:0000033666666666 at 111.111.1.111>
Contact: <sip:0033333333333 at 222.222.22.22:5060>
Call-ID: 2f80a1b0013b3c9200ad6cb1178f7e9b at 222.222.22.22:5060
CSeq: 102 INVITE
User-Agent: TOOTAiAudio
Date: Sun, 27 May 2012 16:14:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, 
INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 199

v=0
o=root 298186421 298186421 IN IP4 222.222.22.22
s=TOOTAiAudio PBX
c=IN IP4 222.222.22.22
t=0 0
m=audio 26704 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

Thanks for any hint.

-- 
Daniel