Administrator TOOTAI
2012-May-27 17:11 UTC
[asterisk-users] Upgrade from version 1.6.24 to 1.8.12 - Retransmission timeout error
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
Matthew J. Roth
2012-May-28 17:28 UTC
[asterisk-users] Upgrade from version 1.6.24 to 1.8.12 - Retransmission timeout error
Administrator TOOTAI wrote:> 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:Asterisk 1.8.12 is not getting responses to the INVITES it sends. Comparing the INVITES, the only significant difference I see is that Asterisk 1.6.24 includes the "rport" field in the Via header and Asterisk 1.8.12 does not: 1.6.24 - Via: SIP/2.0/UDP 222.222.22.22:5060;branch=z9hG4bK58aef527;rport 1.8.12 - Via: SIP/2.0/UDP 222.222.22.22:5060;branch=z9hG4bK0c8907be Try setting "nat=force_rport" in sip.conf. Please reply back to the list with the results. There may be other differences between the versions that you haven't accounted for. Read the CHANGES and UPGRADE.txt files in the root of the Asterisk source tree for details. Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer