Hi Matthew
Le 28/05/2012 19:28, Matthew J. Roth a écrit :
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.
We tested this setting this WE, effectively this problem disappear but
another appears: call get connected but no audio. We installed Asterisk
10.3.1 -> connection and no audio too, so same behaviour.
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.
We did read those files, don't see which parameter we could have forget.
media_address nor nat=comedia seems options for us. Hereunder a debug
from call with force_rport: as you can see, the RTP audio is coming from
another IP (77.77.777.77) We think asterisk doesn't accept this and
don't know which parameter could solve this.
<--- 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=z9hG4bK04e390b0;rport
From: "TOOTAi" <sip:0033333333333@222.222.22.22>;tag=as1335adb1
To: <sip:0000033666666666@111.111.1.111>;tag=4e0313ac670313ac4f9920c31847cea
Contact: sip:0000033666666666@111.111.1.111:5060
Call-ID: 72d5d3df06c07cc6037786ee59f574df@222.222.22.22:5060
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 1338276550 1338276550 IN IP4 77.77.777.77
s=SIP Call
c=IN IP4 77.77.777.77
t=0 0
m=audio 41462 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 - (ulaw|alaw), peer -
audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing),
combined - 0x0 (nothing)
Peer audio RTP is at port 77.77.777.77:41462
list_route: hop: <sip:0000033666666666@111.111.1.111:5060>
set_destination: Parsing <sip:0000033666666666@111.111.1.111:5060> for
address/port to send to
set_destination: set destination to 111.111.1.111:5060
Transmitting (NAT) to 111.111.1.111:5060:
ACK sip:0000033666666666@111.111.1.111:5060 SIP/2.0
Via: SIP/2.0/UDP 222.222.22.22:5060;branch=z9hG4bK0d106caa;rport
Max-Forwards: 70
From: "TOOTAi" <sip:0033333333333@222.222.22.22>;tag=as1335adb1
To: <sip:0000033666666666@111.111.1.111>;tag=4e0313ac670313ac4f9920c31847cea
Contact: <sip:0033333333333@222.222.22.22:5060>
Call-ID: 72d5d3df06c07cc6037786ee59f574df@222.222.22.22:5060
CSeq: 102 ACK
User-Agent: TOOTAiAudio
Content-Length: 0
---
-- SIP/myPeerDef-00000003 answered SIP/104-00000002
Thanks for your support.
--
Daniel
--
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