23 Oct
2007
23 Oct
'07
11:50 a.m.
Hello mates,
Is it possible to use A2bill to correctly obtain the CDRs and bill
OpenSER subscribers who only use Asterisk as a PSTN GTW in just a single
MySQL view without need of replicating them again into another Asterisk
MySQL database?
WBR,
LU.
On Tue, 2007-10-23 at 14:25 +0200, users-request(a)openser.org wrote:
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> Today's Topics:
>
> 1. "480 User not responding" instead of "408 Request Timeout"
> when modparam("tm", "noisy_ctimer", 1)? (I?aki Baz
Castillo)
> 2. INVITE relayed with missing bytes (Papadopoulos Georgios)
> 3. "480 User not responding" instead of "408 Request Timeout"
> when modparam("tm", "noisy_ctimer", 1)? (Juha Heinanen)
> 4. Re: "480 User not responding" instead of "408 Request
> Timeout" when modparam("tm", "noisy_ctimer", 1)?
(I?aki Baz Castillo)
> 5. Re: case sensitivity with avp_db_load (Jiri Kuthan)
> 6. Bug in 200 to CANCEL (wrong To_tag) (I?aki Baz Castillo)
> 7. Re: About "q" values (Klaus Darilion)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Tue, 23 Oct 2007 12:29:02 +0200
> From: I?aki Baz Castillo <ibc(a)aliax.net>
> Subject: [OpenSER-Users] "480 User not responding" instead of "408
> Request Timeout" when modparam("tm", "noisy_ctimer", 1)?
> To: users(a)openser.org
> Message-ID: <200710231229.02671.ibc(a)aliax.net>
> Content-Type: text/plain; charset="utf-8"
>
> Hi, if modparam("tm", "noisy_ctimer", 1) and INVITE exceded
"fr_inv_timer"
> then OpenSer sends "408 Request Timeout" to caller and CANCEL to called.
>
> I'm sure this is correct according to RFC, but wouldn't be better replying
> with "480 User not responding"?
>
> For example, Asterisk does nothing if it receives "408 Request Timeout" (it
> ignores it). I've reported about about it:
> http://bugs.digium.com/view.php?id=11058
>
> Sure it's a fail of Asterisk who shoud accept "408" and terminate the
call,
> but anyway, wouldn't be correct to reply with "480" instead of
"408"?
>
> Regards.
>
>
> --
> Iaki Baz Castillo
>
>
>
> ------------------------------
>
> Message: 2
> Date: Tue, 23 Oct 2007 13:36:02 +0300
> From: "Papadopoulos Georgios" <geop(a)altectelecoms.gr>
> Subject: [OpenSER-Users] INVITE relayed with missing bytes
> To: <users(a)openser.org>
> Message-ID:
> <9DB9DF2949D8774796CB054FBEC15699077F68F2(a)Tyran.int.acn.gr>
> Content-Type: text/plain; charset="us-ascii"
>
> Hello all,
>
> I am noticing that OpenSER relays an INVITE and the packet that is sent
> out is clipped. You can see in the following packets that the forwarded
> packet is missing a few bytes from the SDP part. Counting the bytes
> shows that the outgoing packet is always 1512 bytes. Is this something
> that has do with OpenSER (module parameter, compile options) or is it OS
> related?
>
> thank you for any help
>
> George
>
>
>
> U 2007/10/23 13:27:46.165232 213.5.1.6:57665 -> 213.5.43.4:5060
> INVITE sip:1012118204501@213.5.43.4:5060 SIP/2.0.
> Via: SIP/2.0/UDP
> 213.5.1.6:5060;x-route-tag="cid:CID-ACN-3@213.5.1.6";branch=z9hG4bK1594C
> 1AB9.
> Remote-Party-ID: "GEOP Papadopoul"
> <sip:2116872933@213.5.1.6>;party=calling;screen=no;privacy=off.
> From: "GEOP Papadopoul"
<sip:2116872933@213.5.1.6>;tag=237E8130-1821.
> To: <sip:1012118204501@213.5.43.4>.
> Date: Tue, 23 Oct 2007 10:27:46 GMT.
> Call-ID: 6E2DF805-808911DC-AA258232-F1669B65(a)213.5.1.6.
> Supported: 100rel,timer,resource-priority,replaces.
> Min-SE: 1800.
> Cisco-Guid: 1848010793-2156466652-3218142496-4228086245.
> User-Agent: Cisco-SIPGateway/IOS-12.x.
> Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
> SUBSCRIBE, NOTIFY, INFO, REGISTER.
> CSeq: 101 INVITE.
> Max-Forwards: 70.
> Timestamp: 1193135266.
> Contact: <sip:2116872933@213.5.1.6:5060>.
> Call-Info:
>
<sip:213.5.1.6:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
> .
> Expires: 180.
> Allow-Events: telephone-event.
> Content-Type: application/sdp.
> Content-Disposition: session;handling=required.
> Content-Length: 290.
> .
> v=0.
> o=CiscoSystemsSIP-GW-UserAgent 5211 453 IN IP4 213.5.1.6.
> s=SIP Call.
> c=IN IP4 213.5.1.6.
> t=0 0.
> m=audio 16454 RTP/AVP 18 0 8 100.
> c=IN IP4 213.5.1.6.
> a=rtpmap:18 G729/8000.
> a=fmtp:18 annexb=yes.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:100 X-NSE/8000.
> a=fmtp:100 192-194.
>
>
> U 2007/10/23 13:27:46.173826 213.5.43.4:5060 -> 213.5.1.6:57665
> SIP/2.0 100 Giving a try.
> Via: SIP/2.0/UDP
> 213.5.1.6:5060;x-route-tag="cid:CID-ACN-3@213.5.1.6";branch=z9hG4bK1594C
> 1AB9;rport=57665.
> From: "GEOP Papadopoul"
<sip:2116872933@213.5.1.6>;tag=237E8130-1821.
> To: <sip:1012118204501@213.5.43.4>.
> Call-ID: 6E2DF805-808911DC-AA258232-F1669B65(a)213.5.1.6.
> CSeq: 101 INVITE.
> Server: Altec Telecoms SIP Proxy.
> Content-Length: 0.
> .
>
>
> U 2007/10/23 13:27:46.174241 213.5.43.4:5060 -> 213.5.17.226:5060
> INVITE sip:demo1@192.168.1.39:5060 SIP/2.0.
> Record-Route: <sip:213.5.43.4;lr=on;ftag=237E8130-1821>.
> Via: SIP/2.0/UDP 213.5.43.4;branch=z9hG4bK4acd.5092baa7.0.
> Via: SIP/2.0/UDP
> 213.5.1.6:5060;rport=57665;x-route-tag="cid:CID-ACN-3@213.5.1.6";branch=
> z9hG4bK1594C1AB9.
> Remote-Party-ID: "GEOP Papadopoul"
> <sip:2116872933@213.5.1.6>;party=calling;screen=no;privacy=off.
> From: "GEOP Papadopoul"
<sip:2116872933@213.5.1.6>;tag=237E8130-1821.
> To: <sip:1012118204501@213.5.43.4>.
> Date: Tue, 23 Oct 2007 10:27:46 GMT.
> Call-ID: 6E2DF805-808911DC-AA258232-F1669B65(a)213.5.1.6.
> Supported: 100rel,timer,resource-priority,replaces.
> Min-SE: 1800.
> Cisco-Guid: 1848010793-2156466652-3218142496-4228086245.
> User-Agent: Cisco-SIPGateway/IOS-12.x.
> Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
> SUBSCRIBE, NOTIFY, INFO, REGISTER.
> CSeq: 101 INVITE.
> Max-Forwards: 10.
> Timestamp: 1193135266.
> Contact: <sip:2116872933@213.5.1.6:57665>.
> Call-Info:
>
<sip:213.5.1.6:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
> .
> Expires: 180.
> Allow-Events: telephone-event.
> Content-Type: application/sdp.
> Content-Disposition: session;handling=required.
> Content-Length: 290.
> P-hint: NATed client request.
> .
> v=0.
> o=CiscoSystemsSIP-GW-UserAgent 5211 453 IN IP4 213.5.1.6.
> s=SIP Call.
> c=IN IP4 213.5.1.6.
> t=0 0.
> m=audio 16454 RTP/AVP 18 0 8 100.
> c=IN IP4 213.5.1.6.
> a=rtpmap:18 G729/8000.
> a=fmtp:18 annexb=yes.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:100 X-NSE/8000.
> a=fmtp:100
>
>
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