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@openser.org wrote:
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Today's Topics:
- "480 User not responding" instead of "408 Request Timeout" when modparam("tm", "noisy_ctimer", 1)? (I?aki Baz Castillo)
- INVITE relayed with missing bytes (Papadopoulos Georgios)
- "480 User not responding" instead of "408 Request Timeout" when modparam("tm", "noisy_ctimer", 1)? (Juha Heinanen)
- Re: "480 User not responding" instead of "408 Request Timeout" when modparam("tm", "noisy_ctimer", 1)? (I?aki Baz Castillo)
- Re: case sensitivity with avp_db_load (Jiri Kuthan)
- Bug in 200 to CANCEL (wrong To_tag) (I?aki Baz Castillo)
- Re: About "q" values (Klaus Darilion)
Message: 1 Date: Tue, 23 Oct 2007 12:29:02 +0200 From: I?aki Baz Castillo ibc@aliax.net Subject: [OpenSER-Users] "480 User not responding" instead of "408 Request Timeout" when modparam("tm", "noisy_ctimer", 1)? To: users@openser.org Message-ID: 200710231229.02671.ibc@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@altectelecoms.gr Subject: [OpenSER-Users] INVITE relayed with missing bytes To: users@openser.org Message-ID: 9DB9DF2949D8774796CB054FBEC15699077F68F2@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@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@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@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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