sr-users September 2005

sr-users@lists.kamailio.org

231 participants
507 discussions

by Bogdan-Andrei Iancu

Hi everybody, I just committed on CVS full support for per-branch flags. What's the difference between the flags we already have and these new one? So far we ave in OpenSER flags that will be called "global" flags - global because they were visible everywhere in the transaction (in all routes and in all sequential replies/request). The new flags, called "branch" flags are flags that will be visible only for a specific branch - in all replies and routes connected to this branch. For example: I have a call parallel forking to GW and to a user. And I would like to know from which branch I will get the final negative reply (if so). I will set a branch route before relaying the calls (with the 2 branches). The branch route will be separately executed for each branch; in the branch going to GW (I can identified it by looking to RURI), I will set a branch flag. This flag will appear only in the onreply route run for replied from GW. It will be also be visible in failure route if the final elected reply belongs to the GW branch. This flags will not be visible in the other branch (in routes executing replies from the other branch). how to use it? The request flags may be split in two arrays - globals and branch flags - by using the TM parameter "branch_flag_mask" (see http://www.openser.org/docs/modules/0.10.x/tm.html#AEN268). Flags covered by this mask will be processed as branch flags, the rest of them as global flags. Also, modules may set branch flags before transaction creation (for the moment this feature is not available in script). The REGISTRAR module is the first to use this feature. If the "use_branch_flags" param is enabled (see http://www.openser.org/docs/modules/0.10.x/registrar.html#AEN236), the NAT flag will be push in branch flags instead in message flags - IMPORTANT: be sure that NAT flag is in the range of the branch flags ad defined in TM. Using this, NAT traversal per branch may be experimented.... regards, bogdan

18 years, 8 months

[Serusers] NAT Related Voicemail Timeout Problem (HELP!)

by Corey S. McFadden

We're working through an annoying problem with voicemail and the fr_inv_timer and after banging our heads against it for a while figured we'd reach out for a little help. It's probably something obvious, but here's the scenario: 1. SIP DID rings SER 2. SER Routes to registered UA (UA behind NAT) 3. UA Rings 4. fr_inv_timer is hit and failure_route[1] is invoked 5. failure_route[1] rewrites host to send call to Asterisk for VM 6. Asterisk gets the call and starts processing but the calling party never gets the OK, gets no audio, and dies with a fast busy. When the UA is NOT registered or registered without going through NAT there isn't a problem. Asterisk OK's and works fine. This is probably some moronic NAT issue but we're not sure how to proceed. I'm going to follow with a 'sip debug' message from Asterisk, some ngrep from the SER machine (with commentary), and finally the SER cnf itself. If anyone can help we would be incredibly grateful! (Seriously!) Thanks, -Corey ** IP Legend xxx.xxx.xxx.7 - SIP Originating Source xxx.xxx.xxx.36 - SER xxx.xxx.xxx.80 - Asterisk VM Server 216.NAT.CLI.207 - NAT UA Public IP 192.168.249.83 - Private IP of UA 5410000000 - DID & UA ID 6100000000 - Outside Caller ID ** Versions SER: 0.9.4-rc3 rtpproxy: 20040107 Extension 20050322 ** Asterisk SIP Debug shows this on the second retry: Retransmitting #1 (no NAT) to xxx.xxx.xxx.36:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP xxx.xxx.xxx.36;branch=z9hG4bK9e09.116856a3.2 Via: SIP/2.0/UDP xxx.xxx.xxx.7:5060;branch=z9hG4bK-00101-5Jn7520l3PhAH6h-0 Record-Route: <sip:xxx.xxx.xxx.36;ftag=5Jn7520l3PhAH6h-IPTrunk-81-17-23atxxx.xxx.xxx.7;lr=on> From: sip:6100000000@xxx.xxx.xxx.7:5060;tag=5Jn7520l3PhAH6h-IPTrunk-81-17-23atxxx.xxx.xxx.7 To: sip:5410000000@xxx.xxx.xxx.36:5060;tag=as625cebc0 Call-ID: 5Jn7520l3PhAH6h(a)xxx.xxx.xxx.7 CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: <sip:5410000000@xxx.xxx.xxx.80> Content-Type: application/sdp Content-Length: 237 v=0 o=root 29275 29275 IN IP4 xxx.xxx.xxx.80 s=session c=IN IP4 xxx.xxx.xxx.80 t=0 0 m=audio 11476 RTP/AVP 0 3 8 97 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 iLBC/8000 a=silenceSupp:off - - - - After re-transmitting one more time it will finally fail. *** NGREP OUTPUT (from SER box) # U xxx.xxx.xxx.7:5060 -> xxx.xxx.xxx.36:5060 INVITE sip:5410000000@xxx.xxx.xxx.36:5060 SIP/2.0..Via: SIP/2.0/UDP ... # U xxx.xxx.xxx.36:5060 -> xxx.xxx.xxx.7:5060 SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP ... # U xxx.xxx.xxx.36:5060 -> 216.NAT.CLI.207:5060 INVITE sip:5410000000@192.168.249.83:5060 SIP/2.0..Record-Route: <sip:xxx.xxx.xxx.36;ftag=kYn356Vl3PhGzel-IPTrunk-83-17-23a ... # U xxx.xxx.xxx.36:5060 -> 216.NAT.CLI.207:5060 INVITE sip:5410000000@216.NAT.CLI.207:5060 SIP/2.0..Record-Route: <sip:xxx.xxx.xxx.36;ftag=kYn356Vl3PhGzel-IPTrunk-83-17-23at xxx.xxx.xxx.7;lr=on>..Via: SIP/2.0/UDP ... # U 216.NAT.CLI.207:5060 -> xxx.xxx.xxx.36:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP ... # U 216.NAT.CLI.207:5060 -> xxx.xxx.xxx.36:5060 SIP/2.0 180 Ringing..Via: SIP/2.0/UDP ... # U xxx.xxx.xxx.36:5060 -> xxx.xxx.xxx.7:5060 SIP/2.0 180 Ringing..Via: SIP/2.0/UDP ... # U xxx.xxx.xxx.36:5060 -> 216.NAT.CLI.207:5060 INVITE sip:5410000000@216.NAT.CLI.207:5060 SIP/2.0..Record-Route: ... # U xxx.xxx.xxx.36:5060 -> 216.NAT.CLI.207:5060 INVITE sip:5410000000@216.NAT.CLI.207:5060 SIP/2.0..Record-Route: <sip:xxx.xxx.xxx.36;ftag=kYn356Vl3PhGzel-IPTrunk-83-17-23at xxx.xxx.xxx.7;lr=on>..Via: SIP/2.0/UDP ... # U xxx.xxx.xxx.36:5060 -> 216.NAT.CLI.207:5060 .... # U xxx.xxx.xxx.36:5060 -> 216.NAT.CLI.207:5060 INVITE sip:5410000000@216.NAT.CLI.207:5060 SIP/2.0..Record-Route: <sip:xxx.xxx.xxx.36;ftag=kYn356Vl3PhGzel-IPTrunk-83-17-23at ... # U xxx.xxx.xxx.36:5060 -> 216.NAT.CLI.207:5060 CANCEL sip:5410000000@192.168.249.83:5060 SIP/2.0..Via: SIP/2.0/UDP xxx.xxx.xxx.36;branch=z9hG4bKe2fe.1754f177.0..From: sip ... # U 216.NAT.CLI.207:5060 -> xxx.xxx.xxx.36:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP ... # U 216.NAT.CLI.207:5060 -> xxx.xxx.xxx.36:5060 SIP/2.0 487 Request Cancelled..Via: SIP/2.0/UDP ... # U xxx.xxx.xxx.36:5060 -> 216.NAT.CLI.207:5060 ACK sip:5410000000@192.168.249.83:5060 SIP/2.0..Via: SIP/2.0/UDP ... # U xxx.xxx.xxx.36:5060 -> 216.NAT.CLI.207:5060 INVITE sip:5410000000@216.NAT.CLI.207:5060 SIP/2.0..Record-Route: <sip:xxx.xxx.xxx.36;ftag=kYn356Vl3PhGzel-IPTrunk-83-17-23at xxx.xxx.xxx.7;lr=on>..Via: SIP/2.0/UDP ... # U 216.NAT.CLI.207:5060 -> xxx.xxx.xxx.36:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP ... # U 216.NAT.CLI.207:5060 -> xxx.xxx.xxx.36:5060 SIP/2.0 180 Ringing..Via: SIP/2.0/UDP ... # U xxx.xxx.xxx.36:5060 -> xxx.xxx.xxx.7:5060 SIP/2.0 180 Ringing..Via: SIP/2.0/UDP ... # (ASTERISK SERVER ENTERS THE PICTURE HERE) U xxx.xxx.xxx.36:5060 -> xxx.xxx.xxx.80:5060 INVITE sip:5410000000@xxx.xxx.xxx.80 SIP/2.0..Record-Route: <sip:xxx.xxx.xxx.36;ftag=kYn356Vl3PhGzel-IPTrunk-83-17-23at209. 216.177.7;lr=on>..Via: SIP/2.0/UDP xxx.xxx.xxx.36;branch=z9hG4bKe2fe.1754f177.2..Via: SIP/2.0/UDP xxx.xxx.xxx.7:5060;rport= 5060;branch=z9hG4bK-00101-kYn356Vl3PhGzel-0..Max-Forwards: 5..From: sip:6100000000@xxx.xxx.xxx.7:5060;tag=kYn356Vl3PhGzel-I PTrunk-83-17-23atxxx.xxx.xxx.7..To: sip:5410000000@xxx.xxx.xxx.36:5060..Call-ID: kYn356Vl3PhGzel@xxx.xxx.xxx.7..CSeq: 101 I NVITE..Remote-Party-Id: <sip:6100000000@xxx.xxx.xxx.7:5060>;party=calling;privacy=off;id-type=subscriber..Expires: 180..All ow: INVITE,CANCEL,BYE,ACK..Contact: sip:xxx.xxx.xxx.7:5060..User-Agent: Tekelec-7000/r4.0..Content-Type: application/sdp..C ontent-Length: 169....v=0..o=- 1 1 IN IP4 xxx.xxx.xxx.7..s= ..c=IN IP4 xxx.xxx.xxx.36..t=0 0..m=audio 35314 RTP/AVP 0..a=pt ime:20..a=rtpmap:0 PCMU/8000..a=direction:active..a=nortpproxy:yes.. # U xxx.xxx.xxx.36:5060 -> 216.NAT.CLI.207:5060 CANCEL sip:5410000000@216.NAT.CLI.207:5060 SIP/2.0..Via: SIP/2.0/UDP ... # (ASTERISK SERVER RESPONDS) U xxx.xxx.xxx.80:5060 -> xxx.xxx.xxx.36:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP xxx.xxx.xxx.36;branch=z9hG4bKe2fe.1754f177.2..Via: SIP/2.0/UDP xxx.xxx.xxx.7:5060;bran ch=z9hG4bK-00101-kYn356Vl3PhGzel-0..From: sip:6100000000@xxx.xxx.xxx.7:5060;tag=kYn356Vl3PhGzel-IPTrunk-83-17-23at209.216.1 77.7..To: sip:5410000000@xxx.xxx.xxx.36:5060..Call-ID: kYn356Vl3PhGzel@xxx.xxx.xxx.7..CSeq: 101 INVITE..User-Agent: Asteris k PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY..Contact: <sip:5410000000@xxx.xxx.xxx.80>..Content-Length: 0 .... # U 216.NAT.CLI.207:5060 -> xxx.xxx.xxx.36:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP ... # U 216.NAT.CLI.207:5060 -> xxx.xxx.xxx.36:5060 SIP/2.0 487 Request Cancelled..Via: SIP/2.0/UDP ... # (SER ACKs the UAs' 487) U xxx.xxx.xxx.36:5060 -> 216.NAT.CLI.207:5060 ACK sip:5410000000@216.NAT.CLI.207:5060 SIP/2.0..Via: SIP/2.0/UDP ... # (ASTERISK SENDS THE OK) U xxx.xxx.xxx.80:5060 -> xxx.xxx.xxx.36:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP xxx.xxx.xxx.36;branch=z9hG4bKe2fe.1754f177.2..Via: SIP/2.0/UDP xxx.xxx.xxx.7:5060;branch=z 9hG4bK-00101-kYn356Vl3PhGzel-0..Record-Route: <sip:xxx.xxx.xxx.36;ftag=kYn356Vl3PhGzel-IPTrunk-83-17-23atxxx.xxx.xxx.7;lr=o n>..From: sip:6100000000@xxx.xxx.xxx.7:5060;tag=kYn356Vl3PhGzel-IPTrunk-83-17-23atxxx.xxx.xxx.7..To: sip:5410000000@209.216 .160.36:5060;tag=as6ca38bc6..Call-ID: kYn356Vl3PhGzel@xxx.xxx.xxx.7..CSeq: 101 INVITE..User-Agent: Asterisk PBX..Allow: INV ITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY..Contact: <sip:5410000000@xxx.xxx.xxx.80>..Content-Type: application/sdp..Con tent-Length: 237....v=0..o=root 29275 29275 IN IP4 xxx.xxx.xxx.80..s=session..c=IN IP4 xxx.xxx.xxx.80..t=0 0..m=audio 11184 RTP/AVP 0 3 8 97..a=rtpmap:0 PCMU/8000..a=rtpmap:3 GSM/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:97 iLBC/8000..a=silenceSupp:of f - - - -.. # U xxx.xxx.xxx.36:5060 -> xxx.xxx.xxx.7:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP xxx.xxx.xxx.7:5060;branch=z9hG4bK-00101-kYn356Vl3PhGzel-0..Record-Route: <sip:209.216.160. 36;ftag=kYn356Vl3PhGzel-IPTrunk-83-17-23atxxx.xxx.xxx.7;lr=on>..From: sip:6100000000@xxx.xxx.xxx.7:5060;tag=kYn356Vl3PhGzel -IPTrunk-83-17-23atxxx.xxx.xxx.7..To: sip:5410000000@xxx.xxx.xxx.36:5060;tag=as6ca38bc6..Call-ID: kYn356Vl3PhGzel(a)209.216.1 77.7..CSeq: 101 INVITE..User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY..Contact: <sip:54 14362504(a)xxx.xxx.xxx.80>..Content-Type: application/sdp..Content-Length: 255....v=0..o=root 29275 29275 IN IP4 209.216.160. 80..s=session..c=IN IP4 xxx.xxx.xxx.36..t=0 0..m=audio 35316 RTP/AVP 0 3 8 97..a=rtpmap:0 PCMU/8000..a=rtpmap:3 GSM/8000..a =rtpmap:8 PCMA/8000..a=rtpmap:97 iLBC/8000..a=silenceSupp:off - - - -..a=nortpproxy:yes.. # (ASTERISK SENDS ANOTHER OK) U xxx.xxx.xxx.80:5060 -> xxx.xxx.xxx.36:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP xxx.xxx.xxx.36;branch=z9hG4bKe2fe.1754f177.2..Via: SIP/2.0/UDP xxx.xxx.xxx.7:5060;branch=z 9hG4bK-00101-kYn356Vl3PhGzel-0..Record-Route: <sip:xxx.xxx.xxx.36;ftag=kYn356Vl3PhGzel-IPTrunk-83-17-23atxxx.xxx.xxx.7;lr=o n>..From: sip:6100000000@xxx.xxx.xxx.7:5060;tag=kYn356Vl3PhGzel-IPTrunk-83-17-23atxxx.xxx.xxx.7..To: sip:5410000000@209.216 .160.36:5060;tag=as6ca38bc6..Call-ID: kYn356Vl3PhGzel@xxx.xxx.xxx.7..CSeq: 101 INVITE..User-Agent: Asterisk PBX..Allow: INV ITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY..Contact: <sip:5410000000@xxx.xxx.xxx.80>..Content-Type: application/sdp..Con tent-Length: 237....v=0..o=root 29275 29275 IN IP4 xxx.xxx.xxx.80..s=session..c=IN IP4 xxx.xxx.xxx.80..t=0 0..m=audio 11184 RTP/AVP 0 3 8 97..a=rtpmap:0 PCMU/8000..a=rtpmap:3 GSM/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:97 iLBC/8000..a=silenceSupp:of f - - - -.. # U xxx.xxx.xxx.36:5060 -> xxx.xxx.xxx.7:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP xxx.xxx.xxx.7:5060;branch=z9hG4bK-00101-kYn356Vl3PhGzel-0..Record-Route: <sip:209.216.160. 36;ftag=kYn356Vl3PhGzel-IPTrunk-83-17-23atxxx.xxx.xxx.7;lr=on>..From: sip:6100000000@xxx.xxx.xxx.7:5060;tag=kYn356Vl3PhGzel -IPTrunk-83-17-23atxxx.xxx.xxx.7..To: sip:5410000000@xxx.xxx.xxx.36:5060;tag=as6ca38bc6..Call-ID: kYn356Vl3PhGzel(a)209.216.1 77.7..CSeq: 101 INVITE..User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY..Contact: <sip:54 14362504(a)xxx.xxx.xxx.80>..Content-Type: application/sdp..Content-Length: 255....v=0..o=root 29275 29275 IN IP4 209.216.160. 80..s=session..c=IN IP4 xxx.xxx.xxx.36..t=0 0..m=audio 35316 RTP/AVP 0 3 8 97..a=rtpmap:0 PCMU/8000..a=rtpmap:3 GSM/8000..a =rtpmap:8 PCMA/8000..a=rtpmap:97 iLBC/8000..a=silenceSupp:off - - - -..a=nortpproxy:yes.. # (ASTERISK SENDS YET ANOTHER OK) U xxx.xxx.xxx.80:5060 -> xxx.xxx.xxx.36:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP xxx.xxx.xxx.36;branch=z9hG4bKe2fe.1754f177.2..Via: SIP/2.0/UDP xxx.xxx.xxx.7:5060;branch=z 9hG4bK-00101-kYn356Vl3PhGzel-0..Record-Route: <sip:xxx.xxx.xxx.36;ftag=kYn356Vl3PhGzel-IPTrunk-83-17-23atxxx.xxx.xxx.7;lr=o n>..From: sip:6100000000@xxx.xxx.xxx.7:5060;tag=kYn356Vl3PhGzel-IPTrunk-83-17-23atxxx.xxx.xxx.7..To: sip:5410000000@209.216 .160.36:5060;tag=as6ca38bc6..Call-ID: kYn356Vl3PhGzel@xxx.xxx.xxx.7..CSeq: 101 INVITE..User-Agent: Asterisk PBX..Allow: INV ITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY..Contact: <sip:5410000000@xxx.xxx.xxx.80>..Content-Type: application/sdp..Con tent-Length: 237....v=0..o=root 29275 29275 IN IP4 xxx.xxx.xxx.80..s=session..c=IN IP4 xxx.xxx.xxx.80..t=0 0..m=audio 11184 RTP/AVP 0 3 8 97..a=rtpmap:0 PCMU/8000..a=rtpmap:3 GSM/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:97 iLBC/8000..a=silenceSupp:of f - - - -.. # U xxx.xxx.xxx.36:5060 -> xxx.xxx.xxx.7:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP xxx.xxx.xxx.7:5060;branch=z9hG4bK-00101-kYn356Vl3PhGzel-0..Record-Route: <sip:209.216.160. 36;ftag=kYn356Vl3PhGzel-IPTrunk-83-17-23atxxx.xxx.xxx.7;lr=on>..From: sip:6100000000@xxx.xxx.xxx.7:5060;tag=kYn356Vl3PhGzel -IPTrunk-83-17-23atxxx.xxx.xxx.7..To: sip:5410000000@xxx.xxx.xxx.36:5060;tag=as6ca38bc6..Call-ID: kYn356Vl3PhGzel(a)209.216.1 77.7..CSeq: 101 INVITE..User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY..Contact: <sip:54 14362504(a)xxx.xxx.xxx.80>..Content-Type: application/sdp..Content-Length: 255....v=0..o=root 29275 29275 IN IP4 209.216.160. 80..s=session..c=IN IP4 xxx.xxx.xxx.36..t=0 0..m=audio 35316 RTP/AVP 0 3 8 97..a=rtpmap:0 PCMU/8000..a=rtpmap:3 GSM/8000..a =rtpmap:8 PCMA/8000..a=rtpmap:97 iLBC/8000..a=silenceSupp:off - - - -..a=nortpproxy:yes.. # (ASTERISK SENDS ANOTHER OK) U xxx.xxx.xxx.80:5060 -> xxx.xxx.xxx.36:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP xxx.xxx.xxx.36;branch=z9hG4bKe2fe.1754f177.2..Via: SIP/2.0/UDP xxx.xxx.xxx.7:5060;branch=z 9hG4bK-00101-kYn356Vl3PhGzel-0..Record-Route: <sip:xxx.xxx.xxx.36;ftag=kYn356Vl3PhGzel-IPTrunk-83-17-23atxxx.xxx.xxx.7;lr=o n>..From: sip:6100000000@xxx.xxx.xxx.7:5060;tag=kYn356Vl3PhGzel-IPTrunk-83-17-23atxxx.xxx.xxx.7..To: sip:5410000000@209.216 .160.36:5060;tag=as6ca38bc6..Call-ID: kYn356Vl3PhGzel@xxx.xxx.xxx.7..CSeq: 101 INVITE..User-Agent: Asterisk PBX..Allow: INV ITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY..Contact: <sip:5410000000@xxx.xxx.xxx.80>..Content-Type: application/sdp..Con tent-Length: 237....v=0..o=root 29275 29275 IN IP4 xxx.xxx.xxx.80..s=session..c=IN IP4 xxx.xxx.xxx.80..t=0 0..m=audio 11184 RTP/AVP 0 3 8 97..a=rtpmap:0 PCMU/8000..a=rtpmap:3 GSM/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:97 iLBC/8000..a=silenceSupp:of f - - - -.. # U xxx.xxx.xxx.36:5060 -> xxx.xxx.xxx.7:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP xxx.xxx.xxx.7:5060;branch=z9hG4bK-00101-kYn356Vl3PhGzel-0..Record-Route: <sip:209.216.160. 36;ftag=kYn356Vl3PhGzel-IPTrunk-83-17-23atxxx.xxx.xxx.7;lr=on>..From: sip:6100000000@xxx.xxx.xxx.7:5060;tag=kYn356Vl3PhGzel -IPTrunk-83-17-23atxxx.xxx.xxx.7..To: sip:5410000000@xxx.xxx.xxx.36:5060;tag=as6ca38bc6..Call-ID: kYn356Vl3PhGzel(a)209.216.1 77.7..CSeq: 101 INVITE..User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY..Contact: <sip:54 14362504(a)xxx.xxx.xxx.80>..Content-Type: application/sdp..Content-Length: 237....v=0..o=root 29275 29275 IN IP4 209.216.160. 80..s=session..c=IN IP4 xxx.xxx.xxx.80..t=0 0..m=audio 11184 RTP/AVP 0 3 8 97..a=rtpmap:0 PCMU/8000..a=rtpmap:3 GSM/8000..a =rtpmap:8 PCMA/8000..a=rtpmap:97 iLBC/8000..a=silenceSupp:off - - - -.. # ASTERISK KEEPS TRYING TO SEND OK U xxx.xxx.xxx.80:5060 -> xxx.xxx.xxx.36:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP xxx.xxx.xxx.36;branch=z9hG4bKe2fe.1754f177.2..Via: SIP/2.0/UDP xxx.xxx.xxx.7:5060;branch=z 9hG4bK-00101-kYn356Vl3PhGzel-0..Record-Route: <sip:xxx.xxx.xxx.36;ftag=kYn356Vl3PhGzel-IPTrunk-83-17-23atxxx.xxx.xxx.7;lr=o n>..From: sip:6100000000@xxx.xxx.xxx.7:5060;tag=kYn356Vl3PhGzel-IPTrunk-83-17-23atxxx.xxx.xxx.7..To: sip:5410000000@209.216 .160.36:5060;tag=as6ca38bc6..Call-ID: kYn356Vl3PhGzel@xxx.xxx.xxx.7..CSeq: 101 INVITE..User-Agent: Asterisk PBX..Allow: INV ITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY..Contact: <sip:5410000000@xxx.xxx.xxx.80>..Content-Type: application/sdp..Con tent-Length: 237....v=0..o=root 29275 29275 IN IP4 xxx.xxx.xxx.80..s=session..c=IN IP4 xxx.xxx.xxx.80..t=0 0..m=audio 11184 RTP/AVP 0 3 8 97..a=rtpmap:0 PCMU/8000..a=rtpmap:3 GSM/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:97 iLBC/8000..a=silenceSupp:of f - - - -.. # U xxx.xxx.xxx.36:5060 -> xxx.xxx.xxx.7:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP xxx.xxx.xxx.7:5060;branch=z9hG4bK-00101-kYn356Vl3PhGzel-0..Record-Route: <sip:209.216.160. 36;ftag=kYn356Vl3PhGzel-IPTrunk-83-17-23atxxx.xxx.xxx.7;lr=on>..From: sip:6100000000@xxx.xxx.xxx.7:5060;tag=kYn356Vl3PhGzel -IPTrunk-83-17-23atxxx.xxx.xxx.7..To: sip:5410000000@xxx.xxx.xxx.36:5060;tag=as6ca38bc6..Call-ID: kYn356Vl3PhGzel(a)209.216.1 77.7..CSeq: 101 INVITE..User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY..Contact: <sip:54 14362504(a)xxx.xxx.xxx.80>..Content-Type: application/sdp..Content-Length: 237....v=0..o=root 29275 29275 IN IP4 209.216.160. 80..s=session..c=IN IP4 xxx.xxx.xxx.80..t=0 0..m=audio 11184 RTP/AVP 0 3 8 97..a=rtpmap:0 PCMU/8000..a=rtpmap:3 GSM/8000..a =rtpmap:8 PCMA/8000..a=rtpmap:97 iLBC/8000..a=silenceSupp:off - - - -.. ** At this point the call is in fast-busy and Asterisk shows a failure. **** SER CFG # ser.cfg ############################################################ debug=3 fork=yes log_stderror=yes listen=xxx.xxx.xxx.36 port=5060 children=4 alias=whatever.net alias=whatever.whatever.net dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:serpass@localhost/ser" ### # Modules ### loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/acc.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/exec.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/permissions.so" ### # Module Paramaters ### modparam("auth_db|uri_db|usrloc", "db_url", "mysql://ser:serpass@localhost/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") # VM Timeout modparam("tm", "fr_inv_timer", 20) modparam("usrloc", "db_mode", 2) modparam("permissions", "db_mode", 0) modparam("permissions", "trusted_table", "trusted") modparam("registrar", "nat_flag", 6) modparam("rr", "enable_full_lr", 1) modparam("exec", "setvars", 1) modparam("acc", "log_level", 1) modparam("acc", "log_flag", 1) modparam("acc", "db_url", "mysql://ser:serpass@localhost/ser") modparam("acc", "db_flag", 1) ### # Routing Section ### route { # Basic Checks if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); break; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; }; if (method!="REGISTER") { record_route(); }; if (method=="BYE" || method=="CANCEL") { unforce_rtp_proxy(); } # Loose Routing if (loose_route()) { route(1); break; }; # Call Processing if (uri!=myself) { route(1); break; }; if (uri==myself) { if (method=="CANCEL") { route(3); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; }; route(1); } route[1] { ### # Default Call Handling ### # NAT Fix if (isflagset(6) && !isflagset(9)) { fix_nated_sdp("1"); force_rport(); fix_nated_contact(); force_rtp_proxy(); setflag(9); }; # On call timeout to voicemail t_on_failure("1"); t_on_reply("1"); # Log setflag(1); if (!t_relay()) { if (method=="INVITE" && isflagset(6)) { unforce_rtp_proxy(); }; sl_reply_error(); }; } route[2] { ### # REGISTER method ### if (!search("^Contact:\ +\*") && nat_uac_test("19")) { setflag(6); fix_nated_register(); force_rport(); }; sl_send_reply("100", "Trying"); if (!www_authorize("whatever.net","subscriber")) { www_challenge("whatever.net","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { ### # INVITE & CANCEL # URI matching is here. ### t_on_failure("1"); if (nat_uac_test("19")) { log(1," INVITE: NAT Detected. Setting flag 6.\n"); setflag(6); } if (method=="INVITE" && src_ip==xxx.xxx.xxx.7) { !allow_trusted()) { if (!proxy_authorize("whatever.net","subscriber")) { proxy_challenge("whatever.net","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); }; lookup("aliases"); # if (uri!=myself) { # force_rtp_proxy(); # route(1); # break; # }; if (!lookup("location")) { if (uri=~"^sip:911@") { # Pass 911 calls first route(5); break; }; if (uri=~"^sip:[0-9]{7}@") { # EXPAND 7-DIGIT CALL exec_dset("/root/7digit.pl"); }; # Try location lookup again if (!lookup("location")) { if (uri=~"^sip:[0-9]{10}@") { # DB Lookup of NPANXX route exec_dset("/root/npa_nxx.pl"); route(1); break; }; if (uri=~"^sip:1[0-9]{10}@") { # DB Lookup of NPANXX route exec_dset("/root/npa_nxx.pl"); route(1); break; }; # Call error message route(7); break; }; }; # if (isflagset(6)) { # force_rport(); # fix_nated_contact(); # force_rtp_proxy(); # }; t_on_reply("1"); if (!t_relay()) { if(isflagset(6)) { unforce_rtp_proxy(); } sl_reply_error(); }; } route[4] { ### # Primary LD Gateway ### rewritehost("xxxxxxxxxxxxxxxx"); route(1); } route[5] { ### # Local PSTN Gateway ### rewritehost("xxx.xxx.xxx.7"); route(1); } route[6] { ### # Calls to VM ### rewritehost("xxx.xxx.xxx.80"); route(1); } route[7] { ### # Call Failure ### rewriteuri("sip:7110000001@xxx.xxx.xxx.80"); route(1); } failure_route[1] { ### # Voicemail ### log(1,"Failure Route Hit\n"); rewritehost("xxx.xxx.xxx.80"); append_branch(); route(1); } onreply_route[1] { if (isflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") { if (!search("^Content-Length:\ +0")) { force_rtp_proxy(); }; }; if (nat_uac_test("1") && !isflagset(9)) { fix_nated_contact(); }; } -- Corey S. McFadden (c(a)csma.biz) McFadden Associates - Technology Consultants phone 215-825-2121 x510 - web.csma.biz ********************************************* This message has been scanned for viruses and dangerous content, and is believed to be clean.

18 years, 8 months

Re: [Serusers] xmlrpc calls

by zkeatts

Jan Janak wrote: > On 18-08-2005 19:01, Jan Janak wrote: > > >>> +-----------+--------+-------------------------+----------+---------------------+-------+-------------------------------------+------+---------------------+-----------+-------+-------+---------------------------------------+ >>> >>> | username | domain | contact | received | >>> expires | q | callid | cseq >>> | last_modified | replicate | state | flags | >>> user_agent | >>> +-----------+--------+-------------------------+----------+---------------------+-------+-------------------------------------+------+---------------------+-----------+-------+-------+---------------------------------------+ >>> >>> | 271nv1001 | | sip:271nv1001@localhost | NULL | >>> 2020-05-28 21:32:15 | -1.00 | 3653736-e3498618-a7aa0347@localhost >>> | 42 | 2005-08-15 16:07:31 | 0 | 0 | 0 | >>> PolycomSoundPointIP-SPIP_500-UA/1.4.1 | >>> +-----------+--------+-------------------------+----------+---------------------+-------+-------------------------------------+------+---------------------+-----------+-------+-------+---------------------------------------+ >>> >>> >>> I still get an AOR error when I try the following AOR's in my XML >>> query. >>> >>> <value><string>271nv1001</string></value> >>> <value><string>271nv1001@localhost</string></value> >>> <value><string>sip:271nv1001@localhost</string></value> >>> <value><string>3653736-e3498618-a7aa0347@localhost</string> >>> >>> Those seem to be the only logical choices for the AOR. Why would it >>> still be returning an AOR not found? >> >> I will get back to this later, I have to check the sources, it is >> possible there is a bug. >> > > > I have been trying to reproduce this but without success (it works for > me). Could you send me the output of usrloc.dump function and also > full traces of xmlrpc communication -- showing both requests and > replies ? > > thanks, Jan. > > Jan, Here is the information you requested. 1) Add User - request and response POST /RPC2 HTTP/1.0 User-Agent: Radio UserLand/7.1b7 (WinNT) Host: localhost:5060 Content-Type: text/xml Content-length: 131 <?xml version=\"1.0\"?> <methodCall> <methodName>usrloc.add_contact</methodName> <params> <param> <value><string>location</string></value> </param> <param> <value><string>testUser</string></value> </param> <param> <value><string>sip:testUser@localhost</string></value> </param> <param> <value><i4>30</i4></value> </param> <param> <value><double>1.0</double></value> </param> <param> <value><i4>0</i4></value> </param> </params> </methodCall> HTTP/1.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:32829 Server: Sip EXpress router (0.10.99-janakj_experimental (i386/linux)) Content-Length: 150 Warning: 392 127.0.0.1:5060 "Noisy feedback tells: pid=6165 req_src_ip=127.0.0.1 req_src_port=32829 in_uri=/RPC2 out_uri=/RPC2 via_cnt==1" <?xml version="1.0" encoding="UTF-8"?> <methodResponse> <params> <param><value><boolean>1</boolean></value></param> </params> </methodResponse> 2) Show Contacts request and response POST /RPC2 HTTP/1.0 User-Agent: Radio UserLand/7.1b7 (WinNT) Host: localhost:5060 Content-Type: text/xml Content-length: 131 <?xml version=\"1.0\"?> <methodCall> <methodName>usrloc.show_contacts</methodName> <params> <param> <value><string>location</string></value> </param> <param> <value><string>testUser</string></value> </param> </params> </methodCall> HTTP/1.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:32829 Server: Sip EXpress router (0.10.99-janakj_experimental (i386/linux)) Content-Length: 323 Warning: 392 127.0.0.1:5060 "Noisy feedback tells: pid=6163 req_src_ip=127.0.0.1 req_src_port=32829 in_uri=/RPC2 out_uri=/RPC2 via_cnt==1" <?xml version="1.0" encoding="UTF-8"?> <methodResponse> <fault> <value><struct> <member><name>faultCode</name> <value><i4>-500</i4></value></member> <member><name>faultString</name> <value><string>Can't allocate 3048949949-byte memory block</string></value></member> </struct></value> </fault> </methodResponse> 3) usrloc.dump request and response POST /RPC2 HTTP/1.0 User-Agent: Radio UserLand/7.1b7 (WinNT) Host: localhost:5060 Content-Type: text/xml Content-length: 131 <?xml version=\"1.0\"?> <methodCall> <methodName>usrloc.dump</methodName> <params> </params> </methodCall> HTTP/1.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:32829 Server: Sip EXpress router (0.10.99-janakj_experimental (i386/linux)) Content-Length: 298 Warning: 392 127.0.0.1:5060 "Noisy feedback tells: pid=6165 req_src_ip=127.0.0.1 req_src_port=32829 in_uri=/RPC2 out_uri=/RPC2 via_cnt==1" <?xml version="1.0" encoding="UTF-8"?> <methodResponse> <fault> <value><struct> <member><name>faultCode</name> <value><i4>500</i4></value></member> <member><name>faultString</name> <value><string>Not Yet Implemented</string></value></member> </struct></value> </fault> </methodResponse> Thanks for taking a look, Zach Keatts Software Engineer Nuvio.com

18 years, 8 months

[Serusers] Cacheless usrloc module

by Andreas Granig

Hi, I'd like to announce an experimental module for SER-0.9.x which provides a cacheless user location implementation. It's fully compatible with the original usrloc module (well, in fact you have to apply a small patch to assure full compatibility, but this patch doesn't hurt if you switch back to the original usrloc module). It currently only supports MySQL as backend because some MySQL specific queries are used. Also the watcher/callback API isn't tested, so you currently better not use it in combination with the pa module. You can find the module and a short howto here: http://linguin.org/ser.php#usrloc_db Please note that this module is very experimental. Any feedback is highly appreciated. Andy

18 years, 8 months

[Serusers] new pa entry in serctl usage menu

by Aisling

Hello, I recently upgraded to ser 0.9.3 and I noticed that there's a new entry in the serctl usages document: pa pres <p_uri> <pstate> .......set pstate for a presentity pa loc <p_uri><loc>.........set a location for a presentity Am I correct in assuming that the first command allows a a subscribers "availability" or presence state to be set? i.e. "busy", "unavailable" etc or is this limited to "online", "offline"? Also with the second command, does this represent something like a presentities current location i.e. office etc? If not, what does it represent or how should it be used syntactically? Based on my assumptions above, I tried "serctl pa loc aisling(a)x.x.x.x office" and the following errors occurred (SER was also killed): submit_query(): Unknown column 'pdomain' in 'where clause'. submit_query(): Error while submitting query pdomain_load_presentities: Error while querying presentity submit_query(): Unknown column 'pdomain' in 'where clause'. submit_query(): Error while submitting query new_presentity: Error while querying presentity child process 10644 exited by signal 11 core was generated INFO: signal 15 received INFO: signal 15 received INFO: signal 15 received INFO: signal 15 received Etc etc BUG: shutdown timeout triggered, dying.. Regards, Aisling. -------------------Legal Disclaimer--------------------------------------- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt.

18 years, 8 months

[Serusers] Connecting to Mysql

by Adam King

Hi I am have a problem connecting to the MySql database. I am running Fedora Core 3, ser 0.8.12 (i386/linux), mysql Ver 12.22 Distrib 4.0.24, for pc-linux-gnu (i686). In ser.cfg I have chosen to use the MySql database, and attempt to authenticate using either of (I have tried both at seperate times): modparam("auth_db", "db_url", "sql://root:XXXX@localhost/ser") modparam("auth_db", "db_url", "sql://ser:heslo@localhost/ser") both of which result in: 0(4971) connect_db(): Access denied for user: 'ser@localhost' (Using password: YES) 0(4971) db_init(): Error while trying to connect database 0(4971) mod_init(): Error while connecting database 0(4971) init_mod(): Error while initializing module usrloc However: # mysql -user -p Enter password:heslo AND # mysql -user -p Enter password:XXXX both work. Even if I attempt to connect using sql://root:XXXX@localhost/ser, do I get access denied to user ser? I have not changed the default password 'heslo' for the ser mysql user yet. I have also not changed the DEFAULT_DB_URL in config.h. I presume this is not a bug in SER or anything like that however, and teh credentials are correct on the MySql side, so I am stuck. Does anyone have a suggestion? Thanks in advance Adam

18 years, 8 months

[Serusers] Hi,problem about user managerment.

by viking

Hi,everyone! I wanna to use ser to manage the uers in our group, the process is : First,I update the Makefile of SER-0.9.3 to make mysql module,like follows: modules:=$(addprefix modules/,mysql) \ $(modules) then, I compile it,it prompts me that "can't find the file <mysql/mysql.h>" Secondly, i download mysql-standard-4.1.12-pc-linux-gnu-i686.tar.gz from mysql.com,and mkdir /ser-0.9.3/modules/mysql/mysql copy /usr/local/mysql/include/*.h /ser-0.9.3/modules/mysql/mysql Then, i compile ser again, and i get the targets like mysql.so,ser, serctl and so on. Next, i install the mysql from mysql-standard-4.1.12-pc-linux-gnu-i686.tar.gz according to the INSTALL-BINARY document. then i start the ser , cd /usr/local/sbin ser_mysql.sh create ser the problem is while i use "serctl add tieban 123456 tieban(a)sina.com",it prompts me that: $ serctl add viking 123456 tieban(a)sina.com MySql password: error: 400; check if you use aliases in SER How can i to make it, any suggestions will be appreciated. thx. 　　　　　　　　viking 　　　　　　　　tieban(a)sina.com 　　　　　　　　　　2005-09-02

18 years, 8 months

[Serusers] Adding extra attributes in radius requests

by Ashutosh Kumar

Hi, Similar to retrieving custom attribute pair values from a radius response, isn't there any module/technique to add more attribute value pairs to the radius request. OR if one modifies the ser source, will there by any change required in corresponding radiusclient dictionary at the ser's side. Regards, Ashutosh Kumar Chetu, Inc. Ph : 1(305) 402 6724 - Witin US Ph : 91 120 5323340 - Outside US Fax:1 (305) 832 5987 For more information, please visit http://www.chetu.com

18 years, 8 months

[Serusers] NAT Router problem

by Rick Thompson

Has anyone come across this problem before?? I have the same configuration for each of the following except the first session is going thru a Linksys router and the second one goes thru a Dlink router. The first session works fine, the second session has one way audio. The only difference in the traces is in the following lines. Session one Via: SIP/2.0/UDP 192.168.1.108:5060;rport=5060;received=199.227.155.139;branch=z9hG4bK-389cba 09 Session two Via: SIP/2.0/UDP 192.168.1.40:5060;rport=61034;received=199.227.155.135;branch=z9hG4bK-349caf af I have the following in ser.cfg to correct for a NATed UAC if(nat_uac_test("3")) { xlog("L_ALERT", "%Tf %mf ****** NATed UAC detected.\n"); fix_nated_contact(); force_rport(); if(method == "INVITE") { fix_nated_sdp("3"); }; }; And onreply_route[1] { xlog("L_ALERT", "%Tf %mf ****** On Reply From: <%si> <%tu> <%fu> <%rm> <%rs> <%rr>\n"); if(status=~"(183)|(2[0-9][0-9])" && nat_uac_test("3")) { xlog("L_ALERT", "%Tf %mf ****** Fixing contact on reply\n"); fix_nated_contact(); force_rport(); fix_nated_sdp("3"); }; Can anyone help Thanks for your reply Rick Thompson, CCNA VP of Technology Virtual Communications, LLC O 954-334-4000 ext 420 M 954-217-7322 F 954-252-4580

18 years, 8 months

[Serusers] BYE's not accounted

by Ashutosh Kumar

HI, My BYE's are not getting accounted in the mysql DB, though all other messages are getting reflected. I have put setflag for all messages, at the beginning of cfg. My relevant ser.cfg code is following. ..... ..... if (loose_route()) { if (has_totag() && (method=="INVITE" || method=="ACK")) { if (client_nat_test("3") || search("^Route:.*;nat=yes")) { setflag(6); use_media_proxy(); }; }; route(1); break; }; setflag(1); ..... ..... Please guide. Regards, Ashutosh Kumar

18 years, 8 months

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sr-users September 2005