- sr-users - kamailio.org

[Serusers] SER+ASTERISK voicemail
by harry gaillac 02 Sep '05

02 Sep '05

Hello, I set SER as sip proxy and ASTERISK as voicemail server (ARA) and serweb as TUI (telephone user interface) . Serweb | Ua-------ser-------asterisk voicemail | | Mysql DB I add user agents with address sip:name@domain + aliases sip:123@domain where 123 is mailbox I can forward voice messages to Asterisk with "failure route" for status 408 or 486. However I can't do it for offline users because of SER look for addresses like sip:user@domain not sip:123@domain where 123 is mailbox How could I solve this problem if possible ? Regards Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

1 0

[Users] BYE before 200 OK?
by Michael Ulitskiy 02 Sep '05

02 Sep '05

Hello, Can someone please enlighten me on if this is allowed to send BYE before 200 OK? I thought that BYE should be used only after dialog established, but now I have a SIP PSTN termination provider and they appear to be sending BYE with route headers right after 183 Session Progress in some cases. So the session looks like the following: UAC OpenSER PSTN Provider UAS INVITE to openser 180 Trying back to UAC INVITE to UAS 180 Trying back to openser 183 Session progress back to openser 183 Session Progress back to UAC BYE to openser with Route headers BYE to UAC (loose_routing) 200 OK to openser 200 OK to UAS Is it ok? And there's even more to it. I don't know if above is OK or not, but I would thought that now the session is ended. Nonetheless in exactly 50 seconds I'm getting 183 Session progress back to openser 183 Session progress back to UAC -- in another 60 seconds 183 Session progress back to openser 183 Session progress back to UAC -- this happens 6 times every 60 seconds and then 408 Request Timeout back to openser ACK to UAS 408 Request Timeout to UAC ACK to openser Do you think it's a provider bug? Another question is how openser is routing replies that cannot be matched to transaction as I think transaction has already been deleted after BYE and latest 183 replies were routed by some other principle. Could please someone comment on this? Thanks, See you later, Michael

3 8

[Users] Per-branch flags
by Bogdan-Andrei Iancu 02 Sep '05

02 Sep '05

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

2 3

[Serusers] NAT Related Voicemail Timeout Problem (HELP!)
by Corey S. McFadden 02 Sep '05

02 Sep '05

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.

2 2

Re: [Serusers] xmlrpc calls
by zkeatts 02 Sep '05

02 Sep '05

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

1 0

[Serusers] Cacheless usrloc module
by Andreas Granig 02 Sep '05

02 Sep '05

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

2 1

[Serusers] new pa entry in serctl usage menu
by Aisling 02 Sep '05

02 Sep '05

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.

1 0

[Serusers] Connecting to Mysql
by Adam King 02 Sep '05

02 Sep '05

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

2 1

[Serusers] Hi,problem about user managerment.
by viking 02 Sep '05

02 Sep '05

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

2 1

[Serusers] Adding extra attributes in radius requests
by Ashutosh Kumar 02 Sep '05

02 Sep '05

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

2 1

[Serusers] NAT Router problem
by Rick Thompson 02 Sep '05

02 Sep '05

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

2 1

[Serusers] BYE's not accounted
by Ashutosh Kumar 02 Sep '05

02 Sep '05

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

3 2

[Serusers] SEMS + SER howto
by Ryan Pagquil 01 Sep '05

01 Sep '05

Guys, Anybody has a good and detailed SEMS+SER howto, could please share it with me. I'm making SEMS as a voicemail for SER. I'm having a hard time configuring it. Or if not SEMS + SER maybe Asterisk + SER. Thanks, -- Ryan Pagquil Infodyne Inc. - PhilOnline.com 3603 Antel Global Corporate Center Doña Julia Vargas Ave. Ortigas Center Pasig City Tel: 687-0715 Web: www.philonline.com

5 9

[Serusers] Ser 0.9.3 and mysql module
by Daniel Corbe 01 Sep '05

01 Sep '05

Trying to load mysql.so with ser0.9.3 and I'm getting the following error: jomama:/usr/local/etc root# /usr/local/sbin/ser 0(6444) ERROR: load_module: could not open module </usr/local/lib/ser/modules/mysql.so>: dlopen(/usr/local/lib/ser/modules/mysql.so, 2): Symbol not found: _fm_malloc Referenced from: /usr/local/lib/ser/modules/mysql.so Expected in: /usr/local/sbin/ser 0(6444) parse error (26,13-14): failed to load module ERROR: bad config file (1 errors) I'm running Mac OS X Any help is appriciated. -Daniel

2 2

[Serusers] SEMS on Solaris10
by Jack Damn 01 Sep '05

01 Sep '05

Anyone got SEMS running on a Solaris 10 SPARC box ? Here it was compiled with gcc 3.3.2. When communicating with SER with FIFO, it dies with "(4576) ERROR: run (AmServer.cpp:164): odd trailer" when trying to read a line from the fifo. Any help appreciated. JJ.

1 0

[Users] question about LCR module
by jayesh nambiar 01 Sep '05

01 Sep '05

hi juha, I went through the readme, but i still did'nt understand one thing: In which order are the gateways in same group selected. suppose there are 2 gws in group_id 1, and if that group is given he highest priority, which gw will be selected. pls help me understand this... jayesh --------------------------------- Do you Yahoo!? Yahoo! Mail - You care about security. So do we.

2 1

[Users] Admin Guide, where?
by gmart-openser＠coinet.com 01 Sep '05

01 Sep '05

The INSTALL document refers to "admin's guide available from OpenSer website". I have not been able to find it. Is it currently one and the same as the Admin Guide from SER (iptel.org)? Gary

2 1

[Serusers] Question about auth_radius module
by Ricardo Martinez 01 Sep '05

01 Sep '05

Hello list. I was wondering if the module auth_radius can receive extra parameters from the Radius Server and proccess them as AVP values (something like the avp_radius can do) . I'm using : version: ser 0.9.4-rc1 (i386/linux) flags: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 @(#) $Id: main.c,v 1.197 2004/12/03 19:09:31 andrei Exp $ main.c compiled on 04:26:34 Jul 21 2005 with gcc 3.2 I was browsing the CVS directory and i noticed that this revision : Revision 1.18 / (view) - annotate - [select for diffs] , Thu Jun 30 20:29:31 2005 UTC (2 months ago) by andrei Branch: MAIN CVS Tags: last_merge_to_janakj, HEAD Changes since 1.17: +6 -2 lines Diff to previous 1.17 seems to do what i want. Am i right? Can the version that i'm actually using do the same? I Hope that somone can help me. Thanks! Regards, Ricardo Martinez.-

2 1

[Serusers] Re: [Serdev] PA query
by Mark Aiken 01 Sep '05

01 Sep '05

I've tried without success to get the PA module working properly. I did ask questions on the serusers list bit no one ever responded. I had a bit more luck with the internet2 PA code. I get the feeling they are the only folks working on the PA module. Even that code does not work as one would expect - NOTIFYs are not sent out to reflect changes in the location database. It would appear much work needs doing in this area. Mark On 7/25/05, Chiradeep Chhaya <cbchhaya(a)gmail.com> wrote: > > Hi, > > Is anyone aware of a a documentation (apart from the guide available > online or the README) for the Presence Agent module? I am having some > difficulties figuring out how the code fits together and it's interface > with the router/registrar. > > Any help will be much appreciated. > > Regards > > -Chiradeep Chhaya > (MS in Information Security > Georgia Institute of Technology) > http://chiradeepchhaya.blogspot.com > > _______________________________________________ > Serdev mailing list > serdev(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serdev >

2 2

Re: [Serusers] problem with LCR and CANCEL
by Ben Smithurst 01 Sep '05

01 Sep '05

Miguel Angel Villar wrote: > Hi Ben. > You shouldn't use next_gw() in failure route if you receive Busy or > Cancel from user: > > failure_route[1] { ... thanks for the quick reply! I was sure I'd tried that already... :-( Your example works fine though - I think the key thing I had wrong before was having a t_on_failure call inside the failure route - I'm sure I saw this in an example config somewhere, and while I thought it seemed odd, assumed there was a good reason for it. Evidently not. :-) Using your failure block has made it all seem to work fine. thanks again -ben -- Ben Smithurst ben(a)gradwell.net gradwell dot com Ltd Systems Developer http://bensmithurst.com/ http://www.gradwell.com/

2 1

RE: [Serusers] problem with LCR and CANCEL
by Miguel Angel Villar 01 Sep '05

01 Sep '05

Hi Ben. You shouldn't use next_gw() in failure route if you receive Busy or Cancel from user: failure_route[1] { if (t_check_status("486")) { # Busy Here xlog("L_INFO", "time [%Tf] FAILURE ROUTE - 486 BUSY HERE\n"); t_relay(); break; }; if (t_check_status("487")) { # Request Terminated by CANCEL from Calling Party xlog("L_INFO", "time [%Tf] FAILURE ROUTE - 487 CANCEL BY CALLING PARTY\n"); break; }; if (!next_gw()) { t_reply("503", "Service not available - No more Gateways"); break; } else { t_relay(); }; } Regards. MaV.- > -----Mensaje original----- > De: Ben Smithurst [SMTP:ben@gradwell.net] > Enviado el: Jueves, 01 de Septiembre de 2005 11:20 a.m. > Para: serusers(a)lists.iptel.org > Asunto: [Serusers] problem with LCR and CANCEL > > hello, > > apologies if this is a silly question... > > Basically when using the LCR module, if I CANCEL the call before the > remote party picks up, the LCR system will place the call again through > the next gateway. I have found quite a few references to this in this > list archives etc, so it is probably something fairly simple I have > setup wrong, but none of the messages I found seemed to solve the > problem for me. :-( > > After some googling I have read that this is because the first gateway > is not responding with a 487 (which it is in my case). My config has, > among other bits, > > route { > > ...... > > t_on_failure("1"); > if (!next_gw()) { > sl_send_reply("500", "cannot use next gateway"); > break; > }; > > ..... > } > > failure_route[1] { > > t_on_failure("1"); > if (!next_gw()) { > t_reply("503", "No more gateways"); > break; > }; > > t_relay(); > } > > If required I can provide more details but is there anything obviously > missing from the config above which I should be using? > > thanks > -ben > > -- > > Ben Smithurst ben(a)gradwell.net gradwell dot com Ltd > Systems Developer http://bensmithurst.com/ http://www.gradwell.com/ > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers

1 0

[Serusers] problem with LCR and CANCEL
by Ben Smithurst 01 Sep '05

01 Sep '05

hello, apologies if this is a silly question... Basically when using the LCR module, if I CANCEL the call before the remote party picks up, the LCR system will place the call again through the next gateway. I have found quite a few references to this in this list archives etc, so it is probably something fairly simple I have setup wrong, but none of the messages I found seemed to solve the problem for me. :-( After some googling I have read that this is because the first gateway is not responding with a 487 (which it is in my case). My config has, among other bits, route { ...... t_on_failure("1"); if (!next_gw()) { sl_send_reply("500", "cannot use next gateway"); break; }; ..... } failure_route[1] { t_on_failure("1"); if (!next_gw()) { t_reply("503", "No more gateways"); break; }; t_relay(); } If required I can provide more details but is there anything obviously missing from the config above which I should be using? thanks -ben -- Ben Smithurst ben(a)gradwell.net gradwell dot com Ltd Systems Developer http://bensmithurst.com/ http://www.gradwell.com/

1 0

[Serusers] Skype-Like program
by Nhadie 01 Sep '05

01 Sep '05

Hi, Any developer willing to help us develop a skype-like program. Willing to pay. E-Mail me privately. Thank you Regards, Nhadie

1 0

[Users] question about LCR module
by jayesh nambiar 01 Sep '05

01 Sep '05

hi all, Thanx for clearing my earlier doubts. Now i got a problem in lcr module. I have a doubt regarding the gw table abd lcr table. I'll just explain a scenario: I have 3 gateways in grp_id '1'. and i have 1 gateway in grp_id '2'. Now for example a prefix '11' arrives, i give grp_id as '1' and priority as '1'. Now among the three gateways that i have, which one will it choose to send the call as all the three will have the same priority. What is the logic used to resolve this situation. Do i have to put multiple entries of same prefixes in the lcr table to have different priorities and different groups. Does lcr module load the contents in memory or it queries the DB everytime. can someone please share the working example of script containing lcr properly. It would help me a lot. Thanx a lot.. Jayesh --------------------------------- Yahoo! Mail for Mobile Take Yahoo! Mail with you! Check email on your mobile phone.

2 1

[Serusers] Message - RTPProxy
by Anderson Alves de Albuquerque 01 Sep '05

01 Sep '05

I have problem with RTPProxy, does someone know what is the problema? I am using SER with RTPProxy with config in http://www.voip-info.org/wiki-SER+example+NAThelper. Sep 1 01:28:24 localhost /usr/local/sbin/ser[26003]: WARNING: rtpp_test: support for RTP proxyhas been disabled temporarily Sep 1 01:28:24 localhost /usr/local/sbin/ser[26001]: SER: open_uac_fifo: fifo server up at /tmp/ser_fifo... Sep 1 01:28:24 localhost /usr/local/sbin/ser[26004]: ERROR: send_rtpp_command: can't read reply from a RTP proxy Sep 1 01:28:24 localhost /usr/local/sbin/ser[26004]: WARNING: rtpp_test: can't get version of the RTP proxy Sep 1 01:28:24 localhost /usr/local/sbin/ser[26004]: WARNING: rtpp_test: support for RTP proxyhas been disabled temporarily Sep 1 01:28:24 localhost /usr/local/sbin/ser[26006]: ERROR: send_rtpp_command: can't read reply from a RTP proxy Sep 1 01:28:24 localhost /usr/local/sbin/ser[26006]: WARNING: rtpp_test: can't get version of the RTP proxy Sep 1 01:28:24 localhost /usr/local/sbin/ser[26006]: WARNING: rtpp_test: support for RTP proxyhas been disabled temporarily Sep 1 01:28:24 localhost /usr/local/sbin/ser[26007]: ERROR: send_rtpp_command: can't read reply from a RTP proxy Sep 1 01:28:24 localhost /usr/local/sbin/ser[26007]: WARNING: rtpp_test: can't get version of the RTP proxy Sep 1 01:28:24 localhost /usr/local/sbin/ser[26007]: WARNING: rtpp_test: support for RTP proxyhas been disabled temporarily Sep 1 01:28:24 localhost /usr/local/sbin/ser[25992]: ERROR: send_rtpp_command: can't read reply from a RTP proxy Sep 1 01:28:24 localhost /usr/local/sbin/ser[25992]: WARNING: rtpp_test: can't get version of the RTP proxy Sep 1 01:28:24 localhost /usr/local/sbin/ser[25992]: WARNING: rtpp_test: support for RTP proxyhas been disabled temporarily Sep 1 01:28:24 localhost /usr/local/sbin/ser[26005]: ERROR: send_rtpp_command: can't read reply from a RTP proxy Sep 1 01:28:24 localhost /usr/local/sbin/ser[26005]: WARNING: rtpp_test: can't get version of the RTP proxy Sep 1 01:28:24 localhost /usr/local/sbin/ser[26005]: WARNING: rtpp_test: support for RTP proxyhas been disabled temporarily

2 3

[Users] cvs freeze
by Daniel-Constantin Mierla 01 Sep '05

01 Sep '05

Hello, as scheduled, the next release has to be in September, by the end of the month. The next step is to freeze the CVS for a while -- this will happen in just a few days. No new features will added, only bugfixes must be committed. Afterwards, a branch will be created for release and new features can be included in CVS head So, until the cvs is frozen, it is the last chance to add something in the upcoming release. If you have something in mind, let us know. Testing the development version would help a lot in the next period! Points of high interest are TLS, branch_route, acc, avpops, textops, uac, uac_redirect ... as much as possible with pseudo-variables. Cheers, Daniel

1 0

[Serusers] SERWEB - FIFO not running or bad path to it
by Erdem HAKİ 01 Sep '05

01 Sep '05

Hello, I'm using ser-0.9.3 with serweb-0.9.3 that comes with it. I used "yum install .." command. Ser works good but i have a problem with serweb. When i logged in I'm getting an error on the first page as "FIFO not running or bad path to it", I searched in the list and find something but it's pending so i can't still resolve it. Any idea? Thanks Erdem HAKI

2 2

[Serusers] Mediaproxy proxy all calls.
by serlist＠khatuwala.com 01 Sep '05

01 Sep '05

Hi, I would like to know that after setting up mediaproxy with SER, will all calls' media stream go through the proxy? Will mediaproxy, proxy the call only when absolutely necessary or will it proxy in all cases? From what I understand about 80-90% of cases the rtp stream can get through NAT using STUN. I cannot find any information on this question on the internet. Thanks and regards, Vikash.

2 2

[Serusers] NAT traversal solution by reinvite?
by Federico Giannici 01 Sep '05

01 Sep '05

I'd like to ask to somebody with more knowledge of me if a possible solution to NAT traversal is really feasible. For various reasons, we DON'T want to use an RTP proxy. We'd like to avoid the use of STUN because: 1) creates hairpin problems; 2) many UAC have a bad STUN client code implementation; 3) it requires additional configuration by the final user. It seems to me that with the nathelper's message rewriting functions it is possible to solve every problem for the SIP protocol. Moreover, as we have the REAL IP of the UA (in the original SIP messages) we could also avoid haipin problems: it is sufficient to use the original IP/Port of the two UAs if both have the same natted IP (ie they are behind the same NAT). This doesn't work when the UAs are behind multiple NATs, but this is a relatively uncommon case. So, the unresolved problem is with the RTP data, because we don't know what will be the NATted port so we cannot correctly mangle the SDP data in the INVITE message. Am I correct up to this point? Now, I'm asking myself if it is feasible to use a "MINI RTP Proxy" that receives the initial INVITEs, discovering the NATted RTP ports, and then IMMEDIATLY RE-INVITE the two UAs to connect directly each other. So only the first RTP packet is actually proxed, all subsequent traffic will be directly between the two UAs. I think that something similar is done by Asterisk. Is this feasible? If it is, then we could have a good solution to NAT Traversal: 1) No Hairpin problems (for one NAT cases) 2) No problems of the normal RTP proxy (waste of bandwidth, longer delays, bad scalability). 3) Will work with all type of NATs except for symmetric ones (the same that work with STUN). 4) Simpler UAC configuraton: only username, password and sip server. Thanks. -- ___________________________________________________ __ |- giannici(a)neomedia.it |ederico Giannici http://www.neomedia.it ___________________________________________________

3 7

[Serusers] set timer for "Request Timeout"
by Jens Carl 01 Sep '05

01 Sep '05

Hello all, I have following records in the table 'usr_preferences': username attribute value 1000 fwdbusy sip:1001@example.de 1000 fwdnoanswer sip:1001@example.de I use these records to get the functionallity of "Call Forwarding - Busy" and "Call Forwarding - No Answer" into the OpenSer - server. I use a failure_route to act on the SIP - messages "Request Timeout (408)" and "Busy(486)" and forward the INVITE - message to the URI saved in the table 'usr_preferences'. See my snippet of the openser.cfg: ----snip --- failure_route[1] { if (isflagset(27) && t_check_error(408)) { if (avp_pushto("$ruri", "s:fwdnoanswer")) { avp_delete("s:fwdnoanswer"); resetflag(27); route(6); break; }; }; if (isflagset(26) && t_check_error(486)) { if (avp_pushto("$ruri", "s:fwdbusy")) { avp_delete("s:fwdbusy"); resetflag(26); route(6); break; }; }; }; ----snap --- These configuration works fine. Is there a possibillity to set a timer for the phone after the server receives the message "Request Timeout" from the phone? Or can I let the server ring for example five times and generate the message? Greetings Jens

1 0

[Serusers] ip based auth and billing
by Iqbal 01 Sep '05

01 Sep '05

Hi Has anyone attempted this, to see if it works, where the UA is using IP address only, Iqbal

2 2

[Serusers] forward breaks URI rewrite?
by Chris Roberts 01 Sep '05

01 Sep '05

Heya, This might be an obvious answer, but when I forward a packet to another box using the forward() method, this seems to break the URI? rewriteuri("sip:test@test.co.uk"); forward(x.x.x.x, 5060); Now on my asterisk box and looking at ngrep, the test(a)test.co.uk has disappeared, and asterisk takes the SIPDOMAIN as 'x.x.x.x', rather than 'test.com'. Is this the desired behaviour? Is there a way to do what I want to do? I need to be able to pick out the original SIPDOMAIN in asterisk even after the packet has been forwarded by SER. People seem to have talked about doing this, but I'm not sure how I can forward it without obliterating the URI. Cheers, Chris. -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.10.16/83 - Release Date: 26/08/2005

2 2

[Serusers] suppress CLI in SIP-SIP calls
by Gerd Feiner 01 Sep '05

01 Sep '05

Hi, how do I suppress the CLI (CLIR) in SIP-SIP calls? It seems to me, that SIP-clients of any kind send the CLI information (or originating user-ID) in any case. Can I somehow suppress CLIs in SER on a per- user basis? Brgds, Gerd

1 0

[Serusers] xmlrpc calls
by zkeatts 01 Sep '05

01 Sep '05

I am currently trying to do an xml-rpc call to usrloc.show_contacts. The xml packet I am sending is the following 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> </params> </methodCall> As a return packet I am getting HTTP/1.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:32769 Server: Sip EXpress router (0.10.99-janakj_experimental (i386/linux)) Content-Length: 303 Warning: 392 127.0.0.1:5060 "Noisy feedback tells: pid=5820 req_src_ip=127.0.0.1 req_src_port=32769 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>400</i4></value></member> <member><name>faultString</name> <value><string>More Parameters Expected</string></value></member> </struct></value> </fault> </methodResponse> This tells me that I am not entering the correct information, but I am at a loss to exactly what else I should be inserting. I did a search for rpc_show_contacts and found in ./sip_router/modules/usrloc/ul_rpc.c static void rpc_show_contacts(rpc_t* rpc) After searching some more I could not find the rpc_t structure. Does anyone know what values I should be passing via XML? Thanks, Zach Keatts Software Engineer Nuvio.com

2 6

[Serusers] URGENT:Authenitcation Mode
by Ashutosh Kumar 01 Sep '05

01 Sep '05

Hi, IF I set auth_db:calculate_ha1=no, will the radius authentication also be in non-digest (plaintext/basic) mode?? Reply please. Very urgent. Regards, Ashutosh Kumar

2 1

[Serusers] maxfwd module with Max-Forwards > 255 broken
by Hendrik Scholz 01 Sep '05

01 Sep '05

Hi! I checked how SER would behave with large Max-Forwards values. Paragraph 20.22 of RFC 3261 notes Max-Forwards to be anywhere between 0 and 255 but SER lacks a check for the upper boundry in the maxfwd module. I'm not sure how the correct way to remove the broken header field and introducing a correct one would be so I leave this for somebody who knows :) Hendrik -- freenet Cityline GmbH, Hamburger Chaussee 2-4, 24114 Kiel, Germany Phone: +49 (0)431 9020552, Fax: +49 (0)431 9020559 Internet: http://www.freenet.de, eMail: hendrik.scholz(a)freenet-ag.de

2 1

[Serusers] overflowing Expire values
by Hendrik Scholz 01 Sep '05

01 Sep '05

Hi! I gave 3.1.2.4 of draft-ietf-sipping-torture-tests-07 a try and sent - a REGISTER with Expires > 2^31 and lower than 2^32 (2147483748 in my case). - a REGISTER with Expires > 2^32 (4294967596 = 2^32 + 300) I used print_expires() to print the resulting values SER (0.9.3 + changes) parsed. It's a negative value in first case and 300 in the latter one. parser/parse_expires.h defines exp_body_t to have 'int val' which is not enough to store the parsed values. Changing it to 'unsigned int val' prevents the negative values from coming up in SER but does not solve the main problem. The draft stats that a default value of 3600 seconds should be assumed in case of errors, so an additional check to make sure _e->val will always be below 2^32 is necessary. I do have a small patch for our non-standard branch and could wrap one up for HEAD if needed. Hendrik -- freenet Cityline GmbH, Hamburger Chaussee 2-4, 24114 Kiel, Germany Phone: +49 (0)431 9020552, Fax: +49 (0)431 9020559 Internet: http://www.freenet.de, eMail: hendrik.scholz(a)freenet-ag.de

2 2

Fwd: Re: [Serusers] Blocking calls from unregistered users
by Pavol Segec 01 Sep '05

01 Sep '05

Hi, The code means, that if an user is establishing a call (UA generate INVITE message) and the user is not already registered on your.domain register server, the user will be challenged to authenticate itself before the call is progressed, and only authentication of users from your.domain will be authorized (accepted). Personally I think that good practise is to put some code into ser.cfg where REGISTER message is handled, which allow to register (and make a call) only users, which have account on your.domain register (orproxy with register features) server, for example: sl_send_reply("100","Trying"); if (!www_authorize("your.domain", "subscriber")) { www_challenge("your.domain", "0"); break; }; # only registered users are allowed # do not register someone else if(!check_to()){ log(1,"LOG: unregistered user registration attempt\n"); sl_send_reply("401","Unauthorized"); sl_send_reply("403","Only registered users are allowed"); break; }; # save the registration process results if (!save("location")){ sl_reply_error(); }; BR pavol Citát Ryan Pagquil <rpagquil(a)philonline.com>: > Hi, > I'm currently subscribed to iptel.org. Well iptel.org's proxy doesn't > allow unregistered users to call to anybody, I already tried it. Does > having my domain specified in the proxy_authorize section means that > only users from my domain will be asked to register first before they > can place a call? I think I already tried this.. but I'll still test. =) > > Thanks, > Ryan > > Pavol Segec wrote: > > >Hi, > > > >If I understand well, you just need to put authorization challenge into your > >ser.cfg file where INVITE messages are handled, as follows: > > > >if(!proxy_authorize("your.domain","subscriber")){ > > proxy_challenge("your.domain","0"); > > sl_send_reply("403","Forbidden"); > > break; > > }; > > > >pavol > > > >CitĂĄt Dave <ddx66(a)yahoo.com>: > > > > > > > >>You can't as far as I know. You must use a USer Agent > >>that does not allow a user to make a calls unless the > >>UA is registered. > >> > >>--- rpagquil(a)philonline.com wrote: > >> > >> > >>--------------------------------- > >>Hi, I'm setting up ser so that unregistered users > >>can't make any calls to anybody. I have configured to > >>allow all other domains to make a call to my local > >>users. But when my local user that is unregistered it > >>can still make calls to other local users. How would I > >>do to block him totally? here is my ser.cfg: debug=3 > >>fork=yes log_stderror=yes listen=202.84.24.107 > >>port=5060 children=4 dns=no rev_dns=no > >>fifo="/tmp/ser_fifo" > >>fifo_db_url="mysql://ser:heslo@localhost/ser" > >>alias=sip.philonline.com #load module part > >>loadmodule "/usr/local/lib/ser/modules/mysql.so" > >>loadmodule "/usr/local/lib/ser/modules/domain.so" > >>loadmodule "/usr/local/lib/ser/modules/sl.so" > >>loadmodule "/usr/local/lib/ser/modules/tm.so" > >>loadmodule "/usr/local/lib/ser/modules/rr.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/uri.so" > >>loadmodule "/usr/local/lib/ser/modules/uri_db.so" > >>loadmodule "/usr/local/lib/ser/modules/mediaproxy.so" > >>loadmodule "/usr/local/lib/ser/modules/nathelper.so" > >>loadmodule "/usr/local/lib/ser/modules/textops.so" > >>loadmodule "/usr/local/lib/ser/modules/acc.so" > >>loadmodule "/usr/local/lib/ser/modules/permissions.so" > >> #module parameter setup modparam("rr", > >>"enable_full_lr", 1) > >>modparam("auth_db|uri_db|usrloc|domain|permissions", > >>"db_url", "mysql://ser:heslo@localhost/ser") > >>modparam("auth_db", "calculate_ha1", 1) > >>modparam("auth_db", "password_column", "password") > >>modparam("usrloc", "db_mode", 2) > >>modparam("nathelper", "rtpproxy_disable", 1) > >>modparam("nathelper", "natping_interval", 0) > >>modparam("mediaproxy", "natping_interval", 30) > >>modparam("mediaproxy", "mediaproxy_socket", > >>"/var/run/mediaproxy.sock") modparam("mediaproxy", > >>"sip_asymmetrics", "/usr/local/etc/ser/sip-clients") > >>modparam("mediaproxy", "rtp_asymmetrics", > >>"/usr/local/etc/ser/rtp-clients") > >>modparam("registrar", "nat_flag", 6) modparam("acc", > >>"log_level", 2) modparam("acc", "log_fmt", > >>"cdfimorstup") modparam("acc", "report_ack", 1) > >>#modparam("acc", "failed_transactions", 1) > >>modparam("acc", "log_flag", 1) #modparam("acc", > >>"report_cancels", 1) modparam("acc", "db_flag", 1) > >>modparam("acc", "db_missed_flag", 3) modparam("acc", > >>"db_url", "mysql://ser:heslo@localhost/ser") > >>modparam("uri_db", "uri_table", "uri") > >>modparam("uri_db", "uri_user_column", "username") > >>modparam("uri_db", "uri_domain_column", "domain") > >>modparam("domain", "db_mode", 1) modparam("domain", > >>"domain_table", "domain") modparam("domain", > >>"domain_col", "domain") modparam("permissions", > >>"default_allow_file", > >>"/usr/local/etc/ser/allow.permissions") > >>modparam("permissions", "default_deny_file", > >>"/usr/local/etc/ser/deny.permissions") #our routing > >>logic route { 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; }; ###record > >>route#### if (method=="INVITE" && > >>client_nat_test("3")) { > >>record_route_preset("202.84.24.107:5060;nat=yes"); > >> } else if (method!="REGISTER") { > >>record_route(); }; ###call tear down > >>section### if (method=="BYE" || > >>method=="CANCEL") { > >>end_media_session(); }; ###accounting### > >> if ((!has_totag() && (method=="INVITE" || > >>method=="ACK")) || (method=="BYE")) { > >>setflag(1); }; ###loose route### 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; }; ###call type 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); } ##Default message handler## route[1] { > >> t_on_reply("1"); if (!t_relay()) { > >> if (method=="INVITE" || method=="ACK") { > >> end_media_session(); > >> }; sl_reply_error(); }; } > >> ##Register message handler## route[2] { > >>sl_send_reply("100", "Trying"); if > >>(!search("^Contact:\ +\*") && client_nat_test("7")) { > >> setflag(6); > >>fix_nated_register(); force_rport(); > >> }; if > >>(!www_authorize("sip.philonline.com","subscriber")) { > >> > >>www_challenge("sip.philonline.com","0"); > >> break; }; if (!check_to()) { > >> sl_send_reply("401", "You are Unauthorized"); > >> break; }; > >>consume_credentials(); if (!save("location")) > >>{ sl_reply_error(); }; } > >>##INVITE message Handler## route[3] { > >>if (client_nat_test("3")) { > >>setflag(7); force_rport(); > >> fix_nated_contact(); }; if > >>(!search("To: .*(a)sip.philonline.com")) { if > >>(!proxy_authorize("","subscriber")) { > >>proxy_challenge("", "0"); break; > >> }; if (!check_from() && method=="INVITE") > >>{ sl_send_reply("403", "User > >>From=ID"); break; }; > >>}; lookup("aliases"); if > >>(uri!=myself) { route(1); > >> break; }; if (!lookup("location")) > >>{ sl_send_reply("404", "User not > >>found"); break; }; if > >>(method=="CANCEL") { route(1); > >> break; }; > >>consume_credentials(); if (isflagset(6) || > >>isflagset(7)) { use_media_proxy(); > >> }; route(1); } onreply_route[1] { > >> if (isflagset(6) || isflagset(7) && > >>(status=~"(180)|(183)|2[0-9][0-9]")) { > >> if (!search("^Content-Length:\ +0")) { > >> use_media_proxy(); }; > >>}; if (client_nat_test("1")) { > >> fix_nated_contact(); }; } Thanks, > >>--ryanRyan PagquilInfodyne Inc. > >>(www.philonline.com)Tel. (632)-6870715> > >>_______________________________________________ > >> > >> > >>>Serusers mailing list > >>>serusers(a)lists.iptel.org > >>>http://lists.iptel.org/mailman/listinfo/serusers > >>> > >>> > >>> > >> > >> > >>____________________________________________________ > >>Start your day with Yahoo! - make it your home page > >>http://www.yahoo.com/r/hs > >> > >> > >>_______________________________________________ > >>Serusers mailing list > >>serusers(a)lists.iptel.org > >>http://lists.iptel.org/mailman/listinfo/serusers > >> > >> > >> > > > > > >_______________________________________________ > >Serusers mailing list > >serusers(a)lists.iptel.org > >http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > > > > > -- > Ryan Pagquil > Infodyne Inc. - PhilOnline.com > 3603 Antel Global Corporate Center > DoĂąa Julia Vargas Ave. > Ortigas Center Pasig City > Tel: 687-0715 > Web: www.philonline.com > > ----- Koniec preposlanej správy -----

1 0

[Serusers] Dchp or Static ip
by chinaj123 01 Sep '05

01 Sep '05

To All, Has anyone tried to setup SER using a DHCP assigned address behind a NAT? I just want to know what the pro's and the con's were. Thanks, Alfred ____________________________________________________ Start your day with Yahoo! - make it your home page http://www.yahoo.com/r/hs

1 0

[Serusers] Blocking calls from unregistered users
by rpagquil＠philonline.com 31 Aug '05

31 Aug '05

5 4

[Users] DROP statement
by Bogdan-Andrei Iancu 31 Aug '05

31 Aug '05

Hi there, The core "drop" statement has a new special meaning in some route types (by default it's equivalent to "exit"): - BRANCH ROUTE - the request branch is discarded (not fwd); - ONREPLY ROUTE - if provisional reply (<200), it will be discarded (not fwd, not stored, but only timers updated); Wiki will be updated soon with this text ;) regards, bogdan

1 0

[Serusers] RURI or URI
by Nando 31 Aug '05

31 Aug '05

hello, somebody knows the true difference between this: sip:1111@sip.domain.com y sip:1111 at sip.domain.com Because in logs (with ngrep) only I see the URI default????? (sip:1111@.....) Why i do not see the RURI??? Thanks

1 0

[Serusers] Support for multiple PSTN Gateways (hunting, crank back)
by Frank Fischer 31 Aug '05

31 Aug '05

Hi all i have multiple gateway devices to PSTN. All are equal in terms of what destinations may be reached behind it or the termination costs. I use multiple devices for two reasons: capacity and failover. So what i need is functionality to hunt lines or at least doing a crank back (or even better both). The best way to this (IMHO) would be to track the line usage on each gateway and check it before routing to a gateway to make sure, there is a line available to place the call to (this would be a kind of line hunting). But i could also "live" with a solution that works more like a failover mechanism. In that case i would route the call to the first gateway in my list and if it is not available or has no more free lines, i would need to detect this, take the call back and place it on another gateway. As far as i could find out, there is no hunting, failover or crank back support directly built into SER. Is it possible to implement such functionality using the "common" routing scripts in SER? Did anyone of you ever had similar requirements and did you find a way to fullfill it? I'm very thankfull for any idea, hint, snipplet, .... Thanks a lot for your help. Best Regards Frank

3 4

[Serusers] FWD: URGENT:Authenitcation Mode
by Ashutosh Kumar 31 Aug '05

31 Aug '05

Hi, Can I do basic mode authentication with radius. If yes, how? Regards, Ashutosh Kumar __ ______________________________________ From: Ashutosh Kumar [mailto:ashutoshk@chetu.com] Sent: Wednesday, August 31, 2005 9:51 PM To: 'serusers(a)lists.iptel.org' Subject: URGENT:Authenitcation Mode Hi, IF I set auth_db:calculate_ha1=no, will the radius authentication also be in non-digest (plaintext/basic) mode?? Reply please. Very urgent. Regards, Ashutosh Kumar

1 0

[Serusers] Adding +1 to From:/To: for E.164 compliance
by Ryan Tucker 31 Aug '05

31 Aug '05

Greetings... We are starting to work with a carrier which requires From: and To: to be in E.164 format (e.g. +17005551212). prefix() only seems to change the INVITE, which doesn't appear to cut the mustard with these guys. Is there any way to make this happen within SER? Thanks! -rt -- Ryan S. Tucker Network Engineer ISP Direct, Inc. 1159 Pittsford-Victor Road Building 5, Suite 140 Pittsford, New York 14534 +1 585.419.8200 http://www.isp-direct.com/ http://www.vocal-net.net/ The information contained in this message may be privileged, confidential, and protected from disclosure. If the reader of this message is not the intended recipient, or any employee or agent responsible for delivering this message to the intended recipient, you are hereby notified that any dissemination, distribution, or copying of this communication is strictly prohibited. If you have received this communication in error, please notify us immediately by replying to the message and deleting it from your computer. Thank you.

5 7

[Users] set timer for Request Timeout
by Jens Carl 31 Aug '05

31 Aug '05

Hello all, I have following records in the table 'usr_preferences': username attribute value 1000 fwdbusy sip:1001@example.de 1000 fwdnoanswer sip:1001@example.de I use these records to get the functionallity of "Call Forwarding - Busy" and "Call Forwarding - No Answer" into the OpenSer - server. I use a failure_route to act on the SIP - messages "Request Timeout (408)" and "Busy(486)" and forward the INVITE - message to the URI saved in the table 'usr_preferences'. See my snippet of the openser.cfg: ----snip --- failure_route[1] { if (isflagset(27) && t_check_error(408)) { if (avp_pushto("$ruri", "s:fwdnoanswer")) { avp_delete("s:fwdnoanswer"); resetflag(27); route(6); break; }; }; if (isflagset(26) && t_check_error(486)) { if (avp_pushto("$ruri", "s:fwdbusy")) { avp_delete("s:fwdbusy"); resetflag(26); route(6); break; }; }; }; ----snap --- These configuration works fine. Is there a possibillity to set a timer for the phone after the server receives the message "Request Timeout" from the phone? Or can I let the server ring for example five times and generate the message? Greetings Jens

2 1

[Serusers] Error using TCP
by Federico Giannici 31 Aug '05

31 Aug '05

I'm using SER 0.9.3 with OpenBSD 3.7 amd64. Everything works correctly if I use normal UDP connections. Now I have to use a TCP connection (UDP is blocked in that network). Using ngrep I see SER receives the requests ma no reply is sent. The following errors are logged: Aug 23 12:48:06 eowyn /usr/local/sbin/ser[16406]: ERROR: send_fd: sendmsg failed on 19: Invalid argument Aug 23 12:48:06 eowyn /usr/local/sbin/ser[16406]: ERROR: send2child: send_fd failed Aug 23 12:48:06 eowyn /usr/local/sbin/ser[16406]: ERROR: tcp_main_loop: no children available So it seems that no TCP childrens are available but I can see them using "serctl ps": 0 10963 attendant 1 24595 fifo server 2 19807 receiver child=0 sock= 195.120.250.50:5060 3 24795 receiver child=1 sock= 195.120.250.50:5060 4 6812 receiver child=2 sock= 195.120.250.50:5060 5 27930 receiver child=3 sock= 195.120.250.50:5060 6 22186 timer 7 1034 tcp receiver 8 3296 tcp receiver 9 605 tcp receiver 10 15085 tcp receiver 11 18905 tcp main process What could be the problem? Thanks. -- ___________________________________________________ __ |- giannici(a)neomedia.it |ederico Giannici http://www.neomedia.it ___________________________________________________

4 7

[Users] Combining AVPs
by Martin Kjeldsen 31 Aug '05

31 Aug '05

Hi Can anyone offer some helps on how to combine multiple AVPs. What I would like to do is to create an uri (eg. sip:bob@biloxi.com), where the "bob" part have been taken from e.g. the username column in the alias table and the rest either prefixed/appended from some static data or grabbed from another tuple! Can anyone offer some insight as to how this may best be accomplished. -- Med venlig hilsen / Kind Regards Martin Kjeldsen Cuatro-Group P/S (4G) Hindegade 3, DK-1303 København K Tel: (+45) 82 35 65 00 Fax: (+45) 82 35 65 65 Mob: (+45) 60 64 65 87 Url: http://www.cuatro.dk

2 3

[Serusers] NATHelper IPv4-IPv6
by Ahmed Kellal 31 Aug '05

31 Aug '05

Hello everyone, I am working on setting up a heterogeneous platform with both IPv4 and IPv6 UAs. Problem I am facing right now is that I need to add a little to the code, in order to: 1- Extract the hostname part of the Contact header field, when needed 2- Change the said hostname contact 3- Extract the IP (v4 or v6) address in the owner ("o=") field of the SDP body 4- Change it with the value the RTP Proxy assigns dynamically to represent an IPv4 UA in the IPv6 world and vice versa In order to do these things, I tried to understand as much as I can of the code of the NAtHelper module. I wrote four functions to do the above, but (naturally enough in my case :-)) they're not doing what I want them to. So if anyone could help me out here, I would sure be grateful, esp. that I have a serious deadline! Thanks Here are the sections I added (I drew inspiration from the functions already existing in nathelper.c) /************* Added Section 1: Define the newly inserted functions ****************/ static int extract_sip_contact_host(struct sip_msg *, char *); static int extract_sdp_ownerip(str *, str *, int *); static int alter_sdp_ownerip(struct sip_msg *, str *, str *, int, str *, int, int); static int modify_contact (struct sip_msg *, char *, char *, char *); /************* End of Added Section 1 ******************************/ /********************* Added Section 2: The modify_contact function ******************/ /********* Replaces ip:port pair in the Contact: field with the hostname desired **********/ static int modify_contact (struct sip_msg* msg, char* str1, char* str2, char* hostname) { int offset, len, len1; char *cp, *buf, temp[2]; contact_t* c; struct lump* anchor; struct sip_uri uri; if (get_contact_uri(msg, &uri, &c) == -1) return -1; if (uri.proto != PROTO_UDP && uri.proto != PROTO_NONE) return -1; if (uri.port.len == 0) uri.port.s = uri.host.s + uri.host.len; offset = c->uri.s - msg->buf; anchor = del_lump(msg, offset, c->uri.len, HDR_CONTACT); if (anchor == 0) return -1; cp = hostname; len = c->uri.len + strlen(cp) + 6 /* :port */ - (uri.port.s + uri.port.len - uri.host.s) + 1; buf = pkg_malloc(len); if (buf == NULL) { LOG(L_ERR, "ERROR: fix_contact: out of memory\n"); return -1; } temp[0] = uri.host.s[0]; temp[1] = c->uri.s[c->uri.len]; c->uri.s[c->uri.len] = uri.host.s[0] = '\0'; len1 = snprintf(buf, len, "%s%s:%d%s", c->uri.s, cp, msg->rcv.src_port, uri.port.s + uri.port.len); if (len1 < len) len = len1; uri.host.s[0] = temp[0]; c->uri.s[c->uri.len] = temp[1]; if (insert_new_lump_after(anchor, buf, len, HDR_CONTACT) == 0) { pkg_free(buf); return -1; } c->uri.s = buf; c->uri.len = len; return 1; } /****************************** End of Added Section 2 ************************/ /**** Added Section 3: Function to extract the host part only of the Contact SIP URI *******/ static int extract_sip_contact_host(struct sip_msg* msg, char* hostname) { struct sip_uri uri; contact_t* c; if (get_contact_uri(msg,&uri,&c)==-1) return -1; *hostname = uri.host.s; return 1; } /*********************** End of Added Section 3 ******************************/ /************ Added Section 4: Function to extract the IP @ in 'o' SDP parameter **********/ static int extract_sdp_ownerip(str *body, str *ownerip, int *pf) { char *cp, *cp1; int len, nextisip; cp1 = NULL; for (cp = body->s; (len = body->s + body->len - cp) > 0;) { cp1 = ser_memmem(cp, "o=", len, 2); if (cp1 == NULL || cp1[-1] == '\n' || cp1[-1] == '\r') break; cp = cp1 + 2; } if (cp1 == NULL) { LOG(L_DBG, "ERROR: extract_mediaownerip: no `o=' in SDP\n"); return -1; } ownerip->s = cp1 + 2; ownerip->len = eat_line(ownerip->s, body->s + body->len - ownerip->s) - ownerip->s; trim_len(ownerip->len, ownerip->s, *ownerip); nextisip = 0; for (cp = ownerip->s; cp < ownerip->s + ownerip->len;) { len = eat_token_end(cp, ownerip->s + ownerip->len) - cp; if (nextisip == 1) { ownerip->s = cp; ownerip->len = len; nextisip++; break; } if (len == 3 && memcmp(cp, "IP", 2) == 0) { switch (cp[2]) { case '4': nextisip = 1; *pf = AF_INET; break; case '6': nextisip = 1; *pf = AF_INET6; break; default: break; } } cp = eat_space_end(cp + len, ownerip->s + ownerip->len); } if (nextisip != 2 || ownerip->len == 0) { LOG(L_ERR, "ERROR: extract_mediaownerip: " "no `IP[4|6]' in `c=' field\n"); return -1; } return 1; } /************************** End of Added Section 4 ****************************/ /**** Added Section 5: Function to alter the IP information in the 'o' SDP parameter ********/ static int alter_sdp_ownerip(struct sip_msg *msg, str *body, str *oldip, int oldpf, str *newip, int newpf, int preserve) { char *buf; int offset; struct lump* anchor; str omip, nip, oip; /* check that updating mediaip is really necessary */ if (oldpf == newpf && isnulladdr(oldip, oldpf)) return 0; if (newip->len == oldip->len && memcmp(newip->s, oldip->s, newip->len) == 0) return 0; if (preserve != 0) { anchor = anchor_lump(msg, body->s + body->len - msg->buf, 0, 0); if (anchor == NULL) { LOG(L_ERR, "ERROR: alter_sdp_ownerip: anchor_lump failed\n"); return -1; } if (oldpf == AF_INET6) { omip.s = AOLDMEDIP6; omip.len = AOLDMEDIP6_LEN; } else { omip.s = AOLDMEDIP; omip.len = AOLDMEDIP_LEN; } buf = pkg_malloc(omip.len + oldip->len + CRLF_LEN); if (buf == NULL) { LOG(L_ERR, "ERROR: alter_sdp_ownerip: out of memory\n"); return -1; } memcpy(buf, omip.s, omip.len); memcpy(buf + omip.len, oldip->s, oldip->len); memcpy(buf + omip.len + oldip->len, CRLF, CRLF_LEN); if (insert_new_lump_after(anchor, buf, omip.len + oldip->len + CRLF_LEN, 0) == NULL) { LOG(L_ERR, "ERROR: alter_sdp_ownerip: insert_new_lump_after failed\n"); pkg_free(buf); return -1; } } if (oldpf == newpf) { nip.len = newip->len; nip.s = pkg_malloc(nip.len); if (nip.s == NULL) { LOG(L_ERR, "ERROR: alter_sdp_ownerip: out of memory\n"); return -1; } memcpy(nip.s, newip->s, newip->len); } else { nip.len = newip->len + 2; nip.s = pkg_malloc(nip.len); if (nip.s == NULL) { LOG(L_ERR, "ERROR: alter_sdp_ownerip: out of memory\n"); return -1; } memcpy(nip.s + 2, newip->s, newip->len); nip.s[0] = (newpf == AF_INET6) ? '6' : '4'; nip.s[1] = ' '; } oip = *oldip; if (oldpf != newpf) { do { oip.s--; oip.len++; } while (*oip.s != '6' && *oip.s != '4'); } offset = oip.s - msg->buf; anchor = del_lump(msg, offset, oip.len, 0); if (anchor == NULL) { LOG(L_ERR, "ERROR: alter_sdp_ownerip: del_lump failed\n"); pkg_free(nip.s); return -1; } if (insert_new_lump_after(anchor, nip.s, nip.len, 0) == 0) { LOG(L_ERR, "ERROR: alter_sdp_ownerip: insert_new_lump_after failed\n"); pkg_free(nip.s); return -1; } return 0; } /***************** End of Added Section 5 ******************************/ /***********Added Section 6: added section in force_rtpproxy_2****************/ if (extract_sdp_ownerip(&body, &oldownerip, &ownerpf) == -1) { LOG(L_ERR, "ERROR: force_rtp_proxy2_2: can't extract media owner IP " "from the message\n"); return -1; } if (asymmetric != 0 || real != 0) { newownerip = oldownerip; } else { newownerip.s = ip_addr2a(&msg->rcv.src_ip); newownerip.len = strlen(newownerip.s); } body2.s = oldownerip.s + oldownerip.len; body2.len = body.s + body.len - body2.s; if (extract_sdp_ownerip(&body2, &oldownerip1, &ownerpf1) == -1) { oldownerip1.len = 0; } if (oldownerip1.len > 0 && ownerpf != ownerpf1) { LOG(L_ERR, "ERROR: force_rtp_proxy2_2: mismatching address " "families in SDP\n"); return -1; } /********************* End of Added Section 6********************/ /*********** Added Section 7: added section in force_rtpproxy_2****************/ if (alter_sdp_ownerip(msg, &body, &oldownerip, ownerpf, &newip, pf1, 0) == -1) return -1; if (oldownerip1.len > 0 && alter_sdp_ownerip(msg, &body2, &oldownerip1, ownerpf, &newip, pf1, 0) == -1) return -1; /*********** End of Added Section 7************/ I apologize for this email's length, but I'm trying to be as detailed as I can, and I really do need the help :-) Thank you again Ahmed _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/

1 0

[Serusers] load testing mediaproxy for more than 50 calls
by Simone Cittadini 31 Aug '05

31 Aug '05

I need to test the new ser/radius/mediaproxy server we have bought. As a fast upper bound to the maximum number of concurrent calls I tought to use rtpgenerator, but it can't increase the number of simultaneous calls more than 50 ... Is there any way to increase that limit ? (why it is set anyway ?) Launching a second instance raises cpu usage from 0,5 to 98, so it doesn't seems a good solution ...

1 0

[Serusers] FIFO not running or bad path to it
by Erdem HAKİ 31 Aug '05

31 Aug '05

Hi, I have also this problem when I logged in, I'm getting an error on the first page as "FIFO not running or bad path to it" and can't still resolve. Any idea? Thanks Erdem HAKI ---------------------------------------------------------------- Read the INSTALL file!! Requirements: ------------- - PEAR package LOG (optional - not included in PHP realase) see http://pear.php.net/manual/en/installation.getting.php for details Karel Marc Khayat napsal(a): > Anybody got the Log.php file from serweb (ser-0.9.3), coz it seems like > it's missing from my package. > > Thx, > > Marc > > -----Original Message----- > From: serusers-bounces at iptel.org <http://lists.iptel.org/mailman/listinfo/serusers> [mailto:serusers-bounces at iptel.org <http://lists.iptel.org/mailman/listinfo/serusers> ] On > Behalf Of Marc Khayat > Sent: Friday, August 19, 2005 11:25 AM > To: Karel Kozlik > Cc: serusers at iptel.org <http://lists.iptel.org/mailman/listinfo/serusers> > Subject: RE: [Serusers] Serweb > > /tmp/ser_fifo/ is wrong since it'll be looking for a folder... > > Marc > > -----Original Message----- > From: serusers-bounces at iptel.org <http://lists.iptel.org/mailman/listinfo/serusers> [mailto:serusers-bounces at iptel.org <http://lists.iptel.org/mailman/listinfo/serusers> ] On > Behalf Of Marc Khayat > Sent: Friday, August 19, 2005 11:03 AM > To: Karel Kozlik > Cc: serusers at iptel.org <http://lists.iptel.org/mailman/listinfo/serusers> > Subject: RE: [Serusers] Serweb > > Hi Karel, > Thx for bearing with me :) > In config_paths.php, I changed it from "/tmp/ser_fifo" to > "tmp/ser_fifo/" > The errors at the top disappeared but I'm still getting the "FIFO not > running or bad path to it" message. > > Is it wrong to put "tmp/ser_fifo/"? knowing that in ser.cfg, it's > /tmp/ser_fifo > > Thanks again, > > Marc > > -----Original Message----- > From: Karel Kozlik [mailto:karel at iptel.org <http://lists.iptel.org/mailman/listinfo/serusers> ] > Sent: Friday, August 19, 2005 9:28 AM > To: Marc Khayat > Cc: serusers at iptel.org <http://lists.iptel.org/mailman/listinfo/serusers> > Subject: Re: [Serusers] Serweb > > sorry Marc, haven't idea what may be wrong. Please check again setting > of fifo in serweb. Check variable $config->fifo_server in > config_paths.php. It's value must be '/tmp/ser_fifo'. > > Try enable logging in serweb ($config->enable_loging = true in > config.php), set log level to debug ($config->log_level = > "PEAR_LOG_DEBUG") and examine the log file if there are some messages > refered fifo. > > Also try search for string 'fifo_server' throught log files (or whole > serweb) and check if it's value isn't changed somewhere. You should find > > this string only in file config_paths.php (one times) and in file > functions.php (four times). > > Karel > > Marc Khayat napsal(a): > >>Hi Karel, >>I already tried this one by manually changing the permissions on >>ser_fifo to 666; I just tried adding fifo_mode=0666 to ser.cfg, but >>still the same problem. >> >>Marc >>-----Original Message----- >>From: Karel Kozlik [mailto:karel at iptel.org <http://lists.iptel.org/mailman/listinfo/serusers> ] >>Sent: Friday, August 19, 2005 8:21 AM >>To: Marc Khayat >>Cc: serusers at iptel.org <http://lists.iptel.org/mailman/listinfo/serusers> >>Subject: Re: [Serusers] Serweb >> >>Hi, >>try add: >>fifo_mode=0666 >> >>to your ser.cfg >> >>Karel >> >>Marc Khayat napsal(a): >> >> >>>/tmp/ser_fifo exists, and is pipe type >>>User running Apache is "apache". Owner of ser_fifo is User: "root", >>>Group: "root". >>>I joined user "apache" to group "root", same. >>> >>>I created group "ser" and joined "root" "apache" to that group, and >>>added fifo_group=ser to the ser.cfg, but still same problem: >>> >>>Warning: filetype(): Lstat failed for (null) (errno=13 - Permission >>>denied) in /var/www/html/serweb/html/functions.php on line 306 >>> >>>Warning: Cannot modify header information - headers already sent by >>>(output started at /var/www/html/serweb/html/functions.php:306) in >>>/var/www/html/serweb/html/page.php on line 35 >>> >>> >>>Thanks, >>> >>>Marc >>> >>>-----Original Message----- >>>From: Karel Kozlik [mailto:karel at iptel.org <http://lists.iptel.org/mailman/listinfo/serusers> ] >>>Sent: Thursday, August 18, 2005 1:38 PM >>>To: Marc Khayat >>>Cc: serusers at iptel.org <http://lists.iptel.org/mailman/listinfo/serusers> >>>Subject: Re: [Serusers] Serweb >>> >>>Hello, >>>please check if file /tmp/ser_fifo exists and if it's type is pipe. >> >>Also >> >> >>>check rights - if user under which apache is running have access to >> >>read >> >> >>>and write to this file. >>> >>>Karel >>> >>>Marc Khayat napsal(a): >>> >>> >>> >>>>The domain thing is working, but problem 2 is still bugging me. >>>>Help anyone? >>>> >>>>Marc >>>> >>>>-----Original Message----- >>>>From: serusers-bounces at iptel.org <http://lists.iptel.org/mailman/listinfo/serusers> [mailto:serusers-bounces at iptel.org <http://lists.iptel.org/mailman/listinfo/serusers> ] >>> >>>On >>> >>> >>> >>>>Behalf Of Marc Khayat >>>>Sent: Thursday, August 18, 2005 11:15 AM >>>>To: serusers at iptel.org <http://lists.iptel.org/mailman/listinfo/serusers> >>>>Subject: RE: [Serusers] Serweb >>>> >>>>Never mind guys... it's working! >>>> >>>>Marc >>>>-----Original Message----- >>>>From: serusers-bounces at iptel.org <http://lists.iptel.org/mailman/listinfo/serusers> [mailto:serusers-bounces at iptel.org <http://lists.iptel.org/mailman/listinfo/serusers> ] >>> >>>On >>> >>> >>> >>>>Behalf Of Marc Khayat >>>>Sent: Thursday, August 18, 2005 10:53 AM >>>>To: serusers at iptel.org <http://lists.iptel.org/mailman/listinfo/serusers> >>>>Subject: [Serusers] Serweb >>>> >>>>Hi all, >>>>I'm using SER-0.9.3 and its serweb version. >>>>I have 2 problems with it: >>>>1- On the login page, when I enter the username, it automatically >>>>completes it with the FQDN, while I need only the domain name; i.e. I >>>>login to http://ser.xyz.com, I enter the username aloha, it's >>> >>>completed >>> >>> >>> >>>>as aloha at ser.xyz.com <http://lists.iptel.org/mailman/listinfo/serusers> while I need it as aloha at xyz.com. <http://lists.iptel.org/mailman/listinfo/serusers> Currently, I'm >>>>writing in manually >>>> >>>>2- When I login, I'm getting an error on the first page only as "FIFO >>>>not running or bad path to it". The path to it (in config_paths.php >>> >>>file >>> >>> >>> >>>>is /tmp/ser_fifo, which is correct), and when I enter the command >>> >>>serctl >>> >>> >>> >>>>monitor, the counter isn't constant, and same path. >>>> >>>>Can anyone help me plz, >>>> >>>>Thanks, >>>> >>>>Marc >>>> >>>>_______________________________________________ >>>>Serusers mailing list >>>>Serusers at iptel.org <http://lists.iptel.org/mailman/listinfo/serusers> >>>>http://lists.iptel.org/mailman/listinfo/serusers >>>> >>>> >>>>_______________________________________________ >>>>Serusers mailing list >>>>Serusers at iptel.org <http://lists.iptel.org/mailman/listinfo/serusers> >>>>http://lists.iptel.org/mailman/listinfo/serusers >>>> >>>> >>>>_______________________________________________ >>>>Serusers mailing list >>>>Serusers at iptel.org <http://lists.iptel.org/mailman/listinfo/serusers> >>>>http://lists.iptel.org/mailman/listinfo/serusers >>> >>> >>> >> > > > _______________________________________________ > Serusers mailing list > Serusers at iptel.org <http://lists.iptel.org/mailman/listinfo/serusers> > http://lists.iptel.org/mailman/listinfo/serusers > > > _______________________________________________ > Serusers mailing list > Serusers at iptel.org <http://lists.iptel.org/mailman/listinfo/serusers> > http://lists.iptel.org/mailman/listinfo/serusers > >

1 0

[Serusers] offline v unknown
by Kevin Brennan 31 Aug '05

31 Aug '05

Does anybody have any tips on solving the following I want to differenciate between offline users and unknown users. When you use lookup("location") there is no differenciation between the two. Does this require uri_db ? Using 0.8.14 Br /Kevin

5 26

[Serusers] Hairpin solutions?
by Federico Giannici 31 Aug '05

31 Aug '05

Considering that we DON'T want to use an RTP Proxy (bad scalability, waste of bandwidth and longer delays), is there any solution to the "hairpin" problems? Are there any "boxes" that look at the SIP messages and short-circuit calls from and to the local network? Thanks. -- ___________________________________________________ __ |- giannici(a)neomedia.it |ederico Giannici http://www.neomedia.it ___________________________________________________

7 10

[Users] nat flag and branch routes
by Klaus Darilion 31 Aug '05

31 Aug '05

Hi! The nat flag which will be loaded during lookup() - will it be set for each branch so it can be queried in branch_route for all of the registered contacts, or should I use isdsturiset() instead of testing for the NAT flag? regards klaus

5 10

[Serusers] Cancel'ing filure_route
by Dmitry Isakbayev 31 Aug '05

31 Aug '05

Hello, I need help understanding how to handle "CANCEL" messages. Here is the case - 1 Proxy rebuilds URI, sets 'failure_route', and executes 't_relay' 2 't_realy' times out and the proxy executes 'failure_route' after the fr_timer hits If the OUI cancels the call between steps 1 and 2, the proxy still executes the 'failure_route' after the fr_timer hits. Can I cancel any "t_on_failure" logic after receiving a CANCEL message? Or, while in in the 'failre_rout' block, can I figure out that there was a CANCEL message? Thank You, Dmitry

2 3

[Users] branchroute problem
by Klaus Darilion 31 Aug '05

31 Aug '05

Hi! In the following scenario, a call is forked to 3 phones. 2 of them are behind the same NAT (same publich IP, different port, one uses STUN, one not). I xlog all branches,dest-sets... in route[] and then in branch_route. But the branch route is executed on 2 times, instead of three times (the destination set shows that there are 3 destinations) Although the destination set is changed in the second branch processing. Is this the intended way? regards klaus route[]: request's first branch: $br=sip:10.10.0.50:15360 request's all branches: $bR=sip:10.10.0.50:15360, sip:klaus@83.136.33.3:8718 request's destination set: $ds=Contact: sip:klaus@83.136.33.19:5060, sip:10.10.0.50:15360, sip:klaus@83.136.33.3:8718 request's destination uri: $du= request's uri: $ru=sip:klaus@83.136.33.19:5060 activating t_on_branch(1) branch_route[1]: first branch processing branch_route[1]: request's first branch: $br=sip:10.10.0.50:15360 branch_route[1]: request's all branches: $bR=sip:10.10.0.50:15360, sip:klaus@83.136.33.3:8718 branch_route[1]: request's destination set: $ds=Contact: sip:klaus@83.136.33.19:5060, sip:10.10.0.50:15360, sip:klaus@83.136.33.3:8718 branch_route[1]: request's destination uri: $du= branch_route[1]: request's uri: $ru=sip:klaus@83.136.33.19:5060 branch_route[1]: second branch processing branch_route[1]: request's first branch: $br=sip:10.10.0.50:15360 branch_route[1]: request's all branches: $bR=sip:10.10.0.50:15360, sip:klaus@83.136.33.3:8718 branch_route[1]: request's destination set: $ds=Contact: sip:10.10.0.50:15360, sip:10.10.0.50:15360, sip:klaus@83.136.33.3:8718 branch_route[1]: request's destination uri: $du=sip:83.136.33.3:3003 branch_route[1]: request's uri: $ru=sip:10.10.0.50:15360 ...no third branch procesing :-(

2 5

[Serusers] REGISTER with negative Content-Length
by Hendrik Scholz 31 Aug '05

31 Aug '05

Hi! I've handcrafted a REGISTER that contains a negative Content-Length: from ngrep: # U x.x.x.x:5060 -> y.y.y.y:5060 REGISTER sip:123@mydomain.net SIP/2.0. Via: SIP/2.0/UDP x.x.x.x:5060. From: 123 <sip:123@mydomain.net:5060>;tag=33. To: 123 <sip:123@mydomain.net:5060>. Call-ID: 33-11277(a)x.x.x.x. CSeq: 1 REGISTER. Contact: sip:123@x.x.x.x:5060. Content-Length: -1000. Expires: 300. syslog content: Aug 31 11:04:48 xxx ser[3866]: ERROR:parse_content_length: parse error near char [45][-] Aug 31 11:04:48 xxx ser[3866]: ERROR:get_hdr_field: bad content_length header Aug 31 11:04:48 xxx ser[3866]: ERROR: build_res_buf_from_sip_req: alas, parse_headers failed Aug 31 11:04:48 xxx ser[3866]: pre_auth(): Error while sending 400 reply So the parser works fine but I cannot send an error message to prevent retransmissions from the UAC. The same is true for alphanumeric characters within the Content-Length: value. To test the Content-Length stuff even more I've set the Content-Length within the REGISTER to 1000 which yields a 'ERROR: t_newtran: EoH not parsed' warning but SER still sends the 407 that I was hoping for. Now the question is if there is any way to add something like a failure_route block to allow logging and sending a generic or customized 4xx/5xx error message to prevent retransmissions. Btw: My config is far from a stock one but it all comes down to a standard REGISTER with radius_www_authorize() in the REGISTER block. Any help appreciated, Hendrik -- freenet Cityline GmbH, Hamburger Chaussee 2-4, 24114 Kiel, Germany Phone: +49 (0)431 9020552, Fax: +49 (0)431 9020559 Internet: http://www.freenet.de, eMail: hendrik.scholz(a)freenet-ag.de

1 0

RE: [Serusers] Bye messages not logged
by Marc Khayat 31 Aug '05

31 Aug '05

Anyone can help me with the BYE messages that are not being logged? Marc -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Marc Khayat Sent: Tuesday, August 30, 2005 12:26 PM To: serusers(a)lists.iptel.org Subject: [Serusers] Bye messages not logged Hi all, I upgraded my system from ser-0.8.14 to ser-0.9.3, and updated my ser.cfg. However, with the new thing, I'm getting INVITE and ACK messages, but no the BYE nor the REGISTER messages. I made no changes to the config file, only added the accounting requirements. But is the below stateful? Here's my ser.cfg: loadmodule "/usr/local/lib/ser/modules/acc.so" modparam("acc", "log_flag", 1 ) modparam("acc", "log_fmt", "miocfst") modparam("acc", "failed_transactions", 1) modparam("acc", "log_missed_flag", 2) modparam("acc", "report_cancels", 1) modparam("acc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("acc", "db_flag", 5) modparam("acc", "db_missed_flag", 6) # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages within a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; setflag(5); setflag(6); if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="INVITE") { if (uri=~"^sip:9") { t_relay_to_udp("50.65.11.23", "5060"); break; }; }; if (method=="REGISTER") { # Uncomment this if you want to use digest authentication # if (!www_authorize("mydomain.net", "subscriber")) { # www_challenge("mydomain.net", "0"); # break; # }; save("location"); break; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

1 0

[Users] AVPops trouble
by Martin Kjeldsen 31 Aug '05

31 Aug '05

Hi I've been fiddeling with the AVPops module for some time now and havn't really gotten it to work as intended. I was hoping that someone out there might be able to explain the basics of the module, cause apparently I'm misunderstanding it... I think :-) I having trouble with pulling data from my openser DB with a scheme, and then pushing (pushto) it into a request header. The thing is I cant quite figure out, if I need to store the data I get from my scheme or if I can use it directly in a pushto! I have tried using a scheme: modparam("avpops", "db_scheme", "scheme0:uuid_col=contact;value_col=username;value_type=string;table=aliases") and then loading and pushing it with: avp_db_load("$from","s:pai/$scheme0"); avp_pushto("$P-Asserted-Identity","s:pai"); But this leaves me with no new field in the header, and nothing to show for in my MySQL log! I'm using these additional params for the AVPops: modparam("avpops","avp_url","mysql://openser:heslo@localhost/openser") modparam("avpops","avp_table","avptable") modparam("avpops","use_domain",1) The avptable is one I've created, since I wasn't sure if it needed to be there or not. Any and all help will be greatly appreciated, cause I'm really getting stuck here. Regards Martin

2 2

[Serusers] caller id with name
by Tulika Pradhan 31 Aug '05

31 Aug '05

hi ! has anyone implemented the caller id with name feature on ser ? regards, tulika

1 0

[Serusers] SER's nathelper
by Frank Kostin 31 Aug '05

31 Aug '05

Hi everybody, I have two devices behind a NAT and with different codecs that can not agree upon a common codec and can not make it working. Anyone have a sample config, using SER's nathelper in conjunction with Asterisk to rewrite the SDP so that the private IP addresses are not included in the re-invite. Thanks Frank --------------------------------- Yahoo! Mail Stay connected, organized, and protected. Take the tour

1 0

[Serusers] serweb msilo ser problem
by Walter Willis 31 Aug '05

31 Aug '05

i am problem with msilo and serweb :) i have installed the ser (cvs) and serweb, send IM put any thing in sip address of recipient: and send the message and view: sadasd no exist. Message has been successfully sended to address sip:sadasd@domain.com and debug the SER: 3(9953) Warning: run_failure_handlers: no UAC support (0, 0) the scrit ser: route{ if ( !mf_process_maxfwd_header("10") ) { sl_send_reply("483","To Many Hops"); drop(); }; if (uri==myself) { { # for testing purposes, simply okay all REGISTERs if (method=="REGISTER") { save("location"); log("REGISTER received -> dumping messages with MSILO\n"); # MSILO - dumping user's offline messages if (m_dump()) { log("MSILO: offline messages dumped - if they were\n"); }else{ log("MSILO: no offline messages dumped\n"); }; break; }; # domestic SIP destinations are handled using our USRLOC DB if(!lookup("location")) { if (! t_newtran()) { sl_reply_error(); break; }; # we do not care about anything else but MESSAGEs if (!method=="MESSAGE") { if (!t_reply("404", "Not found")) { sl_reply_error(); }; break; }; log("MESSAGE received -> storing using MSILO\n"); # MSILO - storing as offline message if (m_store("0")) { log("MSILO: offline message stored\n"); if (!t_reply("202", "Accepted")) { sl_reply_error(); }; }else{ log("MSILO: offline message NOT stored\n"); if (!t_reply("503", "Service Unavailable")) { sl_reply_error(); }; }; break; }; # if the downstream UA does not support MESSAGE requests # go to failure_route[1] t_on_failure("1"); t_relay(); break; }; # forward anything else t_relay(); } failure_route[1] { # forwarding failed -- check if the request was a MESSAGE if (!method=="MESSAGE") { break; }; log(1,"MSILO:the downstream UA doesn't support MESSAGEs\n"); # we have changed the R-URI with the contact address, ignore it now if (m_store("1")) { log("MSILO: offline message stored\n"); t_reply("202", "Accepted"); }else{ log("MSILO: offline message NOT stored\n"); t_reply("503", "Service Unavailable"); }; } wath is the problem????

2 3

[Serusers] OT: Monitoring tools
by Daniel Corbe 31 Aug '05

31 Aug '05

Hello, I'm going to attempt to build a box that listens on a SPAN switch port which has all my proxy traffic mirrored to it. I was curious if anyone has ever attempted something like this before, and what software/tools were used to listen for signaling (SIP) traffic and subsequently log the traffic out to disk. -Daniel

2 2

Re: [Serusers] NAT example
by Kamran Ahmad 30 Aug '05

30 Aug '05

hello Why uri!=myself what could be the reason using "SER 0.9.3 source package for Getting Started" from Onsip.org when i try to use port 5060 then it only work for public ip UA and dont work for nated users. But when i change port to 5069 it try to listen for nated UA but the problem is still why uri!=myself i am using and for configuration ftp://siprouter.onsip.org/pub/gettingstarted/configs/nat-rtpproxy.3.05.cfg after lot of changes in ports exec_msg("cat >> /tmp/call.log"); if (uri!=myself) { xlog("L_ERR","uri!=myself"); route(1); break; }; i found this /var/log/messages -------------------------------------------------- Aug 30 21:58:09 SIP ser[10560]: uri!=myself and /tmp/call.log -------------------------------------------------- REGISTER sip:SERPUBIP SIP/2.0 Content-Length: 0 Contact: <sip:8000@192.168.0.21:5060>;events="message-summary" Call-ID: B692E8F0-DDC4-42A7-8A87-79B8133FA79B(a)192.168.0.21 Max-Forwards: 16 From: <sip:8000@SERPUBIP>;tag=587922581424 CSeq: 5 REGISTER To: <sip:8000@SERPUBIP> Via: SIP/2.0/UDP 203.99.58.250:52904;rport;branch=z9h8001009User-Agent: SJLabs-SJphone/1.40.258 --- Samuel Osorio Calvo <samuel.osorio(a)nl.thalesgroup.com> wrote: > Don't worry, > These messages comes from the startup process, when > SER tries to get the PTR DNS records of the IP > addresses present in the config file. These pointers > map IP addresses to domain names, instead of the > "standard" domain->IP mapping provided by the A|AAA > DNS records. > > Probably your SER instance is running, have you > tried ps?? > > Enjoy! > > Samuel. > > > Unclassified. > >>> Kamran Ahmad <p_kami(a)yahoo.com> 08/30/05 10:35AM > >>> > i tried this > > ftp://siprouter.onsip.org/pub/gettingstarted/configs/nat-rtpproxy.3.05.cfg > > > but it is not working > > /var/log/messages > Aug 30 11:12:42 SIP ser: WARNING: fix_socket_list: > could not rev. resolve ip > Aug 30 11:12:42 SIP ser: WARNING: fix_socket_list: > could not rev. resolve ip > > > --- "Greger V. Teigre" <greger(a)teigre.com> wrote: > > > Try to Getting Started document from > > http://onsip.org/ You will find both a > > working config, as well as detailed explanation of > > the theory and actual > > commands. > > g-) > > > > Kamran Ahmad wrote: > > > Hello > > > > > > i am following this link for ser-0.9.3 but Nated > > > Clients > > > are not working. Can any one please tell > > me > > > what is the problem in my configuration. > > > > > > > > > http://www.voip-info.org/tiki-index.php?page=SER+nat+support > > > > > > > __________________________________________________ > Do You Yahoo!? > Tired of spam? Yahoo! Mail has the best spam > protection around > http://mail.yahoo.com > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > ____________________________________________________ Start your day with Yahoo! - make it your home page http://www.yahoo.com/r/hs

1 0

Re: [Users] #DEFINE equivalent?
by Kanakatti Mahesh Subramanya 30 Aug '05

30 Aug '05

because i didn't think of it? :-) cheers, and thanx Bogdan-Andrei Iancu wrote: > Kanakatti Mahesh Subramanya wrote: > >> I totally agree. >> >> Personally, I've been ripping out large chunks of my config file, and >> replacing it with varioud avpops based stuff - it may be *slightly* >> more convoluted, but it is completely database configurable, and >> hence easier to maintain >> >> re: rr, its actually neither. Its a static, which i'm using to >> identify NAT'ed clients to account for re-INVITEs > > > why don't you use the append_rr_param() function to added the NAT > identifier to the RR hdr? do a normal record_route() and if nat is > detected, add the param > > regards, > bogdan > >> >> cheers >> >> Bogdan-Andrei Iancu wrote: >> >>> Hi Kanakatti, >>> >>> I'm for m4'ing (for the moment) for two reasons: >>> 1) since we can do it with m4, I prefer to use the cycles for >>> something still missing; >>> 2) using dynamic parameters (like via AVPs) reduce the >>> performance - time to look for the AVP and most critical the >>> impossibility of calling fixup functions at startup (which will >>> force to fix params at runtime, for each processing). >>> >>> regarding the record_route_preset() - do you use dynamic parameters >>> or dynamic rr URI? >>> >>> regards, >>> bogdan >> >>

1 0

[Serusers] Reviewers needed for Open Solaris SIP API design review
by Rao Shoaib 30 Aug '05

30 Aug '05

Dear Colleagues, We are conducting a community design review for the open solaris SIP API at http://www.opensolaris.org/jive/thread.jspa?threadID=1936&tstart=0 We are very interested in hearing your comments on the design. Please post your comments at the above location. Thanks. Rao.

1 0

[Serusers] supporting multiple calls from one user
by Iqbal 30 Aug '05

30 Aug '05

Hi Does ser support dialing to multiple destinations using one username/password, especially for setups where hardware may have multiple voice channels, but only space for one username. Anyone have any test setups Iqbal

1 0

[Serusers] SER outgoing through TCP?
by Pat Wang 30 Aug '05

30 Aug '05

Hi, Does anyone know how to configure SER proxy signalling through TCP? For example, UAC1 -------> SER -------> UAC2, I would like to have SER proxy the INVITE (and the following) message to UAC2 using TCP. Is that possible and how to setup on ser.cfg? Or if it possible to do TCP signalling between two SERs: SER ------TCP---------SER? Thanks a million, Pat

2 1

[Serusers] Routing - Looking up database table for subscriber accounts not registered at the moment
by Frank Fischer 30 Aug '05

30 Aug '05

Hi all i just started working with SIP and i'm currently trying to implement the routing logic as needed in my setup. Doing this i came across a requirement i couldn't find a solution for. Maybe someone of you may help me? I have the following situation: I have users with (only) e164Numbers as usernames in my SIP domain and i have a PSTN Gateway behind which i may reach also e164Numbers. Now, i have a subscriber base in my domain where not all of the subscribers with a e164Number may be registered at a certain moment (but still have a valid account and phonenumber). When i get a call at such a moment, i will try to look up the callee username in the location database. If i find one, fine. If not, there are two possibilities left - first, it could be a user only reachable in the pstn, so route to the pstn gateway or second it might be a user in my domain with a valid account but not registered at the moment, so reject or redirect to a voicemail or what ever. If I would work with alphanumerical numbers only within my SIP cloud, i knew how this could be done - all numerical numbers automatically would be PSTN numbers. But how can this be done in my setup? Thanks a lot for your help and kind Regards Frank

2 1

[Serusers] ACK
by Sebastian Kühner 30 Aug '05

30 Aug '05

Hi! I noticed a "phenomenon" that I don't understand: I make accounting with ser. If all works fine, ser puts into the database: sip_method: ACK i_uri: sip:0182011542614xxxxxx@xxx.xxx.xxx.xxx:5060 (xxx.xxx.xxx.xxx = IP of PSTN-Gateway) ... and after the call, the BYE message reaches my server. But sometimes, the ACK is logged in the database like this (in the most cases form users "far away"): sip_method: ACK i_uri: sip:0182011542614xxxxxx@my-ip:5060 In this case, the BYE message doesn't reach my server. Does anybody know what could be the reason and why ser logs the ACK from himself? Many thanks! Sebastian

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Re: [Serusers] Radius Server sending reply to another machine!!!
by Ashutosh Kumar 30 Aug '05

30 Aug '05

Many thanks Jan, Adding bindaddr directive to radiusclient.conf worked!!!!. Now, the problem remains as to why is the radiusclient/ser picking up the dictionary.ascend, when I have included the dictionary.ser at the end of my dictionary file. Also, I have auth_radius:service_type modparamed' to value of 15. Please guide. Regards, Ashutosh Kumar -----Original Message----- From: Jan Janak [mailto:jan@iptel.org] Sent: Tuesday, August 30, 2005 2:40 PM To: Ashutosh Kumar Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] Radius Server sending reply to another machine!!! Check the configuration file of the radius client library, perhaps you have set the bindaddr parameter to the wrong IP address ? Jan. On 30-08-2005 03:57, Ashutosh Kumar wrote: > Hi, > I am facing a strange bewildering problem out here. Earlier I was > running ser -0.9 on machine a.b.c.116. Now we shifted it to a.b.c.113 for > testing for radius testing purposes. > But in the radius packet dump, it is showing as the radius packet still > originating from a.b.c.116 machine, despite the fact, my new revised ser.cfg > doesn't contain any listen address or alias for .116 box. > Also, the /etc/hosts doesn't mention .116 address. Also, it is not the case > that .116 and .113 are IPs homes on same machine. My ser.cfg code for radius > authorization is as follows. > > ... > ..... > > if (!radius_www_authorize("")) { > # if (!www_authorize("","subscriber")) { > www_challenge("","0"); > break; > }; > ... > ..... > > In the DB also, there is no table . The Domains table has no records. But > still SER is usng .116 address to send radius packets from. Here is the > radius packet dump. > > [root@sip radiusclient]# tcpdump -i eth1 -s 1500 -w - udp | raddump > -vvvvvvvvvvvvvvvvvvvvvsn > tcpdump: listening on eth1 > Looks like a(n) libpcap/tcpdump packet trace > 17:45:13.051205 radius a.b.c.113:50912 > p.q.r.s:1812 Access-Request 180 > User-Name = test(a)a.b.c.112 > Ascend-PW-Warntime = 168195173 > Ascend-PW-Warntime = 17774137 > Ascend-PW-Warntime = 36320307 > Ascend-PW-Warntime = 68383593 > Ascend-PW-Warntime = 51008069 > Ascend-Menu-Item = 75dd216d3efa9bfb9c2d7e013d54f7bb > Service-Type = 15 (unknown) > Ascend-PW-Lifetime = 1952805748 > NAS-IP-Address = a.b.c.116 > NAS-Port = 5060 > > (Is there any table /file in ser other than ser.cfg or mysql://ser that > contains the domain information that might have been lying there unchanged > since machine migration?) > > Also, why is SER using incorrect SER dictionary NAS client (Ascend)( > shouldn't it have been Cisco or sometinhg like that). I have included the > dictionary.ser included in ser-0.9 source in the radiusclient conf file. > > Help me, its driving me crazy. > > > Regards, > Ashutosh Kumar > > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers

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[Users] CANCEL problem
by Klaus Darilion 30 Aug '05

30 Aug '05

Hi! I just having problems cancelling forked calls. openser forwards the incoming CANCEL to both phones. As you see, openser added 2 via headers to the last CANCEL. The via header of branch 1 and branch 2. Maybe something got wrong when introducing branch routes? I'm using todays CVS. klaus 83.136.33.3:1200 -> 83.136.32.83:5060 CANCEL sip:klaus@enum.at SIP/2.0 Via: SIP/2.0/UDP 10.10.0.76:8418 Max-Forwards: 70 From: "klaus.darilion(a)nic.at43.at" <sip:klaus.darilion@nic.at43.at>;tag=b3669e387d784f1dbd080f83fdd62306;epid=9012264a99 To: <sip:klaus@enum.at> Call-ID: e654c424b1124ce29b93e7dd6a8ad44c CSeq: 2 CANCEL User-Agent: RTC/1.3 Content-Length: 0 # U 83.136.32.83:5060 -> 83.136.33.19:5060 CANCEL sip:klaus@83.136.33.19:5060 SIP/2.0 Record-Route: <sip:83.136.32.83;ftag=b3669e387d784f1dbd080f83fdd62306;nat=yes;lr=on> Via: SIP/2.0/UDP 83.136.32.83;branch=z9hG4bK131f.f3d7378.0 Via: SIP/2.0/UDP 10.10.0.76:8418;rport=1200;received=83.136.33.3 Max-Forwards: 16 From:"klaus.darilion(a)nic.at43.at" <sip:klaus.darilion@nic.at43.at>;tag=b3669e387d784f1dbd080f83fdd62306;epid=9012264a99 To: <sip:klaus@enum.at> Call-ID: e654c424b1124ce29b93e7dd6a8ad44c CSeq: 2 CANCEL User-Agent: RTC/1.3 Content-Length: 0 P-Behind-NAT: Yes # U 83.136.32.83:5060 -> 83.136.33.3:8718 CANCEL sip:klaus@10.10.0.50:8718 SIP/2.0 Record-Route: <sip:83.136.32.83;ftag=b3669e387d784f1dbd080f83fdd62306;nat=yes;lr=on> Via: SIP/2.0/UDP 83.136.32.83;branch=z9hG4bK131f.f3d7378.0 Via: SIP/2.0/UDP 83.136.32.83;branch=z9hG4bK131f.f3d7378.1 Via: SIP/2.0/UDP 10.10.0.76:8418;rport=1200;received=83.136.33.3;rport=1200;received=83.136.33.3 Max-Forwards: 16 From: "klaus.darilion(a)nic.at43.at" <sip:klaus.darilion@nic.at43.at>;tag=b3669e387d784f1dbd080f83fdd62306;epid=9012264a99 To: <sip:klaus@enum.at> Call-ID: e654c424b1124ce29b93e7dd6a8ad44c CSeq:2 CANCEL User-Agent: RTC/1.3 Content-Length: 0 P-Behind-NAT: Yes

2 2

[Serusers] 1 SER & 2 Providers
by Ozan Blotter 30 Aug '05

30 Aug '05

Dear List, Assume that there are two different service providers and one SER in middle of them. A client needs to use/connect SER but also he/she has accounts in both providers and both supplied him/her DDIs (landline phone). Now again assume that client registers both of phones into SER. But still he/she wants to receive calls from both providers, is it possible to receive calls from different providers to that client and/or how to i setup ? In current SER.CFG below, i only setup one prefix which is "0", but i can change it to any and/or add many prefixes, ex: 0049 (germany), 001 (usa), 0044 (uk), 0090 (turkey) to Provider 1 and 0033 (france), 0038 (italy) to Provider 2 and so on. debug=3 fork=yes log_stderror=no listen=212.XXX.104.XXX # This is SER's Public IP port=5060 children=4 alias=212.XXX.104.XXX # This is SER's Public IP dns=no rev_dns=no fifo="/tmp/openser_fifo" fifo_db_url="mysql://openser:openserrw@localhost/openser" loadmodule "/usr/local/lib/openser/modules/mysql.so" loadmodule "/usr/local/lib/openser/modules/sl.so" loadmodule "/usr/local/lib/openser/modules/tm.so" loadmodule "/usr/local/lib/openser/modules/rr.so" loadmodule "/usr/local/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/lib/openser/modules/usrloc.so" loadmodule "/usr/local/lib/openser/modules/registrar.so" loadmodule "/usr/local/lib/openser/modules/auth.so" loadmodule "/usr/local/lib/openser/modules/auth_db.so" loadmodule "/usr/local/lib/openser/modules/uri.so" loadmodule "/usr/local/lib/openser/modules/uri_db.so" loadmodule "/usr/local/lib/openser/modules/mediaproxy.so" loadmodule "/usr/local/lib/openser/modules/nathelper.so" loadmodule "/usr/local/lib/openser/modules/textops.so" loadmodule "/usr/local/lib/openser/modules/domain.so" loadmodule "/usr/local/lib/openser/modules/acc.so" modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("auth_db", "use_domain", 1) modparam("domain", "db_mode", 1) modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 180) modparam("mediaproxy","natping_interval", 30) modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/openser/sip-asymmetric-clients") modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/openser/rtp-asymmetric-clients") modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", 1) modparam("registrar", "default_expires", 60) modparam("registrar", "min_expires", 30) modparam("registrar", "nat_flag", 6) modparam("registrar", "use_domain", 1) modparam("rr", "enable_full_lr", 1) modparam("acc", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("acc", "failed_transactions", 1) modparam("acc", "log_level", 1) modparam("acc", "log_flag", 1) modparam("acc", "db_flag", 1) route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- 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; }; # ----------------------------------------------------------------- # Record Route Section and Acc section # ----------------------------------------------------------------- if (method=="INVITE" && client_nat_test("3")) { record_route_preset("212.XXX.104.XXX:5060;nat=yes"); # This is SER's Public IP } else if (method!="REGISTER") { if!(uri=~"^sip:833[0-9]*@") { # I do not want to make accounting if clients' prefix is beginning with 833 record_route(); setflag(1); } }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { end_media_session(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- 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; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- 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 Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- if (!search("^Contact:\ +\*") && client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); }; sl_send_reply("100", "Trying"); if (!www_authorize("","subscriber")) { www_challenge("","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # CANCEL and INVITE Message Handler # ----------------------------------------------------------------- if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; lookup("aliases"); if (uri!=myself) { route(1); break; }; if (uri=~"^sip:0[0-9]*@*") { # If calling number is beginning with prefix 0, route it to PSTN rewritehost("195.XXX.122.XXX"); # This is PSTN Gateway's Public IP route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; if (method=="CANCEL") { route(1); break; }; if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); if (isflagset(6) || isflagset(7)) { use_media_proxy(); }; route(1); } onreply_route[1] { if ((isflagset(6) || isflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { if (!search("^Content-Length:\ +0")) { use_media_proxy(); }; }; if (client_nat_test("1")) { fix_nated_contact(); }; }

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[Serusers] clarification about authentication
by Iqbal 30 Aug '05

30 Aug '05

Hi www_authorize , will this always fail, ? because it seems to and then a www.challenge is sent. In www.challenge, is it okay to leave the realm blank, as I have done, or is it better practice to have the same realm as that in www_authorise("", "1"); check_to should only be run if auth is correct above , is that correct? Why run check_to at all, ? and then u have proxy_auth and proxy_challenge, and is_from_local and is_uri_local, surely all these dont need to be used as checks in INVITE or do they. Iqbal

2 2

[Serusers] Bye messages not logged
by Marc Khayat 30 Aug '05

30 Aug '05

Hi all, I upgraded my system from ser-0.8.14 to ser-0.9.3, and updated my ser.cfg. However, with the new thing, I'm getting INVITE and ACK messages, but no the BYE nor the REGISTER messages. I made no changes to the config file, only added the accounting requirements. But is the below stateful? Here's my ser.cfg: loadmodule "/usr/local/lib/ser/modules/acc.so" modparam("acc", "log_flag", 1 ) modparam("acc", "log_fmt", "miocfst") modparam("acc", "failed_transactions", 1) modparam("acc", "log_missed_flag", 2) modparam("acc", "report_cancels", 1) modparam("acc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("acc", "db_flag", 5) modparam("acc", "db_missed_flag", 6) # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages within a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; setflag(5); setflag(6); if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="INVITE") { if (uri=~"^sip:9") { t_relay_to_udp("50.65.11.23", "5060"); break; }; }; if (method=="REGISTER") { # Uncomment this if you want to use digest authentication # if (!www_authorize("mydomain.net", "subscriber")) { # www_challenge("mydomain.net", "0"); # break; # }; save("location"); break; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; }

1 0

Re: [Serusers] NAT example
by Kamran Ahmad 30 Aug '05

30 Aug '05

Thanks for reply [root@SIP ~]# ps -A | grep ser 131 ? 00:00:00 kseriod when i try to use asterisk on same port it is working. means port is free. it tried to change port from5060 to 5066. i have used killall ser before starting ser i tried to put in log into route2 but request is not reaching route2 route[2] { xlog("L_ERR","inside route2"); --- Samuel Osorio Calvo <samuel.osorio(a)nl.thalesgroup.com> wrote: > Don't worry, > These messages comes from the startup process, when > SER tries to get the PTR DNS records of the IP > addresses present in the config file. These pointers > map IP addresses to domain names, instead of the > "standard" domain->IP mapping provided by the A|AAA > DNS records. > > Probably your SER instance is running, have you > tried ps?? > > Enjoy! > > Samuel. > > > Unclassified. > >>> Kamran Ahmad <p_kami(a)yahoo.com> 08/30/05 10:35AM > >>> > i tried this > > ftp://siprouter.onsip.org/pub/gettingstarted/configs/nat-rtpproxy.3.05.cfg > > > but it is not working > > /var/log/messages > Aug 30 11:12:42 SIP ser: WARNING: fix_socket_list: > could not rev. resolve ip > Aug 30 11:12:42 SIP ser: WARNING: fix_socket_list: > could not rev. resolve ip > > > --- "Greger V. Teigre" <greger(a)teigre.com> wrote: > > > Try to Getting Started document from > > http://onsip.org/ You will find both a > > working config, as well as detailed explanation of > > the theory and actual > > commands. > > g-) > > > > Kamran Ahmad wrote: > > > Hello > > > > > > i am following this link for ser-0.9.3 but Nated > > > Clients > > > are not working. Can any one please tell > > me > > > what is the problem in my configuration. > > > > > > > > > http://www.voip-info.org/tiki-index.php?page=SER+nat+support > > > > > > > __________________________________________________ > Do You Yahoo!? > Tired of spam? Yahoo! Mail has the best spam > protection around > http://mail.yahoo.com > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

1 0

[Serusers] Different Path for Request and Reply
by Andreas Granig 30 Aug '05

30 Aug '05

Hi, Let's say there are two SER servers A and B, a UAC and a UAS. If a request from UAC to UAS is routed via A and right after that A fails and B takes over, which impacts does it have if 1xx/200 replies from UAS to UAC are routed via B? The replies could theoretically be routed correctly because the next hop is available in the Via header. But since there is no transaction stored in SER for this, does this work? And what about accounting? What other functionality would be affected? Andy

2 1

[Serusers] Radius Server sending reply to another machine!!!
by Ashutosh Kumar 30 Aug '05

30 Aug '05

Hi, I am facing a strange bewildering problem out here. Earlier I was running ser -0.9 on machine a.b.c.116. Now we shifted it to a.b.c.113 for testing for radius testing purposes. But in the radius packet dump, it is showing as the radius packet still originating from a.b.c.116 machine, despite the fact, my new revised ser.cfg doesn't contain any listen address or alias for .116 box. Also, the /etc/hosts doesn't mention .116 address. Also, it is not the case that .116 and .113 are IPs homes on same machine. My ser.cfg code for radius authorization is as follows. ... ..... if (!radius_www_authorize("")) { # if (!www_authorize("","subscriber")) { www_challenge("","0"); break; }; ... ..... In the DB also, there is no table . The Domains table has no records. But still SER is usng .116 address to send radius packets from. Here is the radius packet dump. [root@sip radiusclient]# tcpdump -i eth1 -s 1500 -w - udp | raddump -vvvvvvvvvvvvvvvvvvvvvsn tcpdump: listening on eth1 Looks like a(n) libpcap/tcpdump packet trace 17:45:13.051205 radius a.b.c.113:50912 > p.q.r.s:1812 Access-Request 180 User-Name = test(a)a.b.c.112 Ascend-PW-Warntime = 168195173 Ascend-PW-Warntime = 17774137 Ascend-PW-Warntime = 36320307 Ascend-PW-Warntime = 68383593 Ascend-PW-Warntime = 51008069 Ascend-Menu-Item = 75dd216d3efa9bfb9c2d7e013d54f7bb Service-Type = 15 (unknown) Ascend-PW-Lifetime = 1952805748 NAS-IP-Address = a.b.c.116 NAS-Port = 5060 (Is there any table /file in ser other than ser.cfg or mysql://ser that contains the domain information that might have been lying there unchanged since machine migration?) Also, why is SER using incorrect SER dictionary NAS client (Ascend)( shouldn't it have been Cisco or sometinhg like that). I have included the dictionary.ser included in ser-0.9 source in the radiusclient conf file. Help me, its driving me crazy. Regards, Ashutosh Kumar

2 1

[Serusers] "Makefile.rules:80: action.d: No such file or directory" Errors
by Jaroslaw Gawron 30 Aug '05

30 Aug '05

Hello, I try to install ser-0.9.3 from source file on Mandriva 2005 limited edition and I get the following errors : Makefile.rules:80: action.d: No such file or directory Makefile.rules:80: crc.d: No such file or directory Makefile.rules:80: daemonize.d: No such file or directory Makefile.rules:80: data_lump.d: No such file or directory Makefile.rules:80: data_lump_rpl.d: No such file or directory Makefile.rules:80: dprint.d: No such file or directory Makefile.rules:80: dset.d: No such file or directory Makefile.rules:80: error.d: No such file or directory Makefile.rules:80: fifo_server.d: No such file or directory Makefile.rules:80: flags.d: No such file or directory Makefile.rules:80: forward.d: No such file or directory Makefile.rules:80: hash_func.d: No such file or directory Makefile.rules:80: ip_addr.d: No such file or directory Makefile.rules:80: main.d: No such file or directory Makefile.rules:80: md5.d: No such file or directory Makefile.rules:80: md5utils.d: No such file or directory Makefile.rules:80: modparam.d: No such file or directory Makefile.rules:80: msg_translator.d: No such file or directory Makefile.rules:80: pass_fd.d: No such file or directory Makefile.rules:80: proxy.d: No such file or directory Makefile.rules:80: qvalue.d: No such file or directory Makefile.rules:80: re.d: No such file or directory Makefile.rules:80: receive.d: No such file or directory Makefile.rules:80: resolve.d: No such file or directory Makefile.rules:80: route.d: No such file or directory Makefile.rules:80: route_struct.d: No such file or directory Makefile.rules:80: script_cb.d: No such file or directory Makefile.rules:80: socket_info.d: No such file or directory Makefile.rules:80: sr_module.d: No such file or directory Makefile.rules:80: stats.d: No such file or directory Makefile.rules:80: tcp_main.d: No such file or directory Makefile.rules:80: tcp_read.d: No such file or directory Makefile.rules:80: timer.d: No such file or directory Makefile.rules:80: tsend.d: No such file or directory Makefile.rules:80: udp_server.d: No such file or directory Makefile.rules:80: unixsock_server.d: No such file or directory Makefile.rules:80: usr_avp.d: No such file or directory Makefile.rules:80: mem/f_malloc.d: No such file or directory Makefile.rules:80: mem/mem.d: No such file or directory Makefile.rules:80: mem/memtest.d: No such file or directory Makefile.rules:80: mem/q_malloc.d: No such file or directory Makefile.rules:80: mem/shm_mem.d: No such file or directory Makefile.rules:80: mem/vq_malloc.d: No such file or directory Makefile.rules:80: parser/hf.d: No such file or directory Makefile.rules:80: parser/msg_parser.d: No such file or directory Makefile.rules:80: parser/parse_allow.d: No such file or directory Makefile.rules:80: parser/parse_content.d: No such file or directory Makefile.rules:80: parser/parse_cseq.d: No such file or directory Makefile.rules:80: parser/parse_disposition.d: No such file or directory Makefile.rules:80: parser/parse_diversion.d: No such file or directory Makefile.rules:80: parser/parse_event.d: No such file or directory Makefile.rules:80: parser/parse_expires.d: No such file or directory Makefile.rules:80: parser/parse_fline.d: No such file or directory Makefile.rules:80: parser/parse_from.d: No such file or directory Makefile.rules:80: parser/parse_hname2.d: No such file or directory Makefile.rules:80: parser/parse_hostport.d: No such file or directory Makefile.rules:80: parser/parse_methods.d: No such file or directory Makefile.rules:80: parser/parse_nameaddr.d: No such file or directory Makefile.rules:80: parser/parse_param.d: No such file or directory Makefile.rules:80: parser/parser_f.d: No such file or directory Makefile.rules:80: parser/parse_rpid.d: No such file or directory Makefile.rules:80: parser/parse_rr.d: No such file or directory Makefile.rules:80: parser/parse_sipifmatch.d: No such file or directory Makefile.rules:80: parser/parse_to.d: No such file or directory Makefile.rules:80: parser/parse_uri.d: No such file or directory Makefile.rules:80: parser/parse_via.d: No such file or directory Makefile.rules:80: parser/digest/digest.d: No such file or directory Makefile.rules:80: parser/digest/digest_parser.d: No such file or directory Makefile.rules:80: parser/digest/param_parser.d: No such file or directory Makefile.rules:80: parser/contact/contact.d: No such file or directory Makefile.rules:80: parser/contact/parse_contact.d: No such file or directory Makefile.rules:80: db/db.d: No such file or directory Makefile.rules:80: db/db_fifo.d: No such file or directory Makefile.rules:80: lex.yy.d: No such file or directory Does anyone know how to fix this problem ? Any help will be great ! Best regards, Jarosław Gawron

2 1

Re: [Serusers] NAT example
by Samuel Osorio Calvo 30 Aug '05

30 Aug '05

Don't worry, These messages comes from the startup process, when SER tries to get the PTR DNS records of the IP addresses present in the config file. These pointers map IP addresses to domain names, instead of the "standard" domain->IP mapping provided by the A|AAA DNS records. Probably your SER instance is running, have you tried ps?? Enjoy! Samuel. Unclassified. >>> Kamran Ahmad <p_kami(a)yahoo.com> 08/30/05 10:35AM >>> i tried this ftp://siprouter.onsip.org/pub/gettingstarted/configs/nat-rtpproxy.3.05.cfg but it is not working /var/log/messages Aug 30 11:12:42 SIP ser: WARNING: fix_socket_list: could not rev. resolve ip Aug 30 11:12:42 SIP ser: WARNING: fix_socket_list: could not rev. resolve ip --- "Greger V. Teigre" <greger(a)teigre.com> wrote: > Try to Getting Started document from > http://onsip.org/ You will find both a > working config, as well as detailed explanation of > the theory and actual > commands. > g-) > > Kamran Ahmad wrote: > > Hello > > > > i am following this link for ser-0.9.3 but Nated > > Clients are not working. Can any one please tell > me > > what is the problem in my configuration. > > > > > http://www.voip-info.org/tiki-index.php?page=SER+nat+support > > __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

1 0

Re: [Serusers] "Makefile.rules:80: action.d: No such file or directory"Errors
by Samuel Osorio Calvo 30 Aug '05

30 Aug '05

Don't worry, this files are created after the first compilation. Probably SER is already properly compiled in your host. Enjoy! Samuel. Unclassified. >>> "Jaroslaw Gawron" <j.gawron(a)wasko.pl> 08/30/05 10:20AM >>> Hello, I try to install ser-0.9.3 from source file on Mandriva 2005 limited edition and I get the following errors : Makefile.rules:80: action.d: No such file or directory Makefile.rules:80: crc.d: No such file or directory Makefile.rules:80: daemonize.d: No such file or directory Makefile.rules:80: data_lump.d: No such file or directory Makefile.rules:80: data_lump_rpl.d: No such file or directory Makefile.rules:80: dprint.d: No such file or directory Makefile.rules:80: dset.d: No such file or directory Makefile.rules:80: error.d: No such file or directory Makefile.rules:80: fifo_server.d: No such file or directory Makefile.rules:80: flags.d: No such file or directory Makefile.rules:80: forward.d: No such file or directory Makefile.rules:80: hash_func.d: No such file or directory Makefile.rules:80: ip_addr.d: No such file or directory Makefile.rules:80: main.d: No such file or directory Makefile.rules:80: md5.d: No such file or directory Makefile.rules:80: md5utils.d: No such file or directory Makefile.rules:80: modparam.d: No such file or directory Makefile.rules:80: msg_translator.d: No such file or directory Makefile.rules:80: pass_fd.d: No such file or directory Makefile.rules:80: proxy.d: No such file or directory Makefile.rules:80: qvalue.d: No such file or directory Makefile.rules:80: re.d: No such file or directory Makefile.rules:80: receive.d: No such file or directory Makefile.rules:80: resolve.d: No such file or directory Makefile.rules:80: route.d: No such file or directory Makefile.rules:80: route_struct.d: No such file or directory Makefile.rules:80: script_cb.d: No such file or directory Makefile.rules:80: socket_info.d: No such file or directory Makefile.rules:80: sr_module.d: No such file or directory Makefile.rules:80: stats.d: No such file or directory Makefile.rules:80: tcp_main.d: No such file or directory Makefile.rules:80: tcp_read.d: No such file or directory Makefile.rules:80: timer.d: No such file or directory Makefile.rules:80: tsend.d: No such file or directory Makefile.rules:80: udp_server.d: No such file or directory Makefile.rules:80: unixsock_server.d: No such file or directory Makefile.rules:80: usr_avp.d: No such file or directory Makefile.rules:80: mem/f_malloc.d: No such file or directory Makefile.rules:80: mem/mem.d: No such file or directory Makefile.rules:80: mem/memtest.d: No such file or directory Makefile.rules:80: mem/q_malloc.d: No such file or directory Makefile.rules:80: mem/shm_mem.d: No such file or directory Makefile.rules:80: mem/vq_malloc.d: No such file or directory Makefile.rules:80: parser/hf.d: No such file or directory Makefile.rules:80: parser/msg_parser.d: No such file or directory Makefile.rules:80: parser/parse_allow.d: No such file or directory Makefile.rules:80: parser/parse_content.d: No such file or directory Makefile.rules:80: parser/parse_cseq.d: No such file or directory Makefile.rules:80: parser/parse_disposition.d: No such file or directory Makefile.rules:80: parser/parse_diversion.d: No such file or directory Makefile.rules:80: parser/parse_event.d: No such file or directory Makefile.rules:80: parser/parse_expires.d: No such file or directory Makefile.rules:80: parser/parse_fline.d: No such file or directory Makefile.rules:80: parser/parse_from.d: No such file or directory Makefile.rules:80: parser/parse_hname2.d: No such file or directory Makefile.rules:80: parser/parse_hostport.d: No such file or directory Makefile.rules:80: parser/parse_methods.d: No such file or directory Makefile.rules:80: parser/parse_nameaddr.d: No such file or directory Makefile.rules:80: parser/parse_param.d: No such file or directory Makefile.rules:80: parser/parser_f.d: No such file or directory Makefile.rules:80: parser/parse_rpid.d: No such file or directory Makefile.rules:80: parser/parse_rr.d: No such file or directory Makefile.rules:80: parser/parse_sipifmatch.d: No such file or directory Makefile.rules:80: parser/parse_to.d: No such file or directory Makefile.rules:80: parser/parse_uri.d: No such file or directory Makefile.rules:80: parser/parse_via.d: No such file or directory Makefile.rules:80: parser/digest/digest.d: No such file or directory Makefile.rules:80: parser/digest/digest_parser.d: No such file or directory Makefile.rules:80: parser/digest/param_parser.d: No such file or directory Makefile.rules:80: parser/contact/contact.d: No such file or directory Makefile.rules:80: parser/contact/parse_contact.d: No such file or directory Makefile.rules:80: db/db.d: No such file or directory Makefile.rules:80: db/db_fifo.d: No such file or directory Makefile.rules:80: lex.yy.d: No such file or directory Does anyone know how to fix this problem ? Any help will be great ! Best regards, Jaroslaw Gawron

1 0

[Serusers] NAT example
by Kamran Ahmad 30 Aug '05

30 Aug '05

Hello i am following this link for ser-0.9.3 but Nated Clients are not working. Can any one please tell me what is the problem in my configuration. http://www.voip-info.org/tiki-index.php?page=SER+nat+support ---------------------------------------------------------------- #ser.cfg # $Id: ser.cfg,v 1.21 2003/06/04 13:47:36 jiri Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode debug=7 fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/ser_fifo" alias=mydomain.dyndns.org # ------------------ module loading ---------------------------------- loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.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" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (len_gt( max_len )) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol record_route(); # loose-route processing if (loose_route()) { t_relay(); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { save("location"); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; #inserted by klaus if (method=="INVITE") { record_route(); force_rtp_proxy(); /* set up reply processing */ t_on_reply("1"); }; # forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP if (!t_relay()) { sl_reply_error(); }; } #inserted by klaus # all incoming replies for t_onrepli-ed transactions enter here onreply_route[1] { if (status=~"[12][0-9][0-9]") force_rtp_proxy(); } __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

2 2

[Serusers] RTPProxy
by Anderson Alves de Albuquerque 29 Aug '05

29 Aug '05

Does someone have some doc (how-to, manual ...) about RTPProxy?

3 2

[Users] SER and SIP headers compact form
by Vincent Luba 29 Aug '05

29 Aug '05

Hi, Is there a way to force SER to use compact form when it sends SIP message? i.e. presence notification provided by "pa module" Regards, Luba Vincent

1 0

[Serusers] ser and database: using a view instead of a table for 'subscribers' table
by Arnaud Ligot 29 Aug '05

29 Aug '05

Hello, I already have a database with username, password (clear), firstname, lastname and I want to use ser with this database. I already have an administration interface for tis database so I want to keep it as is. I want to create a view for the 'subscribers' table based on my database. I'm looking for a good documentation about fields of 'subscribers' table. I also want to know if I can use ser without timestamp and encrypted fields of 'subscriber' table and if I have to create other tables in order to use ser only with auth_db. best regards, Arnaud Ligot

1 0

RE: [Serusers] hi,about mysql.so problem.
by Marc Khayat 29 Aug '05

29 Aug '05

- compile the mysql module, by doing: make modules=modules/mysql modules - copy mysql.so (found in modules/mysql) to /usr/lib/ser/modules/ Marc -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of viking Sent: Monday, August 29, 2005 12:12 PM To: serusers(a)lists.iptel.org Subject: [Serusers] hi,about mysql.so problem. hi,everyone, I have installed ser-0.9.3-src on red-hat 9.0，i wanna to use mysql module,the problem is how can get the /usr/lib/ser/modules/mysql.so if i compile them from the src? thanks a lot! 　　　　　　　　yours viking 　　　　　　　　tieban(a)sina.com 　　　　　　　　　　2005-08-29 _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

1 0

[Users] failure route problem
by Klaus Darilion 29 Aug '05

29 Aug '05

Hi! I have a problem with my failure route. In the failure route, I send the call to voicemail, except the call is CANCELED by the caller: failure_route[1] { if(t_check_status("487")) { xlog("L_INFO","caller cancelled call, no voicemail needed...\n"); exit; } route(4); # route to voicemail } The problem is, when a call is forked and one of the destinations is busy or has any other reasons to not accept the call. /-------- <-486 phone1 / caller ----------------- <-488 phone2 \ \-------- <-487 phone3 In the above case, phone1 and 2 immedately response with an error message. Then several seconds later, the caller hangs up and now phone3 sends 487. If I now check the status in the failure route it will be 486 (the lowest one). Also any other algorithm (the highest one, the first one, the last one) would be ambiguous. Thus, deciding if the call will be sent to voicemail or not is not possible using t_check_status. AFAIK there is no other method to check the status of the transaction in the failure route. IMO this is an issue which should be solved. I suggest setting the status explicitly to "487" if the transaction was cancelled by the caller, independent of the resonse status received from the phones. regards klaus

3 9

[Serusers] strange registration !?
by harry gaillac 29 Aug '05

29 Aug '05

Hello, Look at my ethereal file. When User agent send REGISTER message to ser this one try to register with itself so User agents receive authorization + "401 Unauthorized" !? What's wrong ? Regards Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

1 0

[Serusers] hi,about mysql.so problem.
by viking 29 Aug '05

29 Aug '05

hi,everyone, I have installed ser-0.9.3-src on red-hat 9.0，i wanna to use mysql module,the problem is how can get the /usr/lib/ser/modules/mysql.so if i compile them from the src? thanks a lot! 　　　　　　　　yours viking 　　　　　　　　tieban(a)sina.com 　　　　　　　　　　2005-08-29

1 0

[Users] #DEFINE equivalent?
by Kanakatti Mahesh Subramanya 29 Aug '05

29 Aug '05

It would be useful if we could DEFINE parameters in the .cfg file e.g. define SOME_IP=1.2.3.4 and somewhere later in the file record_route_preset("SOME_IP;nat=yes"); Speaking as someone who has to manage openser.cfg instances across multiple servers (not to mention test environments!), it would certainly make my life a heck of a lot easier.... cheers

3 4

[Users] 1 OpenSER & 2 Providers
by Ozan Blotter 29 Aug '05

29 Aug '05

Dear List, Assume that there are two different service providers and one OpenSER in middle of them. A client needs to use/connect OpenSER but also he/she has accounts in both providers and both supplied him/her DDIs (landline phone). Now again assume that client registers both of phones into OpenSER. But still he/she wants to receive calls from both providers, is it possible to receive calls from different providers to that client and/or how to i setup ? In current SER.CFG below, i only setup one prefix which is "0", but i can change it to any and/or add many prefixes, ex: 0049 (germany), 001 (usa), 0044 (uk), 0090 (turkey) to Provider 1 and 0033 (france), 0038 (italy) to Provider 2 and so on. debug=3 fork=yes log_stderror=no listen=212.XXX.104.XXX # This is OpenSER's Public IP port=5060 children=4 alias=212.XXX.104.XXX # This is OpenSER's Public IP dns=no rev_dns=no fifo="/tmp/openser_fifo" fifo_db_url="mysql://openser:openserrw@localhost/openser" loadmodule "/usr/local/lib/openser/modules/mysql.so" loadmodule "/usr/local/lib/openser/modules/sl.so" loadmodule "/usr/local/lib/openser/modules/tm.so" loadmodule "/usr/local/lib/openser/modules/rr.so" loadmodule "/usr/local/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/lib/openser/modules/usrloc.so" loadmodule "/usr/local/lib/openser/modules/registrar.so" loadmodule "/usr/local/lib/openser/modules/auth.so" loadmodule "/usr/local/lib/openser/modules/auth_db.so" loadmodule "/usr/local/lib/openser/modules/uri.so" loadmodule "/usr/local/lib/openser/modules/uri_db.so" loadmodule "/usr/local/lib/openser/modules/mediaproxy.so" loadmodule "/usr/local/lib/openser/modules/nathelper.so" loadmodule "/usr/local/lib/openser/modules/textops.so" loadmodule "/usr/local/lib/openser/modules/domain.so" loadmodule "/usr/local/lib/openser/modules/acc.so" modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("auth_db", "use_domain", 1) modparam("domain", "db_mode", 1) modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 180) modparam("mediaproxy","natping_interval", 30) modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/openser/sip-asymmetric-clients") modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/openser/rtp-asymmetric-clients") modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", 1) modparam("registrar", "default_expires", 60) modparam("registrar", "min_expires", 30) modparam("registrar", "nat_flag", 6) modparam("registrar", "use_domain", 1) modparam("rr", "enable_full_lr", 1) modparam("acc", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("acc", "failed_transactions", 1) modparam("acc", "log_level", 1) modparam("acc", "log_flag", 1) modparam("acc", "db_flag", 1) route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- 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; }; # ----------------------------------------------------------------- # Record Route Section and Acc section # ----------------------------------------------------------------- if (method=="INVITE" && client_nat_test("3")) { record_route_preset("212.XXX.104.XXX:5060;nat=yes"); # This is OpenSER's Public IP } else if (method!="REGISTER") { if!(uri=~"^sip:833[0-9]*@") { # I do not want to make accounting if clients' prefix is beginning with 833 record_route(); setflag(1); } }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { end_media_session(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- 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; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- 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 Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- if (!search("^Contact:\ +\*") && client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); }; sl_send_reply("100", "Trying"); if (!www_authorize("","subscriber")) { www_challenge("","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # CANCEL and INVITE Message Handler # ----------------------------------------------------------------- if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; lookup("aliases"); if (uri!=myself) { route(1); break; }; if (uri=~"^sip:0[0-9]*@*") { # If calling number is beginning with prefix 0, route it to PSTN rewritehost("195.XXX.122.XXX"); # This is PSTN Gateway's Public IP route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; if (method=="CANCEL") { route(1); break; }; if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); if (isflagset(6) || isflagset(7)) { use_media_proxy(); }; route(1); } onreply_route[1] { if ((isflagset(6) || isflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { if (!search("^Content-Length:\ +0")) { use_media_proxy(); }; }; if (client_nat_test("1")) { fix_nated_contact(); }; }

3 5

[Serusers] error starting ser
by chris 29 Aug '05

29 Aug '05

i tried the command ser -c -f ser.cfg and this is the result. bash-2.05# ser -c -f ser.cfg 0(2092) ERROR: load_module: could not open module </usr/local/lib/ser/modules/mysql.so>: ld.so.1: ser: fatal: libmysqlclient.so.10: open failed: No such file or directory 0(2092) parse error (15,13-14): failed to load module ERROR: bad config file (1 errors) bash-2.05# but, /usr/local/lib/ser/modules/mysql.so is existing, pls advise what could have gone wrong.

3 9

[Serusers] Radius packets not reaching the server
by Ashutosh Kumar 29 Aug '05

29 Aug '05

Hi, My ser-0.9.0 is sending the radius packets and is getting the packets reflected in ngrep output. But they are not reaching the radius server.What may be the reason? The bgrep output output is below: U a.b.c.d:34128 -> x.y.z.s:1812 .~...F..vn......./t...abcd@a.b.c.d....abcd....a.b.c.d.,.*430f5a3a34325aca96d 48bea7555c8ed6d0ab56b....sip:a.b.c.d....REGISTER."e035a87503a63d6d2ae8680c27 475cfd........abcd..E..p....... 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

2 1

[Serusers] MARK RODDY/LAKE is out of the office.
by mark.roddy＠lakecommunications.com 29 Aug '05

29 Aug '05

I will be out of the office starting 08/26/2005 and will not return until 09/01/2005. I will not be checking my mail. For Daidalos related issues please contact Stephen Butler. *********************************************************************************** This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify postmaster(a)lakecommunications.com This footnote also confirms that this email message has been scanned for the presence of computer viruses and other security threats. ***********************************************************************************

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[Serusers] SER + ASTERISK voicemail
by harry gaillac 28 Aug '05

28 Aug '05

Hello, I try set Ua---SER----Asterisk (voicemail/ARA) | Ua ser stable asterisk cvs head I read http://lists.iptel.org/pipermail/serusers/2005-February/015997.html to forward unavailable or busy sip agents to asterisk voicemail in failure route. How may I configure extensions.conf and ser.cfg ? I have been trying without success! Regards Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

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[Serusers] nathelper/rtpproxy.
by harry gaillac 27 Aug '05

27 Aug '05

Hello, Could you send me back more informations about nathelper/rtpproxy/MOH support ? Does nathelper/rtpproxy/MOH require a "music server" ? Regards Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

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[Users] $hdr[From] vs $hrd(From)
by Kanakatti Mahesh Subramanya 27 Aug '05

27 Aug '05

why $hdr[From] in avp_write(), and $hdr(From) in avp_printf? just haven't gotten around to it? or some valid syntactical reason? cheers

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[Users] $rt setting avp to the "To" header instead of the "Refer-To" header
by Kanakatti Mahesh Subramanya 27 Aug '05

27 Aug '05

Given the following packet REFER sip:9546236300@69.25.47.137:5090 SIP/2.0 Via: SIP/2.0/UDP 192.168.28.118;branch=z9hG4bK5eb7146d4B8F8EA4 From: <sip:mahesh.aptela@66.23.129.213>;tag=AB447651-36547B1E To: "operator.aptela" <sip:9546236300@69.25.47.137:5090>;tag=as6c1b062a Route: <sip:66.23.129.213;ftag=as6c1b062a;lr=on> CSeq: 2 REFER Call-ID: 688b089d4b10573715a49690228f1586(a)69.25.47.137 Contact: <sip:mahesh.aptela@192.168.28.118> User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.4.1 Refer-To: sip:7100@flareon.aptela.com:5060 Referred-By: <sip:mahesh.aptela@66.23.129.213> Max-Forwards: 70 Content-Length: 0 avp_printf() sets $rt to "sip:9546236300@69.25.47.137:5090" instead of "sip:7100@flareon.aptela.com:5060" trace follows When you run avp_write("$hdr[Refer-To]", "$savedRefer"); avp_print(); you get 2(5034) DEBUG: id=<65> 2(5034) DEBUG: val_str=<sip:7100@flareon.aptela.com:5060> When you run avp_printf("$savedRefer", "$rt"); avp_print(); you get 2(5034) DEBUG: id=<65> 2(5034) DEBUG: val_str=<sip:9546236300@69.25.47.137:5090> t

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[Serusers] NATHelper Issues
by Darren Nay 27 Aug '05

27 Aug '05

Hey All, Got a question for you all. I've had a nathelper implementation in place for months thinking that it was working, but recent events have made me question my configuration. It doesn't seem to be rewriting the contact header when the src_port and contact_port are different. Even though I am using fix_nated_contact() for REGISTER requests. Please see my details below. I am using SER v8.12 Here is an example ngrep capture for a REGISTER. Notice that the source port is 53492, and the contact port is 53616. -------------- U 66.110.232.221:53492 -> 208.253.183.40:5060 at 01:52:55 08/26/2005 REGISTER sip:proxy.ionosphere.net SIP/2.0 Via: SIP/2.0/UDP 66.110.232.221:53492;branch=z9hG4bK-7c0d1eaa From: spectrum wireless <sip:+13367908202@proxy.ionosphere.net>;tag=f8c505c5a8fc2e3o0 To: spectrum wireless <sip:+13367908202@proxy.ionosphere.net> Call-ID: e0bd4b00-2770688f(a)192.168.1.29 CSeq: 10 REGISTER Max-Forwards: 70 Contact: spectrum wireless <sip:+13367908202@66.110.232.221:53616>;expires=600 User-Agent: IONOSPHERE I-PHONE v1.0 Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura ------- U 208.253.183.40:5060 -> 66.110.232.221:53492 at 01:52:55 08/26/2005 SIP/2.0 200 OK Via: SIP/2.0/UDP 66.110.232.221:53492;branch=z9hG4bK-7c0d1eaa From: spectrum wireless <sip:+13367908202@proxy.ionosphere.net>;tag=f8c505c5a8fc2e3o0 To: spectrum wireless <sip:+13367908202@proxy.ionosphere.net>;tag=309667ea73a5d01d816917cdbd4fb3ac .6723 Call-ID: e0bd4b00-2770688f(a)192.168.1.29 CSeq: 10 REGISTER Contact: <sip:+13367908202@66.110.232.221:53616>;q=0.00;expires=600 Server: Sip EXpress router (0.8.12-tcp_nonb (i386/linux)) Content-Length: 0 Warning: 392 208.253.183.40:5060 "Noisy feedback tells: pid=31855 req_src_ip=66.110.232.221 req_src_port=53492 in_uri=sip:proxy.ionosphere.net out_uri=sip:proxy.ionosphere.net via_cnt==1" ------------- Here is a snipet for REGISTER requests from my ser.cfg -------- # Nathelper if (nat_uac_test("3")) { if (method == "REGISTER" || !search("^Record-Route:")) { log("LOG:Someone trying to use private IP, rewriting\n"); fix_nated_contact(); # Rewrite contact with source IP force_rport(); setflag(6); }; }; # Turn on SIP accounting setflag(3); if (!method=="REGISTER") record_route(); # loose-route processing if (loose_route()) { route(1); break; }; lookup("aliases"); if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; if (uri==myself) { if (method=="REGISTER") { if (!www_authorize("proxy.ionosphere.net", "subscriber")) { www_challenge("proxy.ionosphere.net", "0"); break; }; save("location") break; }; ## etc.................. }; -------- Then the registration is always recorded as the contact port. Here is a serctl ul output: <sip:+13367908202@66.110.232.221:53616>;q=0.00;expires=529 Does anyone have idea why this is happening? Is my SER.CFG incorrect? Thanks so much! Darren Nay VoIP Network Development Ionosphere, Inc. dnay(a)ionosphere.net

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[Serusers] recording REGISTER messages and errors
by Iqbal 27 Aug '05

27 Aug '05

Hi Is there a way, to do this, in acc table Iqbal

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