sr-users July 2005

sr-users@lists.kamailio.org

199 participants
544 discussions

[Serusers] Does VOVIDA B2BUA works with SER+MEDIAPROXY?
by Alberto Cruz 21 Jul '05

21 Jul '05

Hi does anybody is running SER+MEDIAPROXY and VOVIDA B2BUA? Please I need help to handle the CANCEL message. I've already try searching the http://www.archivum.info/serusers@lists.iptel.org/ and the http://lists.iptel.org/pipermail/serusers/ for any tip or configuration example but I didn't find anything. Regards Alberto Cruz

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[Serusers] consume_credentials() bug!?
by Rodrigo P. Telles 21 Jul '05

21 Jul '05

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi Folks, Recently I started to use consume_credentials() function in my SER 0.8.14 and some troubles appeared. One of some carriers that I send calls through started to reject calls routed from my SER box with: SIP/2.0 400 Transaction tuple incomplete (9/SL) The carrier said that my SER box is sending an invalid SIP Header. I did some dumps of calls from my SER box and I realise that there is an invalid SIP header as the carrier said, follow the dumps: x.x.x.x:5060 - my SER box y.y.y.y:5060 - carrier z.z.z.z:41593 - UA (GS BT100) x.x.x.x:5060 -> y.y.y.y:5060 INVITE sip:00000000000000@y.y.y.y:5060 SIP/2.0. Record-Route: <sip:0000000000000@x.x.x.x;ftag=8c7829c492ab97d9;lr=on>. Via: SIP/2.0/UDP x.x.x.x;branch=z9hG4bK151b.f401ac77.0. Via: SIP/2.0/UDP z.z.z.z:41593;branch=z9hG4bK0d356ccd0e9fab1d. From: "Client 1" <sip:1005@mysipdomain>;tag=8c7829c492ab97d9. To: <sip:000000000000@mysipdomain>. Contact: <sip:1005@z.z.z.z:41593>. Supported: replaces. sip:000000000000@mysipdomain", nonce="42cc468e4ddef88bad480d1a5dfbd8bf87c03f23", response="0a5825c4e4bf2bb43 e2a0ba850ba32b1".Call-ID: cff0854b39f5c97e(a)z.z.z.z. CSeq: 59598 INVITE. User-Agent: Grandstream BT100 1.0.5.22. Max-Forwards: 69. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE. Content-Type: application/sdp. Content-Length: 388 As you can see the line after "Supported: replaces." does not start with a "capital letter" as RFC3261 recommend. After remove "consume_credentials()" function the result is: x.x.x.x:5060 -> y.y.y.y:5060 INVITE sip:0000000000000@y.y.y.y:5060 SIP/2.0. Record-Route: <sip:000000000000@x.x.x.x;ftag=21cb576a18f9628d;lr=on>. Via: SIP/2.0/UDP x.x.x.x;branch=z9hG4bK41cb.cc6503a1.0. Via: SIP/2.0/UDP z.z.z.z:41593;branch=z9hG4bKa5afc4c7b13b4943. From: "Client 1" <sip:1005@mysipdomain>;tag=21cb576a18f9628d. To: <sip:000000000000@mysipdomain>. Contact: <sip:1005@z.z.z.z:41593>. Supported: replaces. Proxy-Authorization: DIGEST username="1005", realm="mysipdomain", algorithm=MD5, uri="sip:00000000000@mysip domain", nonce="42cc41d72f6d20a05439f0f1ae8c4159b1c4bc3c", response="c111d28d54df340dabd44865f3815966". Call-ID: a059b6a913ffa488(a)z.z.z.z. CSeq: 48239 INVITE. User-Agent: Grandstream BT100 1.0.5.22. Max-Forwards: 69. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE. Content-Type: application/sdp. Content-Length: 388 and the carrier accept the call as expected. So I think the "consume_credentials()" function does not RFC3261 compliance. Maybe its a BUG and an uri subst function works better. regards. - -- ============================================ Rodrigo P. Telles <telles(a)devel.it> TI Manager Devel-IT - http://www.devel.it IVOZ # 1029 +55 14 3324-1200 Bestcom Group ============================================ -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.2.4 (GNU/Linux) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQFC1AUGiLK8unYgEMQRAjTbAJ4qWeu8V1R7fMpMumR9UcUHiZ/1ugCfTv8Y mGmkZkn8w0+R1OhQbSTf4do= =+E+T -----END PGP SIGNATURE-----

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[Serusers] NAT
by Sebastian Kühner 21 Jul '05

21 Jul '05

Hi! Another problem with NAT... I hope I will solve it soon! My rtpproxy is running and ser can connect to the rtpproxy. I can make calls to users that are connected to the ser server behind a NAT (Restricted Cone NAT and symmetrical NAT). That's working fine!!! Thanks for your help!!! But I have the problem that if I want to call to a Cisco PSTN-Gateway, that I can establish the call... but I don't hear anything. The callee is hearing me. What is the difference between an internal user and the Gatway?? The routing for Gateway and Users is almost equal (the only difference is the "P-hint: GATEWAY"). And: I'm wondering why the communication is OK with a symmetrical NAT (Linksys) and WITHOUT rtpproxy. If I turn on the rtpproxy (with symetrical NAT) the voice stream of the Gateway isn't arriving anymore. Any ideas? Thanks! Sebastian

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RE: [Serusers] Problem authorizing with radius
by Naresh Parmar 21 Jul '05

21 Jul '05

Hi Ricardo, Tried it. It still gives me the same error. Please let me know the version of the radius server you are using.?? Also can you please let me know wht did u do to make the accounting work...?? Best Regards, Naresh Ricardo Martinez <rmartinez(a)redvoiss.net> wrote: Hello Naresh. I guess there is an error in the way you call the authorization for the INVITE. As far as i know for the REGISTER message (authentication) you need the statement : radius_www_authorize But for the INVITE you need to call "radius_proxy_authorize". This is what i have in my ser.cfg if (method=="INVITE") { if (!radius_proxy_authorize("")) { proxy_challenge("","1"); break; }; }; maybe you can try this and tell me how it works. Good luck Ricardo Martinez.- -----Mensaje original----- De: Naresh Parmar [mailto:naresh_parmar14@yahoo.com] Enviado el: Miércoles, 20 de Julio de 2005 12:10 Para: Ricardo Martinez; serusers(a)lists.iptel.org Asunto: RE: [Serusers] Problem authorizing with radius Hi Ricardo, We are using freeradius server 0.9.1 and SER 0.9.3. The version of radius client is radiusclient-ng-0.5.1. The users file in the radius server looks like as below: test(a)sip2.zone Auth-Type := Digest, User-Password == "cisco1234" Reply-Message = "Authenticated", Sip-Rpid = "1970" test(a)sip2.zone Auth-Type := Accept Reply-Message = "Authorized", Sip-Group == "ld" The radius authentication and authorization parts in the ser.cfg file are given below: if (uri=~"^sip:9[0-9]*@") { if (method=="INVITE"){ if (!radius_www_authorize("")) { www_challenge("", "1"); break; }else{ if (radius_is_user_in("Credentials", "ld")){ forward(192.168.2.101,5060); break; }else{ break; }; }; }; }; And finally the error is as below: Invalid operator for item Suffix: reverting to '==' modcall[authorize]: module "preprocess" returns ok modcall[authorize]: module "chap" returns noop rlm_eap: No EAP-Message, not doing EAP modcall[authorize]: module "eap" returns noop rlm_digest: Converting Digest-Attributes to something sane... Digest-User-Name = "test" Digest-Realm = "sip2.zone" Digest-Nonce = "42de75b2e9e39194a286e8ccd284646ffa14bcc2" Digest-URI = "sip:94161000@sip2.zone" Digest-Method = "INVITE" Digest-QOP = "auth" Digest-Nonce-Count = "0000000a" Digest-CNonce = "753F926DB8F5415D8D56EE7816410E33" rlm_digest: Adding Auth-Type = DIGEST modcall[authorize]: module "digest" returns ok rlm_realm: Looking up realm "sip2.zone" for User-Name = "test(a)sip2.zone" rlm_realm: No such realm "sip2.zone" modcall[authorize]: module "suffix" returns noop users: Matched entry test(a)sip2.zone at line 226 modcall[authorize]: module "files" returns ok modcall[authorize]: module "mschap" returns noop modcall: group authorize returns ok rad_check_password: Found Auth-Type Digest auth: type "digest" modcall: entering group authenticate A1 = test:sip2.zone:cisco1234 A2 = INVITE:sip:94161000@sip2.zone KD = 53d3b82970bada131a062103f553b8b8:42de75b2e9e39194a286e8ccd284646ffa14bcc2:0000000a:753F926DB8F5415D8D56EE7816410E33:auth:18227b358ffe96049a3745eeb449fae2 modcall[authenticate]: module "digest" returns ok modcall: group authenticate returns ok radius_xlat: 'Authenticated' Login OK: [test(a)sip2.zone/<no User-Password attribute>] (from client proxy port 5060) Sending Access-Accept of id 203 to 192.168.2.1:32831 Reply-Message = "Authenticated" Sip-Rpid = "1970" Finished request 6 Going to the next request --- Walking the entire request list --- Waking up in 6 seconds... rad_recv: Access-Request packet from host 192.168.2.1:32831, id=204, length=53 User-Name = "test" Sip-Group = "ld" Service-Type = Group-Check NAS-IP-Address = 192.168.2.1 NAS-Port = 0 modcall: entering group authorize Invalid operator for item Suffix: reverting to '==' Invalid operator for item Suffix: reverting to '==' Invalid operator for item Suffix: reverting to '==' modcall[authorize]: module "preprocess" returns ok modcall[authorize]: module "chap" returns noop rlm_eap: No EAP-Message, not doing EAP modcall[authorize]: module "eap" returns noop modcall[authorize]: module "digest" returns noop rlm_realm: No '@' in User-Name = "test", looking up realm NULL rlm_realm: No such realm "NULL" modcall[authorize]: module "suffix" returns noop modcall[authorize]: module "files" returns notfound modcall[authorize]: module "mschap" returns noop modcall: group authorize returns ok auth: No authenticate method (Auth-Type) configuration found for the request: Rejecting the user auth: Failed to validate the user. Login incorrect: [test/<no User-Password attribute>] (from client proxy port 0) Delaying request 7 for 1 seconds Finished request 7 Going to the next request Waking up in 6 seconds... As you can see from the above configuration, the authentication works perfect, its only in the authorization where it fails. Also can you please let me know about the accounting configuration?? Thanks a lot.. Naresh Ricardo Martinez <rmartinez(a)redvoiss.net> wrote: Hello Naresh I have authentication, authorization and accounting (AAA) through radius working fine. What radius server are you using?, can you send us more information about the configuration? Cheers, Ricardo.- -----Mensaje original----- De: Naresh Parmar [mailto:naresh_parmar14@yahoo.com] Enviado el: Miércoles, 20 de Julio de 2005 10:37 Para: serusers(a)lists.iptel.org Asunto: [Serusers] Problem authorizing with radius hi friends, I am having problems while authorizing with the radius server. I am using the same configuration as mentioned in the radius-howto. Authentication works perfect as I am able to authenticate using the radius server. However while authorizing against the radius server to make a call I get the following error: auth: No authenticate method (Auth-Type) configuration found for the user request: Rejecting the user auth: Failed to validate the user. Delaying request 2 for 1 seconds Finished request 2 When I authorize against the mysql database, it works fine. Any clue??? Best Regards, Naresh __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com --------------------------------- Start your day with Yahoo! - make it your home page

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[Users] When does an AVP get destroyed? What if you delete an non-existent AVP?
by Kanakatti M. Subramanya 21 Jul '05

21 Jul '05

Does the AVP get deleted at the end of a transaction? At the end of a specific dialog? If the answer is "transaction" (and I think it is), what exactly constitutes a transaction? e.g. when an INVITE is received,is the transaction completed when the corresponding "200 OK" is processed (or CANCEL)? Also, if I have 'avp_delete("$destination")', and the AVP '$destination' has never been set, is this "bad"? or a no-op? The above two are related. I am using AVPs to move information around in my script, and am wondering how much cleanup is necessary (i.e., delete every avp i populate, etc., etc.) cheers

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[Serusers] Problems with Vovida B2BUA and SER
by Alberto Cruz 21 Jul '05

21 Jul '05

Hi does anybody is using VOVIDA B2BUA with SER? I have the following NatedUA --> SER+MEDIAPROXY <----> B2BUA | | PSTN GW I want to know why when the UA sends the CANCEL to SER and SER forwards the CANCEL to B2BUA, The B2BUA only sends a 200 OK to SER and "487 Request Terminated" to the UA but the CALL from the SER to MEDIAPROXY is still in progress because it doesn't receive a CANCEL from the B2BUA. Does anybody have seen this behavior before? I'm attaching my ngrep trace I had to remove some lines because the message body limited size by the mailing list. Regards Alberto Cruz interface: any filter: ip and ( port 5060 or port 5065 ) # U 2005/07/20 21:05:37.876373 200.67.33.247:62295 -> 200.94.75.4:5060 INVITE sip:018183324166@sertest.conecta-te.com SIP/2.0. Via: SIP/2.0/UDP 172.31.253.127:5060;branch=z9hG4bK-492cab15. From: 8183651723 <sip:8183651723@sertest.conecta-te.com>;tag=a3d69093f2a96865o0. To: <sip:018183324166@sertest.conecta-te.com>. Call-ID: 46663eab-92c3d12d(a)172.31.253.127. CSeq: 101 INVITE. Max-Forwards: 70. Contact: 8183651723 <sip:8183651723@172.31.253.127:5060>. Expires: 240. User-Agent: Linksys/PAP2-2.0.12(LS). Content-Length: 424. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Supported: x-sipura. Content-Type: application/sdp. . v=0. o=- 83293 83293 IN IP4 172.31.253.127. s=-. c=IN IP4 172.31.253.127. t=0 0. m=audio 16428 RTP/AVP 0 2 4 8 18 96 97 98 100 101. a=rtpmap:0 PCMU/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:4 G723/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729a/8000. a=rtpmap:96 G726-40/8000. a=rtpmap:97 G726-24/8000. a=rtpmap:98 G726-16/8000. a=rtpmap:100 NSE/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:30. a=sendrecv. # U 2005/07/20 21:05:37.879177 200.94.75.4:5060 -> 200.94.75.14:5065 INVITE sip:018183324166@sertest.conecta-te.com SIP/2.0. Record-Route: <sip:018183324166@200.94.75.4:5060;nat=yes;ftag=a3d69093f2a96865o0;lr=on>. Via: SIP/2.0/UDP 200.94.75.4;branch=0. Via: SIP/2.0/UDP 172.31.253.127:5060;rport=62295;received=200.67.33.247;branch=z9hG4bK-492cab15. From: 8183651723 <sip:8183651723@sertest.conecta-te.com>;tag=a3d69093f2a96865o0. To: <sip:018183324166@sertest.conecta-te.com>. Call-ID: 46663eab-92c3d12d(a)172.31.253.127. CSeq: 101 INVITE. Max-Forwards: 16. Contact: 8183651723 <sip:8183651723@200.67.33.247:62295>. Expires: 240. User-Agent: Linksys/PAP2-2.0.12(LS). Content-Length: 424. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Supported: x-sipura. Content-Type: application/sdp. P-hint: PASANDO POR EL B2BUA. . v=0. o=- 83293 83293 IN IP4 172.31.253.127. s=-. c=IN IP4 172.31.253.127. t=0 0. m=audio 16428 RTP/AVP 0 2 4 8 18 96 97 98 100 101. a=rtpmap:0 PCMU/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:4 G723/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729a/8000. a=rtpmap:96 G726-40/8000. a=rtpmap:97 G726-24/8000. a=rtpmap:98 G726-16/8000. a=rtpmap:100 NSE/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:30. a=sendrecv. # U 2005/07/20 21:07:12.707957 200.94.75.14:5065 -> 200.67.33.247:62295 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 172.31.253.127:5060;rport=62295;received=200.67.33.247;branch=z9hG4bK-492cab15. Via: SIP/2.0/UDP 200.94.75.4:5060;branch=0. To: <sip:018183324166@sertest.conecta-te.com>. From: 8183651723 <sip:8183651723@sertest.conecta-te.com>;tag=a3d69093f2a96865o0. Call-ID: 46663eab-92c3d12d(a)172.31.253.127. CSeq: 101 INVITE. Content-Length: 0. . # U 2005/07/20 21:05:37.892397 200.94.75.14:5065 -> 200.94.75.4:5060 INVITE sip:018183324166@200.94.75.4:5060 SIP/2.0. Via: SIP/2.0/UDP 200.94.75.14:5065. To: <sip:018183324166@sertest.conecta-te.com>. From: 8183651723 <sip:8183651723@sertest.conecta-te.com>;tag=a3d69093f2a96865o0. Call-ID: 776f3ad42100670bbdf82c558882334a(a)200.94.75.14. CSeq: 101 INVITE. Max-Forwards: 16. Expires: 240. Contact: <sip:8183651723@200.94.75.14:5065>. Allow: ACK,BYE,CANCEL,INFO,INVITE,NOTIFY,OPTIONS,REFER. Supported: x-sipura. User-Agent: Linksys/PAP2-2.0.12(LS). Content-Type: application/sdp. Content-Length: 424. P-hint: PASANDO POR EL B2BUA. . v=0. o=- 83293 83293 IN IP4 172.31.253.127. s=-. c=IN IP4 172.31.253.127. t=0 0. m=audio 16428 RTP/AVP 0 2 4 8 18 96 97 98 100 101. a=rtpmap:0 PCMU/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:4 G723/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729a/8000. a=rtpmap:96 G726-40/8000. a=rtpmap:97 G726-24/8000. a=rtpmap:98 G726-16/8000. a=rtpmap:100 NSE/8000. a=rtpmap:101 telephone-event/8000. a=sendrecv. a=fmtp:101 0-15. a=ptime:30. # U 2005/07/20 21:05:37.895586 200.94.75.4:5060 -> 200.94.75.14:5065 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 200.94.75.14:5065;rport=5065. To: <sip:018183324166@sertest.conecta-te.com>. From: 8183651723 <sip:8183651723@sertest.conecta-te.com>;tag=a3d69093f2a96865o0. Call-ID: 776f3ad42100670bbdf82c558882334a(a)200.94.75.14. CSeq: 101 INVITE. Server: Sip EXpress router (0.9.3 (i386/linux)). Content-Length: 0. Warning: 392 200.94.75.4:5060 "Noisy feedback tells: pid=3518 req_src_ip=200.94.75.14 req_src_port=5065 in_uri=sip:018183324166@200.94.75.4:5060 out_uri=sip:018183324166@200.94.75.1:5060 via_cnt==1". . # U 2005/07/20 21:05:37.895990 200.94.75.4:5060 -> 200.94.75.1:5060 INVITE sip:018183324166@200.94.75.1:5060 SIP/2.0. Record-Route: <sip:200.94.75.4;ftag=a3d69093f2a96865o0;lr=on>. Via: SIP/2.0/UDP 200.94.75.4;branch=z9hG4bK6dc7.8fda76d7.0. Via: SIP/2.0/UDP 200.94.75.14:5065;rport=5065. To: <sip:018183324166@sertest.conecta-te.com>. From: 8183651723 <sip:8183651723@sertest.conecta-te.com>;tag=a3d69093f2a96865o0. Call-ID: 776f3ad42100670bbdf82c558882334a(a)200.94.75.14. CSeq: 101 INVITE. Max-Forwards: 15. Expires: 240. Contact: <sip:8183651723@200.94.75.14:5065>. Allow: ACK,BYE,CANCEL,INFO,INVITE,NOTIFY,OPTIONS,REFER. Supported: x-sipura. User-Agent: Linksys/PAP2-2.0.12(LS). Content-Type: application/sdp. Content-Length: 424. P-hint: PASANDO POR EL B2BUA. . v=0. o=- 83293 83293 IN IP4 172.31.253.127. s=-. c=IN IP4 172.31.253.127. t=0 0. m=audio 16428 RTP/AVP 0 2 4 8 18 96 97 98 100 101. a=rtpmap:0 PCMU/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:4 G723/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729a/8000. a=rtpmap:96 G726-40/8000. a=rtpmap:97 G726-24/8000. a=rtpmap:98 G726-16/8000. a=rtpmap:100 NSE/8000. a=rtpmap:101 telephone-event/8000. a=sendrecv. a=fmtp:101 0-15. a=ptime:30. # U 2005/07/20 21:05:37.915106 200.94.75.1:5060 -> 200.94.75.4:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 200.94.75.4;branch=z9hG4bK6dc7.8fda76d7.0,SIP/2.0/UDP 200.94.75.14:5065;rport=5065. From: 8183651723 <sip:8183651723@sertest.conecta-te.com>;tag=a3d69093f2a96865o0. To: <sip:018183324166@sertest.conecta-te.com>;tag=3D52A958-25AF. Date: Thu, 21 Jul 2005 02:05:50 GMT. Call-ID: 776f3ad42100670bbdf82c558882334a(a)200.94.75.14. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 101 INVITE. Allow-Events: telephone-event. Content-Length: 0. . # U 2005/07/20 21:05:42.833409 200.94.75.1:5060 -> 200.94.75.4:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 200.94.75.4;branch=z9hG4bK6dc7.8fda76d7.0,SIP/2.0/UDP 200.94.75.14:5065;rport=5065. From: 8183651723 <sip:8183651723@sertest.conecta-te.com>;tag=a3d69093f2a96865o0. To: <sip:018183324166@sertest.conecta-te.com>;tag=3D52A958-25AF. Date: Thu, 21 Jul 2005 02:05:50 GMT. Call-ID: 776f3ad42100670bbdf82c558882334a(a)200.94.75.14. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 101 INVITE. Allow-Events: telephone-event. Content-Type: application/sdp. Content-Disposition: session;handling=required. Content-Length: 188. . v=0. o=CiscoSystemsSIP-GW-UserAgent 1429 77 IN IP4 200.94.75.1. s=SIP Call. c=IN IP4 200.94.27.30. t=0 0. m=audio 16470 RTP/AVP 0. c=IN IP4 200.94.27.30. a=rtpmap:0 PCMU/8000. a=ptime:20. # U 2005/07/20 21:05:42.835158 200.94.75.4:5060 -> 200.94.75.14:5065 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 200.94.75.14:5065;rport=5065. From: 8183651723 <sip:8183651723@sertest.conecta-te.com>;tag=a3d69093f2a96865o0. To: <sip:018183324166@sertest.conecta-te.com>;tag=3D52A958-25AF. Date: Thu, 21 Jul 2005 02:05:50 GMT. Call-ID: 776f3ad42100670bbdf82c558882334a(a)200.94.75.14. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 101 INVITE. Allow-Events: telephone-event. Content-Type: application/sdp. Content-Disposition: session;handling=required. Content-Length: 188. . v=0. o=CiscoSystemsSIP-GW-UserAgent 1429 77 IN IP4 200.94.75.1. s=SIP Call. c=IN IP4 200.94.27.30. t=0 0. m=audio 16470 RTP/AVP 0. c=IN IP4 200.94.27.30. a=rtpmap:0 PCMU/8000. a=ptime:20. # U 2005/07/20 21:07:17.664495 200.94.75.14:5065 -> 200.67.33.247:62295 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 172.31.253.127:5060;rport=62295;received=200.67.33.247;branch=z9hG4bK-492cab15. Via: SIP/2.0/UDP 200.94.75.4:5060;branch=0. To: <sip:018183324166@sertest.conecta-te.com>;tag=02265f8b. From: 8183651723 <sip:8183651723@sertest.conecta-te.com>;tag=a3d69093f2a96865o0. Call-ID: 46663eab-92c3d12d(a)172.31.253.127. CSeq: 101 INVITE. Record-Route: <sip:018183324166@200.94.75.4:5060>. Content-Type: application/sdp. Content-Length: 188. . v=0. o=CiscoSystemsSIP-GW-UserAgent 1429 77 IN IP4 200.94.75.1. s=SIP Call. c=IN IP4 200.94.27.30. t=0 0. m=audio 16470 RTP/AVP 0. c=IN IP4 200.94.27.30. a=rtpmap:0 PCMU/8000. a=ptime:20. # U 2005/07/20 21:05:53.851125 200.67.33.247:62295 -> 200.94.75.4:5060 CANCEL sip:018183324166@sertest.conecta-te.com SIP/2.0. Via: SIP/2.0/UDP 172.31.253.127:5060;branch=z9hG4bK-492cab15. From: 8183651723 <sip:8183651723@sertest.conecta-te.com>;tag=a3d69093f2a96865o0. To: <sip:018183324166@sertest.conecta-te.com>. Call-ID: 46663eab-92c3d12d(a)172.31.253.127. CSeq: 101 CANCEL. Max-Forwards: 70. User-Agent: Linksys/PAP2-2.0.12(LS). Content-Length: 0. . # U 2005/07/20 21:05:53.854444 200.94.75.4:5060 -> 200.94.75.14:5065 CANCEL sip:018183324166@sertest.conecta-te.com SIP/2.0. Record-Route: <sip:200.94.75.4;ftag=a3d69093f2a96865o0;lr=on>. Via: SIP/2.0/UDP 200.94.75.4;branch=0. Via: SIP/2.0/UDP 172.31.253.127:5060;rport=62295;received=200.67.33.247;branch=z9hG4bK-492cab15. From: 8183651723 <sip:8183651723@sertest.conecta-te.com>;tag=a3d69093f2a96865o0. To: <sip:018183324166@sertest.conecta-te.com>. Call-ID: 46663eab-92c3d12d(a)172.31.253.127. CSeq: 101 CANCEL. Max-Forwards: 16. User-Agent: Linksys/PAP2-2.0.12(LS). Content-Length: 0. P-hint: finishing call. P-hint: PASANDO POR EL B2BUA. . # U 2005/07/20 21:05:53.856779 200.94.75.14:5065 -> 200.94.75.4:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 200.94.75.4:5060;branch=0. Via: SIP/2.0/UDP 172.31.253.127:5060;rport=62295;received=200.67.33.247;branch=z9hG4bK-492cab15. To: <sip:018183324166@sertest.conecta-te.com>;tag=95ccb113. From: 8183651723 <sip:8183651723@sertest.conecta-te.com>;tag=a3d69093f2a96865o0. Call-ID: 46663eab-92c3d12d(a)172.31.253.127. CSeq: 101 CANCEL. Record-Route: <sip:200.94.75.4;ftag=a3d69093f2a96865o0;lr=on>. Content-Length: 0. . # U 2005/07/20 21:07:28.685331 200.94.75.14:5065 -> 200.67.33.247:62295 SIP/2.0 487 Request Terminated. Via: SIP/2.0/UDP 172.31.253.127:5060;rport=62295;received=200.67.33.247;branch=z9hG4bK-492cab15. Via: SIP/2.0/UDP 200.94.75.4:5060;branch=0. To: <sip:018183324166@sertest.conecta-te.com>;tag=95ccb113. From: 8183651723 <sip:8183651723@sertest.conecta-te.com>;tag=a3d69093f2a96865o0. Call-ID: 46663eab-92c3d12d(a)172.31.253.127. CSeq: 101 INVITE. Content-Length: 0. . # U 2005/07/20 21:05:53.858422 200.94.75.4:5060 -> 200.67.33.247:62295 SIP/2.0 200 OK. Via: SIP/2.0/UDP 172.31.253.127:5060;rport=62295;received=200.67.33.247;branch=z9hG4bK-492cab15. To: <sip:018183324166@sertest.conecta-te.com>;tag=95ccb113. From: 8183651723 <sip:8183651723@sertest.conecta-te.com>;tag=a3d69093f2a96865o0. Call-ID: 46663eab-92c3d12d(a)172.31.253.127. CSeq: 101 CANCEL. Record-Route: <sip:200.94.75.4;ftag=a3d69093f2a96865o0;lr=on>. Content-Length: 0. . # U 2005/07/20 21:05:53.925902 200.67.33.247:62295 -> 200.94.75.4:5060 ACK sip:018183324166@sertest.conecta-te.com SIP/2.0. Via: SIP/2.0/UDP 172.31.253.127:5060;branch=z9hG4bK-492cab15. From: 8183651723 <sip:8183651723@sertest.conecta-te.com>;tag=a3d69093f2a96865o0. To: <sip:018183324166@sertest.conecta-te.com>;tag=95ccb113. Call-ID: 46663eab-92c3d12d(a)172.31.253.127. CSeq: 101 ACK. Max-Forwards: 70. Contact: 8183651723 <sip:8183651723@172.31.253.127:5060>. User-Agent: Linksys/PAP2-2.0.12(LS). Content-Length: 0. . # U 2005/07/20 21:05:53.929617 200.94.75.4:5060 -> 200.94.75.4:5060 ACK sip:018183324166@sertest.conecta-te.com SIP/2.0. Record-Route: <sip:200.94.75.4;ftag=a3d69093f2a96865o0;lr=on>. Via: SIP/2.0/UDP 200.94.75.4;branch=0. Via: SIP/2.0/UDP 172.31.253.127:5060;received=200.67.33.247;branch=z9hG4bK-492cab15. From: 8183651723 <sip:8183651723@sertest.conecta-te.com>;tag=a3d69093f2a96865o0. To: <sip:018183324166@sertest.conecta-te.com>;tag=95ccb113. Call-ID: 46663eab-92c3d12d(a)172.31.253.127. CSeq: 101 ACK. Max-Forwards: 16. Contact: 8183651723 <sip:8183651723@172.31.253.127:5060>. User-Agent: Linksys/PAP2-2.0.12(LS). Content-Length: 0. . # U 2005/07/20 21:05:53.933959 200.94.75.4:5060 -> 200.94.75.4:5060 ACK sip:018183324166@sertest.conecta-te.com SIP/2.0. Record-Route: <sip:200.94.75.4;ftag=a3d69093f2a96865o0;lr=on>. Record-Route: <sip:200.94.75.4;ftag=a3d69093f2a96865o0;lr=on>. Via: SIP/2.0/UDP 200.94.75.4;branch=0. Via: SIP/2.0/UDP 200.94.75.4;branch=0. Via: SIP/2.0/UDP 172.31.253.127:5060;received=200.67.33.247;branch=z9hG4bK-492cab15. From: 8183651723 <sip:8183651723@sertest.conecta-te.com>;tag=a3d69093f2a96865o0. To: <sip:018183324166@sertest.conecta-te.com>;tag=95ccb113. Call-ID: 46663eab-92c3d12d(a)172.31.253.127. CSeq: 101 ACK. Max-Forwards: 15. Contact: 8183651723 <sip:8183651723@172.31.253.127:5060>. User-Agent: Linksys/PAP2-2.0.12(LS). Content-Length: 0. . # U 2005/07/20 21:05:53.938833 200.94.75.4:5060 -> 200.94.75.4:5060 ACK sip:018183324166@sertest.conecta-te.com SIP/2.0. Record-Route: <sip:200.94.75.4;ftag=a3d69093f2a96865o0;lr=on>. Record-Route: <sip:200.94.75.4;ftag=a3d69093f2a96865o0;lr=on>. Record-Route: <sip:200.94.75.4;ftag=a3d69093f2a96865o0;lr=on>. Via: SIP/2.0/UDP 200.94.75.4;branch=0. Via: SIP/2.0/UDP 200.94.75.4;branch=0. Via: SIP/2.0/UDP 200.94.75.4;branch=0. Via: SIP/2.0/UDP 172.31.253.127:5060;received=200.67.33.247;branch=z9hG4bK-492cab15. From: 8183651723 <sip:8183651723@sertest.conecta-te.com>;tag=a3d69093f2a96865o0. To: <sip:018183324166@sertest.conecta-te.com>;tag=95ccb113. Call-ID: 46663eab-92c3d12d(a)172.31.253.127. CSeq: 101 ACK. Max-Forwards: 14. Contact: 8183651723 <sip:8183651723@172.31.253.127:5060>. User-Agent: Linksys/PAP2-2.0.12(LS). Content-Length: 0. . . . # 14 times more ACK msg until completes a total of 17 ACK messages adding one new Via with branch=0 each time # U 2005/07/20 21:06:21.427416 200.94.75.1:5060 -> 200.94.75.4:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 200.94.75.4;branch=z9hG4bK6dc7.8fda76d7.0,SIP/2.0/UDP 200.94.75.14:5065;rport=5065. From: 8183651723 <sip:8183651723@sertest.conecta-te.com>;tag=a3d69093f2a96865o0. To: <sip:018183324166@sertest.conecta-te.com>;tag=3D52A958-25AF. Date: Thu, 21 Jul 2005 02:05:50 GMT. Call-ID: 776f3ad42100670bbdf82c558882334a(a)200.94.75.14. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 101 INVITE. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO. Allow-Events: telephone-event. Contact: <sip:018183324166@200.94.75.1:5060>. Record-Route: <sip:200.94.75.4;ftag=a3d69093f2a96865o0;lr=on>. Content-Type: application/sdp. Content-Length: 188. . v=0. o=CiscoSystemsSIP-GW-UserAgent 1429 77 IN IP4 200.94.75.1. s=SIP Call. c=IN IP4 200.94.27.30. t=0 0. m=audio 16470 RTP/AVP 0. c=IN IP4 200.94.27.30. a=rtpmap:0 PCMU/8000. a=ptime:20. # U 2005/07/20 21:06:21.429249 200.94.75.4:5060 -> 200.94.75.14:5065 SIP/2.0 200 OK. Via: SIP/2.0/UDP 200.94.75.14:5065;rport=5065. From: 8183651723 <sip:8183651723@sertest.conecta-te.com>;tag=a3d69093f2a96865o0. To: <sip:018183324166@sertest.conecta-te.com>;tag=3D52A958-25AF. Date: Thu, 21 Jul 2005 02:05:50 GMT. Call-ID: 776f3ad42100670bbdf82c558882334a(a)200.94.75.14. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 101 INVITE. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO. Allow-Events: telephone-event. Contact: <sip:018183324166@200.94.75.1:5060>. Record-Route: <sip:200.94.75.4;ftag=a3d69093f2a96865o0;lr=on>. Content-Type: application/sdp. Content-Length: 188. . v=0. o=CiscoSystemsSIP-GW-UserAgent 1429 77 IN IP4 200.94.75.1. s=SIP Call. c=IN IP4 200.94.27.30. t=0 0. m=audio 16470 RTP/AVP 0. c=IN IP4 200.94.27.30. a=rtpmap:0 PCMU/8000. a=ptime:20. # U 2005/07/20 21:06:21.928003 200.94.75.1:5060 -> 200.94.75.4:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 200.94.75.4;branch=z9hG4bK6dc7.8fda76d7.0,SIP/2.0/UDP 200.94.75.14:5065;rport=5065. From: 8183651723 <sip:8183651723@sertest.conecta-te.com>;tag=a3d69093f2a96865o0. To: <sip:018183324166@sertest.conecta-te.com>;tag=3D52A958-25AF. Date: Thu, 21 Jul 2005 02:05:50 GMT. Call-ID: 776f3ad42100670bbdf82c558882334a(a)200.94.75.14. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 101 INVITE. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO. Allow-Events: telephone-event. Contact: <sip:018183324166@200.94.75.1:5060>. Record-Route: <sip:200.94.75.4;ftag=a3d69093f2a96865o0;lr=on>. Content-Type: application/sdp. Content-Length: 188. . v=0. o=CiscoSystemsSIP-GW-UserAgent 1429 77 IN IP4 200.94.75.1. s=SIP Call. c=IN IP4 200.94.27.30. t=0 0. m=audio 16470 RTP/AVP 0. c=IN IP4 200.94.27.30. a=rtpmap:0 PCMU/8000. a=ptime:20. # U 2005/07/20 21:06:21.929751 200.94.75.4:5060 -> 200.94.75.14:5065 SIP/2.0 200 OK. Via: SIP/2.0/UDP 200.94.75.14:5065;rport=5065. From: 8183651723 <sip:8183651723@sertest.conecta-te.com>;tag=a3d69093f2a96865o0. To: <sip:018183324166@sertest.conecta-te.com>;tag=3D52A958-25AF. Date: Thu, 21 Jul 2005 02:05:50 GMT. Call-ID: 776f3ad42100670bbdf82c558882334a(a)200.94.75.14. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 101 INVITE. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO. Allow-Events: telephone-event. Contact: <sip:018183324166@200.94.75.1:5060>. Record-Route: <sip:200.94.75.4;ftag=a3d69093f2a96865o0;lr=on>. Content-Type: application/sdp. Content-Length: 188. . v=0. o=CiscoSystemsSIP-GW-UserAgent 1429 77 IN IP4 200.94.75.1. s=SIP Call. c=IN IP4 200.94.27.30. t=0 0. m=audio 16470 RTP/AVP 0. c=IN IP4 200.94.27.30. a=rtpmap:0 PCMU/8000. a=ptime:20. # U 2005/07/20 21:06:22.927716 200.94.75.1:5060 -> 200.94.75.4:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 200.94.75.4;branch=z9hG4bK6dc7.8fda76d7.0,SIP/2.0/UDP 200.94.75.14:5065;rport=5065. From: 8183651723 <sip:8183651723@sertest.conecta-te.com>;tag=a3d69093f2a96865o0. To: <sip:018183324166@sertest.conecta-te.com>;tag=3D52A958-25AF. Date: Thu, 21 Jul 2005 02:05:50 GMT. Call-ID: 776f3ad42100670bbdf82c558882334a(a)200.94.75.14. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 101 INVITE. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO. Allow-Events: telephone-event. Contact: <sip:018183324166@200.94.75.1:5060>. Record-Route: <sip:200.94.75.4;ftag=a3d69093f2a96865o0;lr=on>. Content-Type: application/sdp. Content-Length: 188. . v=0. o=CiscoSystemsSIP-GW-UserAgent 1429 77 IN IP4 200.94.75.1. s=SIP Call. c=IN IP4 200.94.27.30. t=0 0. m=audio 16470 RTP/AVP 0. c=IN IP4 200.94.27.30. a=rtpmap:0 PCMU/8000. a=ptime:20. # U 2005/07/20 21:06:22.929462 200.94.75.4:5060 -> 200.94.75.14:5065 SIP/2.0 200 OK. Via: SIP/2.0/UDP 200.94.75.14:5065;rport=5065. From: 8183651723 <sip:8183651723@sertest.conecta-te.com>;tag=a3d69093f2a96865o0. To: <sip:018183324166@sertest.conecta-te.com>;tag=3D52A958-25AF. Date: Thu, 21 Jul 2005 02:05:50 GMT. Call-ID: 776f3ad42100670bbdf82c558882334a(a)200.94.75.14. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 101 INVITE. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO. Allow-Events: telephone-event. Contact: <sip:018183324166@200.94.75.1:5060>. Record-Route: <sip:200.94.75.4;ftag=a3d69093f2a96865o0;lr=on>. Content-Type: application/sdp. Content-Length: 188. . v=0. o=CiscoSystemsSIP-GW-UserAgent 1429 77 IN IP4 200.94.75.1. s=SIP Call. c=IN IP4 200.94.27.30. t=0 0. m=audio 16470 RTP/AVP 0. c=IN IP4 200.94.27.30. a=rtpmap:0 PCMU/8000. a=ptime:20. exit 55 received, 0 dropped

1 0

Re: [Serusers] ACK
by Sebastian Kühner 21 Jul '05

21 Jul '05

Hi, I don't have a rtpproxy.pid file. You mean the *.sock file? Here is the permission: srwxr-xr-x 1 root root 0 2005-07-20 18:18 rtpproxy.sock= The rtpproxy has to create a pid-file? Thanks! ----- Original Message ----- From: "harry gaillac" <gaillacharry(a)yahoo.fr> To: "Sebastian Kühner" <skuehner(a)veraza.com> Sent: Wednesday, July 20, 2005 5:22 PM Subject: Re: [Serusers] ACK > did you check /var/run/*/rtpproxy.pid > > --- Sebastian Kühner <skuehner(a)veraza.com> a écrit : > > > Hi! > > > > Thanks for your question ;-) > > > > I'm using Slackware... > > > > ----- Original Message ----- > > From: "harry gaillac" <gaillacharry(a)yahoo.fr> > > To: "Sebastian Kühner" <skuehner(a)veraza.com> > > Sent: Wednesday, July 20, 2005 5:07 PM > > Subject: Re: [Serusers] ACK > > > > > > > What's your distro Debian, .. ? > > > > > > --- Sebastian Kühner <skuehner(a)veraza.com> a écrit > > : > > > > > > > It should... but it doesn't. I have ser 0.9.0 > > and > > > > the latest rtpproxy > > > > version. > > > > > > > > WARNING: rtpp_test: can't get version of the RTP > > > > proxy > > > > > > > > > > > > > > > > > > > > ----- Original Message ----- > > > > From: "harry gaillac" <gaillacharry(a)yahoo.fr> > > > > To: "Sebastian Kühner" <skuehner(a)veraza.com> > > > > Sent: Wednesday, July 20, 2005 1:44 PM > > > > Subject: Re: [Serusers] ACK > > > > > > > > > > > > > your rtpproxy should work ! > > > > > > > > > > --- Sebastian Kühner <skuehner(a)veraza.com> a > > écrit > > > > : > > > > > > > > > > > Hi, > > > > > > > > > > > > Ok, my rtpproxy doesn't work, so I try it > > with > > > > STUN. > > > > > > When I look at my > > > > > > SIP-messages I get the information, that the > > > > audio > > > > > > stream has to go through > > > > > > my public IP... but I don't hear anything (I > > > > have > > > > > > the volume on maximum). > > > > > > > > > > > > The Invite comes with this message: > > > > > > > > > > > > v=0. > > > > > > o=- 3330865830 3330865830 IN IP4 > > > > xxx.xxx.xxx.xxx. > > > > > > <-- Public IP > > > > > > s=SJphone. > > > > > > c=IN IP4 xxx.xxx.xxx.xxx > > <-- > > > > > > Public IP > > > > > > t=0 0. > > > > > > a=direction:active. > > > > > > m=audio 16482 RTP/AVP 3 8 0 101. > > > > > > a=rtpmap:3 GSM/8000. > > > > > > a=rtpmap:8 PCMA/8000. > > > > > > a=rtpmap:0 PCMU/8000. > > > > > > a=rtpmap:101 telephone-event/8000. > > > > > > a=fmtp:101 0-11,16. > > > > > > > > > > > > Doesn't that mean, that the audio-stream has > > to > > > > go > > > > > > through my public IP now? > > > > > > Both sides doesn't hear anything... > > > > > > > > > > > > What's wrong? > > > > > > > > > > > > Sebastian > > > > > > > > > > > > > > > > > > > > > > > > ----- Original Message ----- > > > > > > From: "Greger V. Teigre" <greger(a)teigre.com> > > > > > > To: "Sebastian Kühner" > > <skuehner(a)veraza.com>; > > > > > > <serusers(a)lists.iptel.org> > > > > > > Sent: Wednesday, July 20, 2005 2:24 AM > > > > > > Subject: Re: [Serusers] ACK > > > > > > > > > > > > > > > > > > > Sebastian, > > > > > > > I know many people don't like STUN. > > However, I > > > > > > have good experiences with > > > > > > > STUN and prefer to use STUN as a "first > > layer > > > > > > defence." For many NATs I > > > > > > > then avoid the proxying. However, there > > are > > > > some > > > > > > things that can go wrong: > > > > > > > For one, you need to make sure that the > > STUN > > > > > > server is running correctly > > > > > > on > > > > > > > two ports and two IP addresses. If you for > > > > example > > > > > > have a firewall > > > > > > blocking > > > > > > > one port, STUN will give the wrong result. > > But > > > > the > > > > > > biggest problem can be > > > > > > > faulty STUN implementations in the EUCs. > > They > > > > > > normally behave ok for the > > > > > > > most standard NATs, but there are some > > > > > > non-standard NATs and the EUC's > > > > > > > behavior can be unpredictable. Also, some > > > > EUCs > > > > > > try to rewrite the IP:port > > > > > > > even if they are behind a symmetric NAT > > (or if > > > > the > > > > > > STUN server is not > > > > > > > correctly set up, the EUC will conclude > > with > > > > the > > > > > > wrong result). > > > > > > > If you know the clients you are going > > to > > > > use, > > > > > > you can test and limit > > > > > > the > > > > > > > problems and STUN can be a great cost > > saver! > > > > If > > > > > > your gateway supports > > > > > > > active media (direction=active), then you > > only > > > > > > have IP-2-IP phone calls to > > > > > > > proxy. > > > > > > > > > > > > > > To your question: Sipura has a good > > > > implementation > > > > > > of STUN, but has MANY > > > > > > > options for NAT. Your problem is that the > > RTP > > > > and > > > > > > RTCP is not traversing > > > > > > the > > > > > > > NAT to your Sipura. Either you don't > > force > > > > > > proxying in onreply for OKs, > > > > > > or > > > > > > > something goes wrong. An ngrep trace of > > the > > > > call > > > > > > setup will reveal what > > > > > > the > > > > > > > problem can be. > > > > > > > g-) > > > > > > > > > > > > > > Sebastian Kühner wrote: > > > > > > > > Thank you Nils, > > > > > > > > > > > > > > > > Now it's working better! > > > > > > > > > > > > > > > > The problem that I have now is that I > > don't > > > > hear > > > > > > anything if I call > > > > > > > > from the SIPURA to a Gateway, but the > > callee > > > > is > > > > > > hearing me. > > > > > > > > > > > > > > > > What could be the problem of that > > one-way > > > > > > conversation? Had anyone of > > > > > > > > you the same problem using a Restricted > > Cone > > > > > > NAT? > > > > > > > > > > > > > > > > Thanks! > > > > > > > > > > > > > > > > Sebastian > > > > > > > > > > > > > > > > > > > > > > > > ----- Original Message ----- > > > > > > > > From: "Nils Ohlmeier" > > <lists(a)ohlmeier.org> > > > > > > > > To: <serusers(a)lists.iptel.org> > > > > > > > > Cc: "Sebastian Kühner" > > <skuehner(a)veraza.com> > > > > > > > > Sent: Tuesday, July 19, 2005 3:58 PM > > > === message truncated === > > > > > > > > ___________________________________________________________________________ > Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger > Téléchargez cette version sur http://fr.messenger.yahoo.com > >

2 3

[Users] hello
by Iqbal 21 Jul '05

21 Jul '05

Hi Just a quick hello, thought I'd might as well join this list. Maybe I'll get round to installing it soon, and seeing whats diff, I believe the config files ser--->openser is the same, Is there any problem running both ser and openser, on the same machine. Iqbal

3 2

[Users] users acl
by Daniel-Constantin Mierla 21 Jul '05

21 Jul '05

Hello, the access control list in openser is based on group membership checking which does a database query each time when user's ACL is verified. We are considering to optimize this operations since they are very often used and propose the following solutions: 1. Load all groups to whom an user belongs once per request processing (one database query) and then store some bitmap flags to be used further when doing group checking. At start up time, some fixups will be made to replace the names with positions in bitmap - advantages: the old group table structure is used and changes in script should be minimal - disadvantages: after loading all group names, string comparisons are required to set the bitmap 2. Introduce a new column in the subscriber table that stores the ACL bitmap and load it once per request processing - advantages: very fast load and checking -- old version of group membership checking is kept - disadvantages: more complex provisioning system What do you think? Any other idea? Daniel

4 7

[Users] Re: [Devel] accounting source n dest IP address
by Bogdan-Andrei Iancu 21 Jul '05

21 Jul '05

Hi Jayesh, in OpenSER 0.9.5, you may try the following configuration of the acc module: .... modparam("acc","db_extra","src_ip=avp/i:10") .... .... # were setting the acc flag do also avp_write("src_ip","i:10"); ..... The idea is to set the acc to extra account the i:10 (ID 10) AVP in which, from script, you store the SRC IP of the message. NOTE you need to add the src_ip column to your acc/missed_calls table. the destination IP cannot be logged yet..... regards, bogdan jayesh nambiar wrote: > hey Bogdan, > > Thanx a lot for that information. I am completely new to Openser. > I've been working on it since last 10 days only. I am still going > through the developers guide. Our company is very much intersested > in implementing the openser for our clients. > > The version that i m using is Openser-0.9.5. > > Any information regarding accounting would be very much valuable > for me. > > Thanx again. > > Bye > > ------------------------------------------------------------------------ > Yahoo! Mail for Mobile > Take Yahoo! Mail with you! > <http://us.rd.yahoo.com/evt=31132/*http://mobile.yahoo.com/learn/mail> > Check email on your mobile phone.

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sr-users July 2005