sr-users March 2007

sr-users@lists.kamailio.org

192 participants
380 discussions

Re: R: R: [Serusers] SER -> PSTN Gateway+NAT: BYE handling problem

by Greger V. Teigre

Thanks for the detailed trace, I wouldn't have spent on this without it ;-) Its' the ACK for the OK that is not done fix_nated_contact for. In fact, checking the getting started rtpproxy file, it seems that this error is also found there. All messages coming from behind NAT should have the contact fixed. Strictly speaking, RFC3261 states that the dialog state should be created from the information found in the INVITE, not the ACK on an OK. Your BYE generating entity does this error. I have seen this problem before, but has not really thought about it from the getting started config file perspective. It is thus safer to change Contact on ALL messages behind NAT, regardless of they are loose routed or not. g-) Fabio Macchi wrote: > > Hi Greger, > > > > I attached an ethereal SIP call trace of a test call ( summary and > detailed, I simple maskerade final ip numbers ): below only the INVITE > relayed from proxy to gateway: > > > > Session Initiation Protocol > > Request-Line: INVITE sip:9999001234@194.244.gatewayIP:5060 SIP/2.0 > > Method: INVITE > > [Resent Packet: False] > > Message Header > > Record-Route: <sip:194.244.Proxy__IP;ftag=12e1e2e19d527792;lr=on> > > Via: SIP/2.0/UDP 194.244.Proxy__IP;branch=z9hG4bKb24f.5133a2c4.0 > > Transport: UDP > > Sent-by Address: 194.244.Proxy__IP > > Branch: z9hG4bKb24f.5133a2c4.0 > > Via: SIP/2.0/UDP > 1.255.ua_priv__IP;rport=1176;received=213.156.ua_pub_IP;branch=z9hG4bK35ca7adf63e3094f > > Transport: UDP > > Sent-by Address: 1.255.ua_priv__IP > > RPort: 1176 > > Received: 213.156.ua_pub_IP > > Branch: z9hG4bK35ca7adf63e3094f > > From: "000003" <sip:000003@194.244.Proxy__IP>;tag=12e1e2e19d527792 > > SIP Display info: "000003" > > SIP from address: sip:000003@194.244.Proxy__IP > > SIP tag: 12e1e2e19d527792 > > To: <sip:9999001234@194.244.Proxy__IP> > > SIP to address: sip:9999001234@194.244.Proxy__IP > > Contact: <sip:000003@213.156.ua_pub_IP:1176> > > Contact Binding: <sip:000003@213.156.ua_pub_IP:1176> > > URI: <sip:000003@213.156.ua_pub_IP:1176> > > SIP contact address: sip:000003@213.156.ua_pub_IP:1176 > > Supported: replaces > > Call-ID: 953e8996cfcc4ccc(a)1.255.ua_priv__IP > > CSeq: 60577 INVITE > > Sequence Number: 60577 > > Method: INVITE > > User-Agent: Grandstream HT386 1.0.3.64 FXS0 > > Max-Forwards: 16 > > Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE > > Content-Type: application/sdp > > Content-Length: 327 > > Message body > > > > As you can see, contact informations are correctly fixed with the > pubblic UA address, but when the callee hungs up, the BYE is relayed > to the private IP address: am I missing something ? > > In this invite I see UA private address only in VIA: does BYE look to > this parameter ? > > Later caller hangs up too, and the OK is relayed to the correct IP/port. > > > > Any help would be high appreciate, thanks in advance. > > > > Fabio > > > >

17 years, 7 months

[Users] NAT Problem with OpenSER behind load balancer

by Tim Madorma

Hi, I have put the OpenSER version 1.2 behind a BIG IP load balancer which provides a static private IP address (10.1.1.20) via NAT to the OpenSER. In our application, the clients will send to one of the three public addresses that are defined as aliases: alias="65.185.232.62:5061" alias="65.185.232.62:5062" alias="65.185.232.62:5063" An incoming OPTIONS message is received with the host in the REQ URI set to 65.185.232.62:5063 is received and when it hits this section of openser.cfg: if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); }; it executes within the if conditional and tries to route the OPTIONS message. I am wondering why uri does not match myself. The debug output from the log is as follows (it seems that the alias assignments are not working): Mar 19 16:41:11 homer openser[2267]: [ID 487083 local0.debug] grep_sock_info - checking if host==us: 13==9 && [65.185.232.62] == [1 0.1.1.20] Mar 19 16:41:11 homer openser[2267]: [ID 375670 local0.debug] grep_sock_info - checking if port 5061 matches port 5063 Mar 19 16:41:11 homer openser[2267]: [ID 487083 local0.debug] grep_sock_info - checking if host==us: 13==9 && [65.185.232.62] == [1 0.1.1.20] Mar 19 16:41:11 homer openser[2267]: [ID 375670 local0.debug] grep_sock_info - checking if port 5062 matches port 5063 Mar 19 16:41:11 homer openser[2267]: [ID 487083 local0.debug] grep_sock_info - checking if host==us: 13==9 && [65.185.232.62] == [1 0.1.1.20] Mar 19 16:41:11 homer openser[2267]: [ID 375670 local0.debug] grep_sock_info - checking if port 5063 matches port 5063 Mar 19 16:41:11 homer openser[2267]: [ID 140248 local0.debug] DEBUG:check_self: host != me Any suggestions? I have not tried this in any other version. Is this a possible bug in 1.2? thanks, Tim

17 years, 7 months

RV: RV: [Users] WeSIP session question

by Antonio Abajo

Buenas tardes Tele In the capture, I see the following call flow. GW WESIP MS ------ INV(S1)-----> -------INV (S2) ------> <--------183 ----------- ------- 183 -----------> <----------ACK --------- <---------503 ---------- ... 14:07:57 19Mar2007 DEBUG SipProcessor [SipProcessor[4]]- <<<<<<<<< Request Received <<<<<<<<< INVITE sip:390104491079@82.215.163.67 SIP/2.0 Via: SIP/2.0/UDP 82.215.163.5:5060;branch=z9hG4bK6a4fa6b1 Max-Forwards: 69 From: <sip:3405300695@82.215.163.5>;tag=6a4fa6b1 To: <sip:390104491079@82.215.163.67> Call-ID: 1783604896-30787@SVIGateway CSeq: 1 INVITE Contact: <sip:82.215.163.5:5060> Content-Type: application/sdp Content-Length: 213 14:07:58 19Mar2007 DEBUG SipRequest [SipProcessor[4]]- >>>>>>>>> Sending Request >>>>>>>>> INVITE sip:199@82.215.133.50 SIP/2.0 Max-Forwards: 69 From: <sip:3405300695@82.215.163.5>;tag=01E9CCA04D3CF9B8B3BE20CE1385BC2A To: <sip:390104491079@82.215.163.67> CSeq: 1 INVITE Content-Type: application/sdp Call-ID: 11111783604896-30787@SVIGateway Contact: <sip:82.215.163.67:5060;transport=udp> Via: SIP/2.0/UDP 82.215.163.67:5060;branch=z9hG4bK477735439 Content-Length: 21314:07:58 19Mar2007 DEBUG SipProcessor [SipProcessor[4]]- <<<<<<<<< Response Received <<<<<<<<< SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 82.215.163.67:5060;branch=z9hG4bK477735439 From: <sip:3405300695@82.215.163.5>;tag=01E9CCA04D3CF9B8B3BE20CE1385BC2A To: <sip:390104491079@82.215.163.67>;tag=13FB8534-58 Date: Mon, 25 Mar 2002 04:04:50 GMT Call-ID: 11111783604896-30787@SVIGateway Server: Cisco-SIPGateway/IOS-12.x CSeq: 1 INVITE Allow: INVITE,CANCEL,ACK,BYE,INFO,OPTIONS,UPDATE,REGISTER,SUBSCRIBE,NOTIFY,PRACK,RE FER Allow-Events: telephone-event Contact: <sip:199@82.215.133.50:5060> Content-Type: application/sdp Content-Disposition: oSystemsSIP-GW-UserAg Content-Length: 235 14:07:58 19Mar2007 DEBUG SipResponse [SipProcessor[4]]- >>>>>>>>> Sending Response >>>>>>>>> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 82.215.163.5:5060;branch=z9hG4bK6a4fa6b1 Max-Forwards: 69 From: <sip:3405300695@82.215.163.5>;tag=6a4fa6b1 To: <sip:390104491079@82.215.163.67> CSeq: 1 INVITE Call-ID: 11111783604896-30787@SVIGateway Content-Type: application/sdp Content-Length: 235 v=0 o=CiscoSystemsSIP-GW-UserAgent 9756 1908 IN IP4 82.215.133.50 s=SIP Call c=IN IP4 82.215.133.50 t=0 0 m=audio 19450 RTP/AVP 0 99 c=IN IP4 82.215.133.50 a=rtpmap:0 PCMU/8000 a=rtpmap:99 telephone-event/8000 a=fmtp:99 0-15 14:08:07 19Mar2007 DEBUG SipProcessor [SipProcessor[3]]- <<<<<<<<< Request Received <<<<<<<<< ACK sip:199@82.215.133.50 SIP/2.0 Via: SIP/2.0/UDP 82.215.163.67;branch=z9hG4bK07de.99a696f6.0 From: <sip:3405300695@82.215.163.5>;tag=01E9CCA04D3CF9B8B3BE20CE1385BC2A Call-ID: 11111783604896-30787@SVIGateway To: <sip:390104491079@82.215.163.67>;tag=13FB8534-58 CSeq: 1 ACK User-Agent: OpenSer (1.2.0-pre6-notls (i386/linux)) Content-Length: 0 14:08:07 19Mar2007 DEBUG SipProcessor [SipProcessor[4]]- <<<<<<<<< Response Received <<<<<<<<< SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 82.215.163.67:5060;branch=z9hG4bK477735439 From: <sip:3405300695@82.215.163.5>;tag=01E9CCA04D3CF9B8B3BE20CE1385BC2A To: <sip:390104491079@82.215.163.67>;tag=13FB8534-58 Date: Mon, 25 Mar 2002 04:04:50 GMT Call-ID: 11111783604896-30787@SVIGateway Server: Cisco-SIPGateway/IOS-12.x CSeq: 1 INVITE Allow-Events: telephone-event Reason: Q.850;cause=41 Content-Length: 0 The signalling is not correct but, I have looked your code and believe that it is right, revise it and you check if you have the last version of WeSIP and you return to try, because the last release commited in HYPERLINK "http://www.wesip.com/mediawiki/index.php/Downloads:WeSIP_Beta_0.1"http://ww w.wesip.com/mediawiki/index.php/Downloads:WeSIP_Beta_0.1 was fixed a bug with the ApplicationSessions. In my test application I reply your behaviour(with 5xx, with AppSession attributes) and all have working perfectly, I dont understand nothing :( Sorry Good Luck. Antonio -----Mensaje original----- De: tele [mailto:tele@plexia.com] Enviado el: martes, 20 de marzo de 2007 16:46 Para: users(a)openser.org CC: antonio.abajo(a)voztele.com Asunto: Re: RV: [Users] WeSIP session question Hola Antonio, Your example works great and the code is more clear and correct than mine :) i was not clear, my problem was only with the 503 Service Unavailable response, that means a 5xx Server Failure, is it possibile that when the container or openser with seas recive a 5xx response is not able to do for example a creation of a new invite to UA3 recover from the same session?. Or probably i'm occur in a 473 response for an incorrect use of SipApplicationSession. anyway with a 480 Temporarily Unavailable response from the UA2 i solve my problem. thank you very much :tele On Mon, 2007-03-19 at 19:27 +0100, Antonio Abajo wrote: > Hola tele ;) > > I have tested an example similar to yours, and it works ok. It's de > following: > > UA1 ---[1]--> WeSIP --[2]--486 Busy Here---> UA2 > | > [3] > ---OK----_> UA3 > > [1] UA1 generates an INVITE to UA2 ant it 's forwarded to UA2 with a B2BUA > behaviour. In do INVITE. I have implemented the following: > > protected void doInvite(SipServletRequest invite){ > SipServletRequest otherInvite = sf.createRequest(invite, false); > SipURI sipUri = sf.createSipURI("UA2","proxy"); > otherInvite.setRequestURI(sipUri); > otherInvite.getSession().setAttribute("REQUEST", otherInvite); > > otherInvite.getSession().setAttribute("PEER_SESSION",invite.getSession()); > > invite.getSession().setAttribute("PEER_SESSION",otherInvite.getSession()); > invite.getSession().setAttribute("REQUEST", invite); > otherInvite.send(); > } > > [2] UA2 declines the call and a 486 response is received by WeSIP and > process it in doErrorResponse: > > protected void doErrorResponse(SipServletResponse errorResponse) { > switch(errorResponse.getStatus()){ > case 486: > SipServletRequest request = (SipServletRequest) > errorResponse.getSession().getAttribute("REQUEST"); > SipSession otherSession = (SipSession) > errorResponse.getSession().getAttribute("PEER_SESSION"); > SipURI sipUri = sf.createSipURI("UA3", "proxy"); > request.setRequestURI(sipUri); > SipServletRequest newRequest = > sf.createRequest(request,false); > newRequest.getSession().setAttribute("REQUEST", newRequest); > newRequest.getSession().setAttribute("PEER_SESSION", > otherSession); > otherSession.setAttribute("PEER_SESSION", > newRequest.getSession()); > newRequest.setHeader("X-SSVTPBX", "yes"); > newRequest.send(); > break; > } > } > > [3] UA3 Recevies the second call and it takes down. > > > I do not understand the cause of error. I annex you the code of simple sip > application example that replies to your problem. You can start it up and > checking if it works. Else if work for you, I would attempt watching the > particular case when a 5XX responses are received. > > Sorry... > > Best regards. > Antonio. > > -----Mensaje original----- > De: tele [mailto:tele@plexia.com] > Enviado el: lunes, 19 de marzo de 2007 14:33 > Para: Antonio Abajo Álvarez > Asunto: RE: [Users] WeSIP session question > > Hola Antonio, > > The scenario is more complex, i'll try to explain it: > > PSTN > | > | > UA---->GW ------> WeSIP(B2BUA) > | > | > mediaserver > > > GW: 82.215.163.5 > WeSIP: 82.215.163.67 > MS: 82.215.133.50 > > in the mediaserver there is an vxml script that i play in early media > and in case of particular event return to wesip a 503 temporaly > unavailable or e 410 Gone. So my B2BUA application have control of this > and can do stuff with the 503 and 410. > > in particular, in case of 503 temporaly unavailable i get the upstream > session and create a new invite to the media server for play another > announcement associated. in case of 410 gone i generate a new invite to > the GW with the original URI request for the correct termination. > > Yes when the session is removed i'm able to send another call. > > attached here the logs and the servlet. > > don't care about the hardcoded IP and the repeated code :-) > i'm doing testing.. > > regards, > > :tele > > > On Mon, 2007-03-19 at 13:29 +0100, Antonio Abajo Álvarez wrote: > > Hi Tele, > > > > I don't understand very well the problem., for what I understand you have > > the following: > > > > > > UA1 --------------------- WeSIP --------------------- UA2 > > > > ---INV/4XX-6XX/ACK (SS1) --> ----INV/4XX-6XX/ACK(SS2)--> > > > > You try to send another call and receive the 473 response of WeSIP... > > > > ------INV/473/ACK (SS1) --> > > > > And when the session has been removed you can send another call. > > > > If it is the case, I understand that the 473 response is send from > > application or from openser script configuration, because the internal > > behaviour of SIP doesn't send this response automatically. > > > > Can you verify if it is the case? > > > > Thank you very much... > > > > Antonio. > > > > -----Mensaje original----- > > De: users-bounces(a)openser.org [mailto:users-bounces@openser.org] En nombre > > de tele > > Enviado el: lunes, 19 de marzo de 2007 12:35 > > Para: users(a)openser.org > > Asunto: [Users] WeSIP session question > > > > Hi, > > > > I've a problem with WeSIP in B2BUA mode, in case of failed call 4xx-6xx > > correctly terminated, when i try to send another call to WeSIP i recive > > a "473 Filtered destination" then for send another call i've to wait > > WeSIP complete some management with session: > > > > 14:10:16 19Mar2007 DEBUG SipConnector [SipProcessor[3]]- recycle: > > Recycling processor SipProcessor[3] > > 14:10:56 19Mar2007 DEBUG StandardAppSessionManager > > [StandardAppSessionManager[/inapp]]- AppSession Id > > [B4D6C9C4288784A68E032D20AB78BD8E] with a number of sessions =1 > > 14:10:56 19Mar2007 DEBUG StandardAppSessionManager > > [StandardAppSessionManager[/inapp]]- SipSession > > [z9hG4bK69e6006f] in state [3] with lifetime of :74490 > > 14:10:56 19Mar2007 DEBUG StandardAppSessionManager > > [StandardAppSessionManager[/inapp]]- AppSession Id > > [E3E1D9414BC47572CBC73EC7B4A53531] with a number of sessions =1 > > 14:10:56 19Mar2007 DEBUG StandardAppSessionManager > > [StandardAppSessionManager[/inapp]]- SipSession > > [z9hG4bK69e68616] in state [3] with lifetime of :40281 > > 14:11:56 19Mar2007 DEBUG StandardAppSessionManager > > [StandardAppSessionManager[/inapp]]- AppSession Id > > [B4D6C9C4288784A68E032D20AB78BD8E] with a number of sessions =1 > > 14:11:56 19Mar2007 DEBUG StandardAppSessionManager > > [StandardAppSessionManager[/inapp]]- SipSession > > [z9hG4bK69e6006f] in state [3] with lifetime of :134500 > > 14:11:56 19Mar2007 DEBUG StandardAppSessionManager > > [StandardAppSessionManager[/inapp]]- AppSession Id > > [E3E1D9414BC47572CBC73EC7B4A53531] with a number of sessions =1 > > 14:11:56 19Mar2007 DEBUG StandardAppSessionManager > > [StandardAppSessionManager[/inapp]]- SipSession > > [z9hG4bK69e68616] in state [3] with lifetime of :100291 > > 14:12:56 19Mar2007 DEBUG StandardAppSessionManager > > [StandardAppSessionManager[/inapp]]- AppSession Id > > [B4D6C9C4288784A68E032D20AB78BD8E] with a number of sessions =0 > > 14:12:56 19Mar2007 DEBUG StandardAppSessionManager > > [StandardAppSessionManager[/inapp]]- Remove AppSession > > [B4D6C9C4288784A68E032D20AB78BD8E] > > > > When i see Remove AppSession i'm able to send another call... > > I've read the sip servlet spec about that and i'm trying to invalidate() > > or setExpires() to SiApplicationSession in case of failed call. but it's > > not clear how to do yet. > > > > i can provide the full debug if needed. > > > > thank you! > > > > regards > > > > :tele > > > > > > > > _______________________________________________ > > Users mailing list > > Users(a)openser.org > > http://openser.org/cgi-bin/mailman/listinfo/users > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.446 / Virus Database: 268.18.14/727 - Release Date: 19/03/2007 11:49 -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.446 / Virus Database: 268.18.14/727 - Release Date: 19/03/2007 11:49 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.446 / Virus Database: 268.18.14/727 - Release Date: 19/03/2007 11:49 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.446 / Virus Database: 268.18.14/727 - Release Date: 19/03/2007 11:49

17 years, 7 months

[Users] DB migration

by Leonid Fainshtein

Hi, I didn't understand the proposed process of MySQL DB migration from ver.1.1 to 1.2 For example, the result of "openser_mysqldb.sh migrate openser openser12" will be a new database openser12. But how can I copy/rename openser12 to openser? Does it mean that from now, all of my auxiliary software must work with openser12? Thank you, Leonid Fainshtein ******************************************** This outgoing mail message was scanned by McAfee GroupShield Engine

17 years, 7 months

[Users] Help with dbtext location table - Almost there!

by Daryl Sanders

I have been struggling to get usrloc to work with... modparam("usrloc", "db_mode", 1) OpenSER 1.2 starts fine but gives the following error when a phone tries to register: 2(26634) db_insert_ucontact(): Error while inserting contact 2(26634) ERROR:usrloc:insert_ucontact: failed to insert in database Regards, Daryl

17 years, 7 months

RV: [Users] WeSIP session question

by Antonio Abajo

Hola tele ;) I have tested an example similar to yours, and it works ok. It's de following: UA1 ---[1]--> WeSIP --[2]--486 Busy Here---> UA2 | [3] ---OK----_> UA3 [1] UA1 generates an INVITE to UA2 ant it 's forwarded to UA2 with a B2BUA behaviour. In do INVITE. I have implemented the following: protected void doInvite(SipServletRequest invite){ SipServletRequest otherInvite = sf.createRequest(invite, false); SipURI sipUri = sf.createSipURI("UA2","proxy"); otherInvite.setRequestURI(sipUri); otherInvite.getSession().setAttribute("REQUEST", otherInvite); otherInvite.getSession().setAttribute("PEER_SESSION",invite.getSession()); invite.getSession().setAttribute("PEER_SESSION",otherInvite.getSession()); invite.getSession().setAttribute("REQUEST", invite); otherInvite.send(); } [2] UA2 declines the call and a 486 response is received by WeSIP and process it in doErrorResponse: protected void doErrorResponse(SipServletResponse errorResponse) { switch(errorResponse.getStatus()){ case 486: SipServletRequest request = (SipServletRequest) errorResponse.getSession().getAttribute("REQUEST"); SipSession otherSession = (SipSession) errorResponse.getSession().getAttribute("PEER_SESSION"); SipURI sipUri = sf.createSipURI("UA3", "proxy"); request.setRequestURI(sipUri); SipServletRequest newRequest = sf.createRequest(request,false); newRequest.getSession().setAttribute("REQUEST", newRequest); newRequest.getSession().setAttribute("PEER_SESSION", otherSession); otherSession.setAttribute("PEER_SESSION", newRequest.getSession()); newRequest.setHeader("X-SSVTPBX", "yes"); newRequest.send(); break; } } [3] UA3 Recevies the second call and it takes down. I do not understand the cause of error. I annex you the code of simple sip application example that replies to your problem. You can start it up and checking if it works. Else if work for you, I would attempt watching the particular case when a 5XX responses are received. Sorry... Best regards. Antonio. -----Mensaje original----- De: tele [mailto:tele@plexia.com] Enviado el: lunes, 19 de marzo de 2007 14:33 Para: Antonio Abajo Álvarez Asunto: RE: [Users] WeSIP session question Hola Antonio, The scenario is more complex, i'll try to explain it: PSTN | | UA---->GW ------> WeSIP(B2BUA) | | mediaserver GW: 82.215.163.5 WeSIP: 82.215.163.67 MS: 82.215.133.50 in the mediaserver there is an vxml script that i play in early media and in case of particular event return to wesip a 503 temporaly unavailable or e 410 Gone. So my B2BUA application have control of this and can do stuff with the 503 and 410. in particular, in case of 503 temporaly unavailable i get the upstream session and create a new invite to the media server for play another announcement associated. in case of 410 gone i generate a new invite to the GW with the original URI request for the correct termination. Yes when the session is removed i'm able to send another call. attached here the logs and the servlet. don't care about the hardcoded IP and the repeated code :-) i'm doing testing.. regards, :tele On Mon, 2007-03-19 at 13:29 +0100, Antonio Abajo Álvarez wrote: > Hi Tele, > > I don't understand very well the problem., for what I understand you have > the following: > > > UA1 --------------------- WeSIP --------------------- UA2 > > ---INV/4XX-6XX/ACK (SS1) --> ----INV/4XX-6XX/ACK(SS2)--> > > You try to send another call and receive the 473 response of WeSIP... > > ------INV/473/ACK (SS1) --> > > And when the session has been removed you can send another call. > > If it is the case, I understand that the 473 response is send from > application or from openser script configuration, because the internal > behaviour of SIP doesn't send this response automatically. > > Can you verify if it is the case? > > Thank you very much... > > Antonio. > > -----Mensaje original----- > De: users-bounces(a)openser.org [mailto:users-bounces@openser.org] En nombre > de tele > Enviado el: lunes, 19 de marzo de 2007 12:35 > Para: users(a)openser.org > Asunto: [Users] WeSIP session question > > Hi, > > I've a problem with WeSIP in B2BUA mode, in case of failed call 4xx-6xx > correctly terminated, when i try to send another call to WeSIP i recive > a "473 Filtered destination" then for send another call i've to wait > WeSIP complete some management with session: > > 14:10:16 19Mar2007 DEBUG SipConnector [SipProcessor[3]]- recycle: > Recycling processor SipProcessor[3] > 14:10:56 19Mar2007 DEBUG StandardAppSessionManager > [StandardAppSessionManager[/inapp]]- AppSession Id > [B4D6C9C4288784A68E032D20AB78BD8E] with a number of sessions =1 > 14:10:56 19Mar2007 DEBUG StandardAppSessionManager > [StandardAppSessionManager[/inapp]]- SipSession > [z9hG4bK69e6006f] in state [3] with lifetime of :74490 > 14:10:56 19Mar2007 DEBUG StandardAppSessionManager > [StandardAppSessionManager[/inapp]]- AppSession Id > [E3E1D9414BC47572CBC73EC7B4A53531] with a number of sessions =1 > 14:10:56 19Mar2007 DEBUG StandardAppSessionManager > [StandardAppSessionManager[/inapp]]- SipSession > [z9hG4bK69e68616] in state [3] with lifetime of :40281 > 14:11:56 19Mar2007 DEBUG StandardAppSessionManager > [StandardAppSessionManager[/inapp]]- AppSession Id > [B4D6C9C4288784A68E032D20AB78BD8E] with a number of sessions =1 > 14:11:56 19Mar2007 DEBUG StandardAppSessionManager > [StandardAppSessionManager[/inapp]]- SipSession > [z9hG4bK69e6006f] in state [3] with lifetime of :134500 > 14:11:56 19Mar2007 DEBUG StandardAppSessionManager > [StandardAppSessionManager[/inapp]]- AppSession Id > [E3E1D9414BC47572CBC73EC7B4A53531] with a number of sessions =1 > 14:11:56 19Mar2007 DEBUG StandardAppSessionManager > [StandardAppSessionManager[/inapp]]- SipSession > [z9hG4bK69e68616] in state [3] with lifetime of :100291 > 14:12:56 19Mar2007 DEBUG StandardAppSessionManager > [StandardAppSessionManager[/inapp]]- AppSession Id > [B4D6C9C4288784A68E032D20AB78BD8E] with a number of sessions =0 > 14:12:56 19Mar2007 DEBUG StandardAppSessionManager > [StandardAppSessionManager[/inapp]]- Remove AppSession > [B4D6C9C4288784A68E032D20AB78BD8E] > > When i see Remove AppSession i'm able to send another call... > I've read the sip servlet spec about that and i'm trying to invalidate() > or setExpires() to SiApplicationSession in case of failed call. but it's > not clear how to do yet. > > i can provide the full debug if needed. > > thank you! > > regards > > :tele > > > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.446 / Virus Database: 268.18.13/726 - Release Date: 18/03/2007 15:34

17 years, 7 months

[Users] mi_fifo question

by Andreas Granig

Hi, I have a OpenSER 1.2 with mi_fifo running, but adding an alias fails: /usr/local/openser-1.2# ./sbin/openserctl alias add 1000 sip:foo1000@mydomain.org database engine 'MYSQL' loaded Control engine 'FIFO' loaded is_user: user counter=0 entering fifo_cmd ul_add aliases 1000(a)mydomain.org sip:foo1000@mydomain.org 0 1.00 0 0 4294967295 400 Too few or too many arguments FIFO command was: :ul_add:openser_receiver_17638 aliases 1000(a)mydomain.org sip:foo1000@mydomain.org 0 1.00 0 0 4294967295 In etc/openser/openserctlrc I have ALIASES_TYPE="UL", and the mi_fifo module config looks like this: 88 loadmodule "mi_fifo.so" 89 modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") 90 modparam("mi_fifo", "fifo_mode", 0660) 91 modparam("mi_fifo", "fifo_group", "openser") 92 modparam("mi_fifo", "fifo_user", "openser") 93 modparam("mi_fifo", "reply_dir", "/tmp/") 94 modparam("mi_fifo", "reply_indent", "\t") Am I missing something somewhere? Cheers, Andreas This e-mail is confidential and may well also be legally privileged. If you have received it in error, you are on notice of its status. Please notify us immediately by reply e-mail and then delete this message from your system. Please do not copy it or use it for any purposes, or disclose its contents to any other person: to do so could be a breach of confidence. Thank you for your cooperation. Information pursuant to paragraph 14 Austrian Companies Code: UPC Austria GmbH; Registered Office: Wolfganggasse 58-60, 1120 Vienna Company Register Number: FN 189858d at the Commercial Court of Vienna

17 years, 7 months

[Users] loose_route problem with 1.2

by jh＠tutpro.com

i have in openser.cfg file listen=x.x.x.x port=5060 alias="sip.test.fi:5060" tcp_accept_aliases=yes out-of-dialog request comes in with pre-loaded route header: Route: <sip:sip.test.fi;lr;transport=TCP> loose_route() returns true although according to rr/readme it should return false: If the request is out-of-dialog (no to-tag) and there is only one Route: header indicating the local proxy, then the Route: header is removed and the function returns FALSE. this is with openser 1.2. i don't remember having this problems with openser 1.1.1. has something changed in config or what it is that i'm missing? -- juha

17 years, 7 months

[Serusers] Find an issue with FIFO after crash

by inge

Hi all, After a crash, the server runs normally SER process. In /tmp/ I've my ser_fifo with same rights than an other server. But my SERWeb says : Warning: filetype(): Lstat failed for (null) (errno=13 - Permission denied) in /var/www/client/functions.php on line 332 FIFO not running or bad path to it, file: /var/www/client/functions.php:335 I've make a test by extracting the if bloc in functions.php near line 332. I put this in an other file. When I execute "php test.php", filetype is Okay. But when I access by http://.../test.php I've got the same error than serweb. Thanks for your support. Adrien

17 years, 7 months

[Serusers] Unable to compile the xmlrpc module

by Fabian Dedu

Hi all, I'm trying to compile the ser-2.0.0+cvs20070315_src.tar.gz version and I'm obtaining the following error when trying to compile with xmlrpc support: Input: make include_modules="mysql xmlrpc" all ------------------------------------------------------------------------------------------------------------ make[1]: Entering directory `/downloads/ser2.0/ser/modules/mysql' make[1]: `mysql.so' is up to date. make[1]: Leaving directory `/downloads/ser2.0/ser/modules/mysql' make[1]: Entering directory `/downloads/ser2.0/ser/modules/xmlrpc' gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -minline-all-stringops -falign-loops -mcpu=athlon -Wall -DNAME='"ser"' -DVERSION='" 2.0.0+cvs20070309"' -DARCH='"i386"' -DOS='linux_' -DOS_QUOTED='"linux"' -DCOMPILER='"gcc 3.2.3"' -D__CPU_i386 -D__OS_linux -DSER_VER=2000000 -DCFG_DIR='"/usr/local/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DUSE_DNS_CACHE -DUSE_DNS_FAILOVER -DUSE_DST_BLACKLIST -DDBG_QM_MALLOC -DUSE_TLS -DTLS_HOOKS -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DCC_GCC_LIKE_ASM -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -DHAVE_TIMEGM -DHAVE_SIGIO_RT -DSIGINFO64_WORKARROUND -DHAVE_SELECT -I/usr/include/libxml2 -I/usr/local/include/libxml2 -I/usr/local/include -c xmlrpc.c -o xmlrpc.o xmlrpc.c: In function `open_doc': xmlrpc.c:1159: warning: implicit declaration of function `xmlReadMemory' xmlrpc.c:1160: `XML_PARSE_NOBLANKS' undeclared (first use in this function) xmlrpc.c:1160: (Each undeclared identifier is reported only once xmlrpc.c:1160: for each function it appears in.) xmlrpc.c:1161: `XML_PARSE_NONET' undeclared (first use in this function) xmlrpc.c:1162: `XML_PARSE_NOCDATA' undeclared (first use in this function) xmlrpc.c:1162: warning: assignment makes pointer from integer without a cast xmlrpc.c: In function `select_method': xmlrpc.c:1505: `XML_PARSE_NOBLANKS' undeclared (first use in this function) xmlrpc.c :1505: `XML_PARSE_NONET' undeclared (first use in this function) xmlrpc.c:1505: `XML_PARSE_NOCDATA' undeclared (first use in this function) xmlrpc.c:1505: warning: assignment makes pointer from integer without a cast make[1]: *** [xmlrpc.o] Error 1 make[1]: Leaving directory `/downloads/ser2.0/ser/modules/xmlrpc' make[1]: Entering directory `/downloads/ser2.0/ser/modules/acc_db' make[1]: `acc_db.so' is up to date. make[1]: Leaving directory `/downloads/ser2.0/ser/modules/acc_db' etc.... all the other modules are compiling okay ------------------------------------------------------------------------------------------------------------ I'm using Red Hat Enterprise AS 3.0 with all updates included. Did anyone experiencing the above compile issue regarding this new xmlrpc module ? Thank you, Fabian.

17 years, 7 months

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sr-users March 2007