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37545 discussions

[Users] regarding on openser and softphones
by raviprakash sunkara 23 May '06

23 May '06

Hello everybody, I need help from you. Please sovle my issue in regard openser and softphone . I installed openser and x-lite softphone in box. In ' LAN ' network its working fine. But I keep openser server in other place giving with specific public ip as sip-domain and also sip-proxy damain. its not working its give " Time out contact network admin " What i need to give "listen " parameter in openser.cfg , i gave ip address of the system, if i gave the public ip as " listen " parameter its giving the error... in Dedug mode..as udp_init:bind(5, ox811994,16) on xx.xx.xx.xx:cannot assign requested address. These are my issues.. please help me. please excuse in my english if any mistakes.. -- Thanks and Regards with cheers Sunkara Ravi Prakash (Voip Developer) Hyperion Technology www.hyperion-tech.com <b>

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[Users] Regard on oepnser and softphone
by raviprakash sunkara 23 May '06

23 May '06

Hello everybody, I need help from you. Please sovle my issue in regard openser and softphone . I installed openser and x-lite softphone in box. In ' LAN ' network its working fine. But I keep openser server in other place giving with specific public ip as sip-domain and also sip-proxy damain. its not working its give " Time out contact network admin " What i need to give "listen " parameter in openser.cfg , i gave ip address of the system, if i gave the public ip as " listen " parameter its giving the error... in Dedug mode... These are my issues.. please help me. please excuse in my english if any mistakes.. -- Thanks and Regards with cheers Sunkara Ravi Prakash (Voip Developer) Hyperion Technology www.hyperion-tech.com <b>

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[Users] LCR
by Andrew And 22 May '06

22 May '06

Does someone know a good document about LCR in OpenSER?

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Re: [Serusers] ser: cannot call behind a Nat using rtpproxy
by Americania .it 22 May '06

22 May '06

I've set the re-reg. to 1 minute but same problem. This is the log from X-Lite when I try to call the user ('vicky' @ LAN with ser; user 'claudio' behind Nat) : I receive this message on teh X.Lite screen : "We don't forward to private IP addresses" ---- Log ---- RECEIVE TIME: 21141441 RECEIVE << 80.105.2.110:5060 SIP/2.0 479 We don't forward to private IP addresses Via: SIP/2.0/UDP 192.168.0.102:5060;rport=12810;branch=z9hG4bK88EC2781C2F64AFA8F9194EE403C60CB;received=62.101.126.230 From: claudio <sip:claudio@80.105.2.110>;tag=546620589 To: <sip:vicky@80.105.2.110>;tag=2a08c327b8d597781b526eaf86695180.75e6 Call-ID: 01FB4B57-F823-44D9-B54A-B203472DEE31(a)192.168.0.102 CSeq: 53348 INVITE Server: Sip EXpress router (0.9.4 (i386/linux)) Content-Length: 0 Warning: 392 10.0.0.133:5060 "Noisy feedback tells: pid=13858 req_src_ip=62.101.126.230 req_src_port=12810 in_uri=sip:vicky@80.105.2.110 out_uri=sip:vicky@10.0.0.100:5060 via_cnt==1" SEND TIME: 21141444 SEND >> 80.105.2.110:5060 ACK sip:vicky@80.105.2.110 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.102:5060;rport;branch=z9hG4bK88EC2781C2F64AFA8F9194EE403C60CB From: claudio <sip:claudio@80.105.2.110>;tag=546620589 To: <sip:vicky@80.105.2.110>;tag=2a08c327b8d597781b526eaf86695180.75e6 Contact: <sip:claudio@192.168.0.102:5060> Call-ID: 01FB4B57-F823-44D9-B54A-B203472DEE31(a)192.168.0.102 CSeq: 53348 ACK Max-Forwards: 70 Content-Length: 0 ----- > >Try in your X-Lite set the Re-Registration to one minute then the default >1800. >Advanced Settings-> SIP settings -> ReRegister proxy >Ser will not detect the user agent as behind NAT and does not keep the >binding alive. >I don't know how it is done in Asterisk. Maybe Asterisk keeps in touch >with User agents by sending keep-alives no matter if the UA is behind NAT >or not. > >Ladislav > > >Americania .it wrote: >> >> >>What should I do? Something to set ? Why if I use the asterisk server >>instead of the ser one and the same voip clients (using users with the >>nat flag enabled) I can enstablish the communication without any problem ? >> >>Should I open port 8000 on the router of the Lan ? >> >>I 've attached the X-Lite Log when logging in : >> >>--------------------- >> >> >>© 2005 CounterPath Solutions Inc. All rights reserved. >>X-Lite release 1105x build stamp 21407 >>License key: B608ADE2A7C14E53835E7FF2EFAE95D0 >> >>Established SIP protocol listen on: 192.168.0.102:5060 >> >>NAT firewall type discovered is forced. >> >>Discovered Symmetric NAT Firewall >> >>SIP: 192.168.0.102:5060 >>RTP: 192.168.0.102:8000 >>NAT: 62.101.126.230 >> >>PROXY#0: 80.105.2.110:5060 >> >> >>SEND TIME: 1079327 >>SEND >> 80.105.2.110:5060 >>REGISTER sip:80.105.2.110 SIP/2.0 >>Via: SIP/2.0/UDP >>62.101.126.230:5060;rport;branch=z9hG4bK01C14B200F494466A016FE0E0E46B658 >>From: claudio <sip:claudio@80.105.2.110>;tag=4251097143 >>To: claudio <sip:claudio@80.105.2.110> >>Contact: "claudio" <sip:claudio@62.101.126.230:5060> >>Call-ID: 4DAD46A40FE546019F15CFD05AAE23FD(a)80.105.2.110 >>CSeq: 9157 REGISTER >>Expires: 1800 >>Max-Forwards: 70 >>User-Agent: X-Lite release 1105x >>Content-Length: 0 >> >> >>RECEIVE TIME: 1079425 >>RECEIVE << 80.105.2.110:5060 >>SIP/2.0 401 Unauthorized >>Via: SIP/2.0/UDP >>62.101.126.230:5060;rport=39914;branch=z9hG4bK01C14B200F494466A016FE0E0E46B658 >> >>From: claudio <sip:claudio@80.105.2.110>;tag=4251097143 >>To: claudio >><sip:claudio@80.105.2.110>;tag=2a08c327b8d597781b526eaf86695180.4235 >>Call-ID: 4DAD46A40FE546019F15CFD05AAE23FD(a)80.105.2.110 >>CSeq: 9157 REGISTER >>WWW-Authenticate: Digest realm="80.105.2.110", >>nonce="446f5b8ae893e0b8c2a781a06a5257a485ebe0b6" >>Server: Sip EXpress router (0.9.4 (i386/linux)) >>Content-Length: 0 >>Warning: 392 10.0.0.133:5060 "Noisy feedback tells: pid=13858 >>req_src_ip=62.101.126.230 req_src_port=39914 in_uri=sip:80.105.2.110 >>out_uri=sip:80.105.2.110 via_cnt==1" >> >> >>SEND TIME: 1079427 >>SEND >> 80.105.2.110:5060 >>REGISTER sip:80.105.2.110 SIP/2.0 >>Via: SIP/2.0/UDP >>62.101.126.230:5060;rport;branch=z9hG4bKF82CE6FE2C624791B7ABF06D7B8E7BFD >>From: claudio <sip:claudio@80.105.2.110>;tag=4251097143 >>To: claudio <sip:claudio@80.105.2.110> >>Contact: "claudio" <sip:claudio@62.101.126.230:5060> >>Call-ID: 4DAD46A40FE546019F15CFD05AAE23FD(a)80.105.2.110 >>CSeq: 9158 REGISTER >>Expires: 1800 >>Authorization: Digest >>username="claudio",realm="80.105.2.110",nonce="446f5b8ae893e0b8c2a781a06a5257a485ebe0b6",response="91b05be86378bf515209116e96bd6314",uri="sip:80.105.2.110" >> >>Max-Forwards: 70 >>User-Agent: X-Lite release 1105x >>Content-Length: 0 >> >> >>RECEIVE TIME: 1079533 >>RECEIVE << 80.105.2.110:5060 >>SIP/2.0 200 OK >>Via: SIP/2.0/UDP >>62.101.126.230:5060;rport=39914;branch=z9hG4bKF82CE6FE2C624791B7ABF06D7B8E7BFD >> >>From: claudio <sip:claudio@80.105.2.110>;tag=4251097143 >>To: claudio >><sip:claudio@80.105.2.110>;tag=2a08c327b8d597781b526eaf86695180.d540 >>Call-ID: 4DAD46A40FE546019F15CFD05AAE23FD(a)80.105.2.110 >>CSeq: 9158 REGISTER >>Contact: <sip:claudio@62.101.126.230:5060>;expires=1800 >>Server: Sip EXpress router (0.9.4 (i386/linux)) >>Content-Length: 0 >>Warning: 392 10.0.0.133:5060 "Noisy feedback tells: pid=13858 >>req_src_ip=62.101.126.230 req_src_port=39914 in_uri=sip:80.105.2.110 >>out_uri=sip:80.105.2.110 via_cnt==1" >> >>--------------------------------------- >> >> >>> >>>Hi, >>>probably the solution might be that your user agent in public will have >>>to re-register every minute or shorter time. >>>It is because the NAT binding between SER and the public user will be >>>closed if there is no SIP traffic. >>> >>>Ladislav >>> >>>Americania .it wrote: >>>>I've installed ser 0.9.4 on Suse 10 and it's running with mysql. >>>> >>>>I can register users (I'm using serweb) but I can only make a call from >>>>a user inside the Lan (where ser is installed) to a user outside behind >>>>a Nat (Fastweb network in Italy, Natted). If i call from this user >>>>behind Nat the call does not reach the other user. Viceversa it works >>>>fine. >>>> >>>>I installed the nathelper module and rtpproxy too. >>>> >>>>I don't know if I configured it well . Have I got to open some >>>>additional port in my router on the lan where ser is running (I opened >>>>port 5060 udp and tcp only)? >>>> >>>>I've tried to use Asterix on another pc in the same lan and I don't have >>>>this kind of problems with the some users and some clients (3cx phone). >>>> >>>>Any suggestions ? >>>> >>>>thanks >>>> >>>> >>>>_______________________________________________ >>>>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 >> >> >

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[Users] RADIUS and multi-leg accounting
by Helge Waastad 22 May '06

22 May '06

Hi, Is there an attribute for Sip-Leg_Source and Sip-leg-Dest? When I enable multi_leg_enabled I receive: ERROR: acc: can't get code for the Sip-Leg-Source attribute I have'nt seen this attribute in any dictionary files. br hw

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[Users] LCR problem
by Andrew And 22 May '06

22 May '06

I am using LCR but I have problem that I show below. Does someone know what is wrong? 0(0) PG[224] str2valp got string $IP1.$IP2.$IP3.236 0(0) PG[200] str2valp got int 5060 0(0) PG[224] str2valp got string g1.domain.com 0(0) PG[224] str2valp got string 0(0) PG[224] str2valp got string 19991 0(0) PG[224] str2valp got string $IP1.$IP2.$IP3.208 0(0) PG[200] str2valp got int 5060 0(0) PG[224] str2valp got string g2.domain.com 0(0) PG[224] str2valp got string 0(0) PG[224] str2valp got string 19992 0(0) PG[224] str2valp got string $IP1.$IP2.$IP3.196 0(0) PG[200] str2valp got int 5060 0(0) PG[224] str2valp got string g3.domain.com 0(0) PG[224] str2valp got string 0(0) PG[224] str2valp got string 19992 * 0(0) reload_gws(): Unknown or unsupported URI scheme: 136483472 * 0(0) PG[270] db_close entry * 0(0) ERROR: lcr:mod_init(): failed to reload gateways* *0(0) init_mod(): Error while initializing module lcr* *ERROR: error while initializing modules*

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[Users] Relay in OpenSER
by Andrew And 22 May '06

22 May '06

SER, I have problem with OpenSER and ENUM, when I have 2 URI destination to 1 prefix e.164 my OpenSER is doing fork. How can I do fork? when I have 2 destination I need to send 2 Invites (FORK), but now my OpenSER is sendind only one Invite.

2 3

[Users] Empty proto in trusted table causes segfault
by Script Head 22 May '06

22 May '06

Just stumbled on this. Obviously there shouldn't be an empty field but also OpenSER should handle this gracefully.

5 6

[Users] CVS freez - testing for 1.1.0 release
by Bogdan-Andrei Iancu 22 May '06

22 May '06

Hi everybody, At midnight, Tuesday 00:00 GMT, the CVS will be frozen and the 1.1.0 branch will enter in a testing and fixing phase - no new stuff will be committed on the cvs, just bugs and conformity fixes. We encourage the users to intensively run and test this version and to report any bugs or misbehaviour. The target is to have the next release in about one month, when hopefully the cumulate fixes will provide a high level of the stability. So, please test and report any problem you found / think you found - this will help us to accelerate the process. Thanks and regards, Bogdan

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[Users] rtpproxy issue
by krishnamurthy kandi 22 May '06

22 May '06

Hi, I am getting the problem with rtp proxy in the case of call forking from ser. After forking, one callee sends 183 session progress (early media),with this rtpsession is established in rtp proxy. Then other callee accept the call, starts sending media, in this case after calling recvfrom function ,raddr structure is not containing exact destination and port, it containts previous packet IP ,current packet port at comparing session IP and remote IPs, once IP is updated, canupdate flag is disabled, then for further invalid IP comparisons , it wont update the remote IP. Thats why rtp is not getting relayed to user agents. Why raddr variable is not containing correct IP and port, why it is showing corrupted information. For my convention I am not checking canupdate flag, now it is disabled, then it is working now.any suggitions. Thanks, kkmurthy --------------------------------- Love cheap thrills? Enjoy PC-to-Phone calls to 30+ countries for just 2¢/min with Yahoo! Messenger with Voice.

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