sr-users November 2008

sr-users@lists.kamailio.org

97 participants
170 discussions

[Kamailio-Users] Fwd: Openser as Proxy and Asterisk as Registrar

by Cosimo Fadda

Hi David, thank you for you answer, 2008/11/6, ingdavidcespedes(a)cable.net.co <ingdavidcespedes(a)cable.net.co>: > > Hi! > > I had the same problem (My PC was responding everything it receives through > the default gateway "eth0" ). Everybody say that it was an Asterisk bug, > but it was a problem in the routing table. Yes, I'm new to Openser but I know quite well Asterisk, and I don't think it is a bug or a misconfiguration . Type route in a terminal and check if they are the way you are thinking. If > not, add the correct route. Tell me if it solve your problem. The route table is correct, but a strange thing happens since the SIP packet is sent from the correct nic, but un the UDP header of that packet I see as source the IP of the other nic (the one that received the orinal packet) I have to investigate a little more on this aspect anyway. C. David Cespedes > > ----- Mensaje original ----- > De: Cosimo Fadda <cfadda.lists(a)gmail.com> > Fecha: Miércoles, Noviembre 5, 2008 9:36 am > Asunto: [Kamailio-Users] Openser as Proxy and Asterisk as Registrar > > > Hi everybody > > I'd like to implement this scenario: > > > > ----------- ------------------- ---------- > > ----- > > | GW/FW > > |<----LAN1---->|eth0|Openser|eth1|<----LAN2---->|eth0|asterisk| -- > > >PSTN----------- ------------------- | ----- > > ---------- > > |_______________________________________________| > > > > > > > > > > > > Where: > > - Openser is reachable from the outside with a public IP > > forwarded to > > eth0 private address; > > > > - Asterisk is connected to Openser using another separated LAN > > - Openser acts as proxy (for requests coming from the outside) and > > Asterisk act as registrar and gateway. > > > > > > This is the description, now the questions: > > -since opneser has two different nics, how can configure it to > > properly > > send register request to asterisk? > > This is what currently happens: > > 192.168.40.68:5060: is eth0 Openser Address > > 192.168.12.165 is eth1 Openser Address > > 192.168.12.106 is Asterisk address > > 82.187.X.X is Openser public address > > Network 192.168.40.X is not reachable from net 192.168.12.X > > > > <-- SIP read from 192.168.40.68:5060: REGISTER sip:192.168.12.106 > > SIP/2.0 // 12.106 is Asterisk address > > Via: SIP/2.0/UDP 82.187.X.X;branch=z9hG4bKfe0f.21364c47.0 Via: > > SIP/2.0/UDP > > 192.168.40.254:2660 > ;rport=2265;received=82.187.Y.Ybranch=z9hG4bK532058BAC7C544E0961E038CC29B2106 > > > > From: Cfadda <sip:225@82.187.90.68 <sip%3A225(a)82.187.90.68> > >;tag=3189722840 > > To: Cfadda <sip:225@82.187.90.68 <sip%3A225(a)82.187.90.68>> > > Contact: "Cfadda" <sip:225@192.168.40.254:2660> > > Call-ID: 487DA353295C484FA58B2FB1464A2210(a)82.187.X.X > > CSeq: 65140 REGISTER > > Expires: 1800 > > Max-Forwards: 69 > > User-Agent: X-PRO release 1105x > > Content-Length: 0 > > > > > > Nov 5 10:37:28 VERBOSE[23848] logger.c: --- (12 headers 0 lines) -- > > - > > Nov 5 10:37:28 VERBOSE[23848] logger.c: Using latest REGISTER > > request > > as basis request > > Nov 5 10:37:28 VERBOSE[23848] logger.c: Sending to 82.187.X.X : > > 5060 > > (non-NAT) > > Nov 5 10:37:28 VERBOSE[23848] logger.c: Transmitting (NAT) to > > 192.168.40.68:5060: > > SIP/2.0 100 Trying > > Via: SIP/2.0/UDP 82.187.X.X > > ;branch=z9hG4bKfe0f.21364c47.0;received=192.168.40.68 > > Via: SIP/2.0/UDP > > 192.168.40.254:2660 > ;rport=2265;received=82.187.Y.Y;branch=z9hG4bK532058BAC7C544E0961E038CC29B2106 > > > > From: Cfadda <sip:225@82.187.90.68 <sip%3A225(a)82.187.90.68> > >;tag=3189722840 > > To: Cfadda <sip:225@82.187.X.X > > > Call-ID: 487DA353295C484FA58B2FB1464A2210(a)82.187.X.X > > CSeq: 65140 REGISTER > > User-Agent: Asterisk PBX > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > > Contact: <sip:225@192.168.12.106 <sip%3A225(a)192.168.12.106>> > > Content-Length: 0 > > > > > > --- > > Nov 5 10:37:28 VERBOSE[23848] logger.c: Transmitting (NAT) to > > 192.168.40.68:5060: > > SIP/2.0 401 Unauthorized > > Via: SIP/2.0/UDP 82.187.X.X > > ;branch=z9hG4bKfe0f.21364c47.0;received=192.168.40.68 > > Via: SIP/2.0/UDP > > 192.168.40.254:2660 > ;rport=2265;received=82.187.Y.Y;branch=z9hG4bK532058BAC7C544E0961E038CC29B2106 > > > > From: Cfadda <sip:225@82.187.X.X >;tag=3189722840 > > To: Cfadda <sip:225@82.187.X.X >;tag=as4c420a53 > > Call-ID: 487DA353295C484FA58B2FB1464A2210(a)82.187.X.X > > CSeq: 65140 REGISTER > > User-Agent: Asterisk PBX > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > > WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", > > nonce="05f14a3e"Content-Length: 0 > > > > So asterisk sends response to the wrong interface. > > > > How can I solve this? > > > > Thanks in advance, > > > > C. > > > > _______________________________________________ > > Users mailing list > > Users(a)lists.kamailio.org > > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > > > >

16 years

[Kamailio-Users] Fwd: Openser as Proxy and Asterisk as Registrar

by Cosimo Fadda

Hi Daniel, I'll try to use the force_send_sock function, maybe this solve the issue; btw I don't think it is a strictly Asterisk related problem, thank you for the hint, C. 2008/11/5, Daniel-Constantin Mierla <miconda(a)gmail.com>: > > Hello, > > On 11/05/08 16:36, Cosimo Fadda wrote: > >> Hi everybody >> I'd like to implement this scenario: >> >> ----------- ------------------- --------------- >> | GW/FW |<----LAN1---->|eth0|Openser|eth1|<----LAN2---->|eth0|asterisk| >> -->PSTN >> ----------- ------------------- | --------------- >> |_______________________________________________| >> >> >> >> Where: >> - Openser is reachable from the outside with a public IP forwarded to >> eth0 private address; >> >> - Asterisk is connected to Openser using another separated LAN >> - Openser acts as proxy (for requests coming from the outside) and >> Asterisk act as registrar and gateway. >> >> >> This is the description, now the questions: >> -since opneser has two different nics, how can configure it to properly >> send register request to asterisk? >> >> > if you want to force an interface for outgoing sip requests, see > force_send_sock() function: > > http://www.kamailio.org/dokuwiki/doku.php/core-cookbook:1.4.x#force_send_so… > > This is what currently happens: >> 192.168.40.68:5060: is eth0 Openser Address >> 192.168.12.165 is eth1 Openser Address >> 192.168.12.106 is Asterisk address >> 82.187.X.X is Openser public address >> Network 192.168.40.X is not reachable from net 192.168.12.X >> >> <-- SIP read from 192.168.40.68:5060: REGISTER sip:192.168.12.106SIP/2.0 // 12.106 is Asterisk address >> Via: SIP/2.0/UDP 82.187.X.X;branch=z9hG4bKfe0f.21364c47.0 Via: SIP/2.0/UDP >> 192.168.40.254:2660;rport=2265;received=82.187.Y.Ybranch=z9hG4bK532058BAC7C544E0961E038CC29B2106 >> >> From: Cfadda <sip:225@82.187.90.68 <sip%3A225(a)82.187.90.68> >> >;tag=3189722840 >> To: Cfadda <sip:225@82.187.90.68 <sip%3A225(a)82.187.90.68>> >> Contact: "Cfadda" <sip:225@192.168.40.254:2660> >> Call-ID: 487DA353295C484FA58B2FB1464A2210(a)82.187.X.X >> CSeq: 65140 REGISTER >> Expires: 1800 >> Max-Forwards: 69 >> User-Agent: X-PRO release 1105x >> Content-Length: 0 >> >> >> Nov 5 10:37:28 VERBOSE[23848] logger.c: --- (12 headers 0 lines) --- >> Nov 5 10:37:28 VERBOSE[23848] logger.c: Using latest REGISTER request as >> basis request >> Nov 5 10:37:28 VERBOSE[23848] logger.c: Sending to 82.187.X.X : 5060 >> (non-NAT) >> Nov 5 10:37:28 VERBOSE[23848] logger.c: Transmitting (NAT) to >> 192.168.40.68:5060: >> SIP/2.0 100 Trying >> Via: SIP/2.0/UDP 82.187.X.X ;branch=z9hG4bKfe0f.21364c47.0;received= >> 192.168.40.68 >> Via: SIP/2.0/UDP 192.168.40.254:2660;rport=2265;received=82.187.Y.Y;branch=z9hG4bK532058BAC7C544E0961E038CC29B2106 >> >> From: Cfadda <sip:225@82.187.90.68 <sip%3A225(a)82.187.90.68> >> >;tag=3189722840 >> To: Cfadda <sip:225@82.187.X.X > >> Call-ID: 487DA353295C484FA58B2FB1464A2210(a)82.187.X.X >> CSeq: 65140 REGISTER >> User-Agent: Asterisk PBX >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY >> Contact: <sip:225@192.168.12.106 <sip%3A225(a)192.168.12.106>> >> Content-Length: 0 >> >> >> --- >> Nov 5 10:37:28 VERBOSE[23848] logger.c: Transmitting (NAT) to >> 192.168.40.68:5060: >> SIP/2.0 401 Unauthorized >> Via: SIP/2.0/UDP 82.187.X.X ;branch=z9hG4bKfe0f.21364c47.0;received= >> 192.168.40.68 >> Via: SIP/2.0/UDP 192.168.40.254:2660;rport=2265;received=82.187.Y.Y;branch=z9hG4bK532058BAC7C544E0961E038CC29B2106 >> >> From: Cfadda <sip:225@82.187.X.X >;tag=3189722840 >> To: Cfadda <sip:225@82.187.X.X >;tag=as4c420a53 >> Call-ID: 487DA353295C484FA58B2FB1464A2210(a)82.187.X.X >> CSeq: 65140 REGISTER >> User-Agent: Asterisk PBX >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY >> WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="05f14a3e" >> Content-Length: 0 >> >> So asterisk sends response to the wrong interface. >> >> How can I solve this? >> >> > > Probably the right answer you will get on the asterisk mailing lists. > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla > http://www.asipto.com > >

16 years

[Kamailio-Users] reminder: irc devel meeting in 1h20

by Daniel-Constantin Mierla

Hello, just a quick reminder about the irc devel meeting: - channel: #kamailio - server: irc.freenode.net - time: 15:00UTC Deatils and agenda: http://www.kamailio.org/dokuwiki/doku.php/development:irc-meeting-agenda-dr… Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com

16 years

Re: [Kamailio-Users] SER/Asterisk interworking mailing list

by luzango mfupe

Hi Alex,This is good news coming from you!!! Yes We Can....!!! Rgds, Lu. On Wed, Nov 5, 2008 at 7:48 PM, <users-request(a)lists.kamailio.org> wrote: > Send Users mailing list submissions to > users(a)lists.kamailio.org > > > 5. SER/Asterisk interworking mailing list. (Alex Balashov) > > > Message: 5 > Date: Wed, 05 Nov 2008 12:04:32 -0500 > From: Alex Balashov <abalashov(a)evaristesys.com> > Subject: [Kamailio-Users] SER/Asterisk interworking mailing list. > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users(a)lists.digium.com>, Commercial and > Business-Oriented > Asterisk Discussion <asterisk-biz(a)lists.digium.com>, users > <users(a)lists.kamailio.org>, openser users < > users(a)lists.opensips.org> > Message-ID: <4911D220.1040506(a)evaristesys.com> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Greetings, > > As a developer and consultant who spends considerable time on projects > involving the fusion of Asterisk and products derived from the SER > ecosystem (OpenSER, Kamailio, OpenSIPS, the new SIP-Router), I have > found that there is a great volume of interest in this topic on the > mailing lists associated with all communities involved, but a > comparative lack of focus that results in duplicated effort and lack of > specialised response. > > This is mainly due, I think, to the fact that detailed Asterisk > experience - while common - is not a prerequisite for working with the > SER products, while for Asterisk people SER can often be a next step in > scalability and VoIP service delivery platform enhancement that they are > just getting into. And so on. There's pollution in the respective > discursive spaces; a lot of Asterisk people posting to the SER lists > ask a lot of Asterisk-specific questions in addition to any they may > have about SER which can be construed as potentially off-topic by some > members, and the opposite is true on the Asterisk lists when detailed, > involved discussion about SER occurs. > > We need to capture that discussion that exists at the overlap and is > specifically concerned with making these two systems work together, > requiring somewhat detailed and esoteric understanding of both and a > community of user support and knowledge that focuses on both of these > conceptual and product universes. > > Toward that end, I am hosting a new mailing list with this succinct > purpose, if slightly unwieldy name, and encourage all interested to > join. It is called 'SER-Asterisk-Interwork' and can be accessed for > subscription here: > > http://lists.evaristesys.com/mailman/listinfo/ser-asterisk-interwork > > The archives are available here: > > http://lists.evaristesys.com/pipermail/ser-asterisk-interwork/ > > You can post to the list at: > > ser-asterisk-interwork(a)lists.evaristesys.com > > It's the same GNU Mailman stuff you are already used to. > > While it could be argued that this cross-product discussion is valuable > to retain in both communities, I think there is considerable benefit to > creating a specialised mailing list that focuses specifically on this > integration path and the unique interoperation and configuration issues > it creates. I think it would be good to get some of this discussion off > of the SER and Asterisk-specific mailing lists where it has somewhat > marginal relevance at times and refocus it. If you agree and are > interested in this topic, you are invited to join the list. > > One last note: The SER/OpenSER community has been in a state of flux > recently, with OpenSER undergoing a name change to Kamailio and > subsequently seeing a fork. The incumbent Kamailio project is now > in the process of merging with the original SER project. The choice of > nomenclature for list is not meant to imply an endorsement of or > affinity for the IPTel SER project per se. It is just that right now it > serves the aim of terseness to use a common denominator, to refer to > this family of projects as the "SER ecosystem." Whether you are a SER, > OpenSER, Kamailio, or OpenSIPS user, you are part of that "SER > ecosystem." That is why the list is named what it is. > > Thank you, > > -- Alex > > -- > Alex Balashov > Evariste Systems > Web : http://www.evaristesys.com/ > Tel : (+1) (678) 954-0670 > Direct : (+1) (678) 954-0671 > Mobile : (+1) (706) 338-8599 > > > > ------------------------------ > > _______________________________________________ > Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > > > End of Users Digest, Vol 42, Issue 15 > ************************************* > -- Luzango Mfupe TUUNE MOBILE Tel:0128440528/0123825710 Tshwane-RSA "...Ships are safe in harbor, but they were never meant to stay there......."

16 years

Re: [Kamailio-Users] Is it possible to break a "if" stament?

by ingdavidcespedes＠cable.net.co

----- Mensaje original ----- De: Iñaki Baz Castillo <ibc(a)aliax.net> Fecha: Miércoles, Noviembre 5, 2008 6:50 pm Asunto: [Kamailio-Users] Is it possible to break a "if" stament? > Hi, is it possible to break the body of an "if"? something like: > > -------------- > if $fU == "alice" { > xlog "hello"; > break; > xlog "bye"; > } > xlog "this is after the 'if'" > -------------- > > So the result would be: > > ------- > hello > this is after the 'if' > ------- > > Is it possible? > Not, you can't do brake inside an If, only for Switch. Why don't you do if $fU == "alice" { xlog "hello"; } else{ xlog "bye"; } xlog "this is after the 'if'" > -- > Iñaki Baz Castillo > > _______________________________________________ > Users mailing list > Users(a)lists.kamailio.org > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users >

16 years

[Kamailio-Users] high processing delay

by Fabian Borot

hello All my Register request take longer than normal to process, what I mean is that an UAC sends a Register and kamailio responds almost 0.5 secs later, causing the UAC to re-transmit its request several times 0.000000 192.168.10.10 -> 192.168.1.20 SIP Request: REGISTER sip:192.168.1.20 0.469256 192.168.1.20 -> 192.168.10.10 SIP Status: 401 Unauthorized (0 bindings) 0.518263 192.168.10.10 -> 192.168.1.20 SIP Request: REGISTER sip:192.168.1.20 0.628294 192.168.10.10 -> 192.168.1.20 SIP Request: REGISTER sip:192.168.1.20 1.130722 192.168.10.10 -> 192.168.1.20 SIP Request: REGISTER sip:192.168.1.20 1.269301 192.168.1.20 -> 192.168.10.10 SIP Status: 401 Unauthorized (0 bindings) 2.224942 192.168.10.10 -> 192.168.1.20 SIP Request: REGISTER sip:192.168.1.20 2.325358 192.168.1.20 -> 192.168.10.10 SIP Status: 200 OK (1 bindings) 3.457509 192.168.1.20 -> 192.168.10.10 SIP Status: 200 OK (1 bindings) 4.497532 192.168.1.20 -> 192.168.10.10 SIP Status: 200 OK (1 bindings) the cpu shows almost 100 % idle at that time and the pc were kamailio is running has nothing else running but kamailio. this is my config file: ####### Routing Logic ######## # main request routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests xlog("L_DBG", "mylog, time [$Tf] : starting_main_logic.\n"); if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); xlog("L_DGB","mylog, time [$Tf] : Too Many Hops.\n"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); xlog("L_DGB","mylog, time [$Tf] : Message too big.\n"); exit; }; if (!method=="REGISTER") record_route(); if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (uri==myself) { if (method=="REGISTER") { xlog("L_DGB","mylog, time [$Tf] : starting to process REGISTER.Info: [$au,$ad,$ci,$ct,$cs,$rd,$si,$sp].\n"); if (!www_authorize("", "subscriber")) { xlog("L_DGB","mylog, time [$Tf] : REGISTER came without auth, sending challenge.\n"); www_challenge("", "0"); exit; }; save("location"); xlog("L_DBG","mylog, time [$Tf] : save-location successful.\n"); exit; }; if (!lookup("location")) { sl_send_reply("404", "Not Found"); xlog("L_DGB","mylog, time [$Tf] : lookup-location failed, sending 404 Not Found.\n"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; route(1); } route[1] { if (!t_relay()) { sl_reply_error(); }; exit; } _________________________________________________________________ Stay up to date on your PC, the Web, and your mobile phone with Windows Live http://clk.atdmt.com/MRT/go/119462413/direct/01/

16 years

[Kamailio-Users] Is it possible to break a "if" stament?

by Iñaki Baz Castillo

Hi, is it possible to break the body of an "if"? something like: -------------- if $fU == "alice" { xlog "hello"; break; xlog "bye"; } xlog "this is after the 'if'" -------------- So the result would be: ------- hello this is after the 'if' ------- Is it possible? -- Iñaki Baz Castillo

16 years

Re: [Kamailio-Users] high processing delay

by Daniel-Constantin Mierla

On 11/05/08 20:15, Fabian Borot wrote: > debug was set to 9, I commented that line out and now it is fine: ok > > > 0.000000 192.168.10.10 -> 192.168.1.20 SIP Request: REGISTER > sip:192.168.1.20 > 0.001650 192.168.1.20 -> 192.168.10.10 SIP Status: 401 > Unauthorized (0 bindings) > 0.191670 192.168.10.10 -> 192.168.1.20 SIP Request: REGISTER > sip:192.168.1.20 > 0.192518 192.168.1.20 -> 192.168.10.10 SIP Status: 200 OK (1 > bindings) > > but how can I check about this "syslog set to asynchronous mode" ? check your syslog.conf or so, depends on you distribution. Probably the safest is to check the manual, the usual way for old versions of syslog was to put a - (dash) in front of the file for logging. Cheers, Daniel > > regards > fabian > > > > > > Date: Wed, 5 Nov 2008 19:37:35 +0200 > > From: miconda(a)gmail.com > > To: fborot(a)hotmail.com > > CC: users(a)lists.kamailio.org > > Subject: Re: [Kamailio-Users] high processing delay > > > > hello, > > > > what is the debug level? is you syslog set to asynchronous mode? > > > > Cheers, > > Daniel > > > > > > On 11/05/08 17:43, Fabian Borot wrote: > > > hello > > > > > > All my Register request take longer than normal to process, what I > > > mean is that an UAC sends a Register and kamailio responds almost 0.5 > > > secs later, causing the UAC to re-transmit its request several times > > > > > > 0.000000 192.168.10.10 -> 192.168.1.20 SIP Request: REGISTER > > > sip:192.168.1.20 > > > 0.469256 192.168.1.20 -> 192.168.10.10 SIP Status: 401 > > > Unauthorized (0 bindings) > > > 0.518263 192.168.10.10 -> 192.168.1.20 SIP Request: REGISTER > > > sip:192.168.1.20 > > > 0.628294 192.168.10.10 -> 192.168.1.20 SIP Request: REGISTER > > > sip:192.168.1.20 > > > 1.130722 192.168.10.10 -> 192.168.1.20 SIP Request: REGISTER > > > sip:192.168.1.20 > > > 1.269301 192.168.1.20 -> 192.168.10.10 SIP Status: 401 > > > Unauthorized (0 bindings) > > > 2.224942 192.168.10.10 -> 192.168.1.20 SIP Request: REGISTER > > > sip:192.168.1.20 > > > 2.325358 192.168.1.20 -> 192.168.10.10 SIP Status: 200 OK (1 > > > bindings) > > > 3.457509 192.168.1.20 -> 192.168.10.10 SIP Status: 200 OK (1 > > > bindings) > > > 4.497532 192.168.1.20 -> 192.168.10.10 SIP Status: 200 OK (1 > > > bindings) > > > > > > > > > the cpu shows almost 100 % idle at that time and the pc were kamailio > > > is running has nothing else running but kamailio. this is my > config file: > > > > > > > > > > > > ####### Routing Logic ######## > > > > > > > > > # main request routing logic > > > > > > > > > route{ > > > # initial sanity checks -- messages with > > > # max_forwards==0, or excessively long requests > > > xlog("L_DBG", "mylog, time [$Tf] : starting_main_logic.\n"); > > > if (!mf_process_maxfwd_header("10")) { > > > sl_send_reply("483","Too Many Hops"); > > > xlog("L_DGB","mylog, time [$Tf] : Too Many Hops.\n"); > > > exit; > > > }; > > > > > > if (msg:len >= 2048 ) { > > > sl_send_reply("513", "Message too big"); > > > xlog("L_DGB","mylog, time [$Tf] : Message too big.\n"); > > > exit; > > > }; > > > > > > > > > if (!method=="REGISTER") > > > record_route(); > > > > > > if (loose_route()) { > > > # mark routing logic in request > > > append_hf("P-hint: rr-enforced\r\n"); > > > route(1); > > > }; > > > > > > > > > if (uri==myself) { > > > if (method=="REGISTER") { > > > xlog("L_DGB","mylog, time [$Tf] : starting to > > > process REGISTER.Info: [$au,$ad,$ci,$ct,$cs,$rd,$si,$sp].\n"); > > > if (!www_authorize("", "subscriber")) { > > > xlog("L_DGB","mylog, time [$Tf] : REGISTER > > > came without auth, sending challenge.\n"); > > > www_challenge("", "0"); > > > exit; > > > }; > > > save("location"); > > > xlog("L_DBG","mylog, time [$Tf] : save-location successful.\n"); > > > exit; > > > }; > > > > > > > > > if (!lookup("location")) { > > > sl_send_reply("404", "Not Found"); > > > xlog("L_DGB","mylog, time [$Tf] : lookup-location > > > failed, sending 404 Not Found.\n"); > > > exit; > > > }; > > > append_hf("P-hint: usrloc applied\r\n"); > > > }; > > > route(1); > > > } > > > > > > route[1] { > > > if (!t_relay()) { > > > sl_reply_error(); > > > }; > > > exit; > > > } > > > > > > > > > > > > > ------------------------------------------------------------------------ > > > Stay up to date on your PC, the Web, and your mobile phone with > > > Windows Live Click here > > > <http://clk.atdmt.com/MRT/go/119462413/direct/01/> > > > > ------------------------------------------------------------------------ > > > > > > _______________________________________________ > > > Users mailing list > > > Users(a)lists.kamailio.org > > > http://lists.kamailio.org/cgi-bin/mailman/listinfo/users > > > > > > > -- > > Daniel-Constantin Mierla > > http://www.asipto.com > > > > ------------------------------------------------------------------------ > Color coding for safety: Windows Live Hotmail alerts you to suspicious > email. Sign up today. > <http://windowslive.com/Explore/Hotmail?ocid=TXT_TAGLM_WL_hotmail_acq_safety…> -- Daniel-Constantin Mierla http://www.asipto.com

16 years

[Kamailio-Users] Openser as Proxy and Asterisk as Registrar

by Cosimo Fadda

Hi everybody I'd like to implement this scenario: ----------- ------------------- --------------- | GW/FW |<----LAN1---->|eth0|Openser|eth1|<----LAN2---->|eth0|asterisk| -->PSTN ----------- ------------------- | --------------- |_______________________________________________| Where: - Openser is reachable from the outside with a public IP forwarded to eth0 private address; - Asterisk is connected to Openser using another separated LAN - Openser acts as proxy (for requests coming from the outside) and Asterisk act as registrar and gateway. This is the description, now the questions: -since opneser has two different nics, how can configure it to properly send register request to asterisk? This is what currently happens: 192.168.40.68:5060: is eth0 Openser Address 192.168.12.165 is eth1 Openser Address 192.168.12.106 is Asterisk address 82.187.X.X is Openser public address Network 192.168.40.X is not reachable from net 192.168.12.X <-- SIP read from 192.168.40.68:5060: REGISTER sip:192.168.12.106 SIP/2.0 // 12.106 is Asterisk address Via: SIP/2.0/UDP 82.187.X.X;branch=z9hG4bKfe0f.21364c47.0 Via: SIP/2.0/UDP 192.168.40.254:2660;rport=2265;received=82.187.Y.Ybranch=z9hG4bK532058BAC7C544E0961E038CC29B2106 From: Cfadda <sip:225@82.187.90.68>;tag=3189722840 To: Cfadda <sip:225@82.187.90.68> Contact: "Cfadda" <sip:225@192.168.40.254:2660> Call-ID: 487DA353295C484FA58B2FB1464A2210(a)82.187.X.X CSeq: 65140 REGISTER Expires: 1800 Max-Forwards: 69 User-Agent: X-PRO release 1105x Content-Length: 0 Nov 5 10:37:28 VERBOSE[23848] logger.c: --- (12 headers 0 lines) --- Nov 5 10:37:28 VERBOSE[23848] logger.c: Using latest REGISTER request as basis request Nov 5 10:37:28 VERBOSE[23848] logger.c: Sending to 82.187.X.X : 5060 (non-NAT) Nov 5 10:37:28 VERBOSE[23848] logger.c: Transmitting (NAT) to 192.168.40.68:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 82.187.X.X ;branch=z9hG4bKfe0f.21364c47.0;received=192.168.40.68 Via: SIP/2.0/UDP 192.168.40.254:2660;rport=2265;received=82.187.Y.Y;branch=z9hG4bK532058BAC7C544E0961E038CC29B2106 From: Cfadda <sip:225@82.187.90.68>;tag=3189722840 To: Cfadda <sip:225@82.187.X.X > Call-ID: 487DA353295C484FA58B2FB1464A2210(a)82.187.X.X CSeq: 65140 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:225@192.168.12.106> Content-Length: 0 --- Nov 5 10:37:28 VERBOSE[23848] logger.c: Transmitting (NAT) to 192.168.40.68:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 82.187.X.X ;branch=z9hG4bKfe0f.21364c47.0;received=192.168.40.68 Via: SIP/2.0/UDP 192.168.40.254:2660;rport=2265;received=82.187.Y.Y;branch=z9hG4bK532058BAC7C544E0961E038CC29B2106 From: Cfadda <sip:225@82.187.X.X >;tag=3189722840 To: Cfadda <sip:225@82.187.X.X >;tag=as4c420a53 Call-ID: 487DA353295C484FA58B2FB1464A2210(a)82.187.X.X CSeq: 65140 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="05f14a3e" Content-Length: 0 So asterisk sends response to the wrong interface. How can I solve this? Thanks in advance, C.

16 years

[Serusers] DNS failover

by Nuno Ribeiro

Hi all, I have a scenario where I was testing DNS failover and this feature seems to fail. I will try to explain the scenario: I'm trying to establish a call between two user Agents, where one of the users has services associated (trigger to ASs) to the session establishment signaling (INVITE). So when SER makes the DNS query receives 2 ip addresses for this AS. The sip message will be correctly forwarded to one of this two entries and if in case of failure it will try to send the message to the next entry (second ip address). But when we are triggering to the AS we will add a route header and this header is not created in the second case, i.e. after DNS failover as the SIP INVITE will be created based in the SIP message received and not in the SIP message sent in the first place. So anyone has any thoughts about this question and a solution to it? Thanks in any advance. Best Regards -- Nuno Ribeiro

16 years

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sr-users November 2008